In my post I have 2 examples: H.323 trunk and SIP trunk.
For SIP trunk I have 2 configurations: ABCF trunk group or ISDN trunk group. Routing is done either through routing number or ARS (ABCF trunk) or just ARS (ISDN trunk group).
Instead of 192.168.0.123 in my example should be the IP of Alcatel GD (this is for H.323 trunk). The string dials using ooh323 package for H.323 protocol in Asterisk (config file should be ooh323.conf). You need H.323 trunk licenses in OXE.
connecting elastix (freebpx based, witch is based on asterisk) to omnipcx enterprise
Re: connecting elastix (freebpx based, witch is based on ast
I'm really stuck 
My existing PBX have ISDN-SIP configured already which we use to connect it to other audio codec. And I check on trkstat #trunk, all state is Free can use.
On Elastix I'm doing the same setting as your setting already, but I'm using #1 to call OXE Extension, It's running well I can call from Elastix to Alcatel.
The problem is I can't call from Alcatel to Elastix Extension.
When I create Ext gateway on Alcatel for my ISDN trunk, my Elastix become can't use. I try #1 to call to Alcatel the error message is "All circuit is busy now"
I've no Idea about this.. after try to check almost 2 months
If I delete the Ext gateway on Alcatel, the Elastix can use again.
I'm really need help to create call from Alcatel to Elastix
sorry I'm Newbie on Alcatel, just learn this system for a few months.. also can't find good Alcatel training school in my country philippine.
I'm also have one problem, If I create a new ABC-F SIP trunk, the state always in hs
My existing PBX have ISDN-SIP configured already which we use to connect it to other audio codec. And I check on trkstat #trunk, all state is Free can use.
On Elastix I'm doing the same setting as your setting already, but I'm using #1 to call OXE Extension, It's running well I can call from Elastix to Alcatel.
The problem is I can't call from Alcatel to Elastix Extension.
When I create Ext gateway on Alcatel for my ISDN trunk, my Elastix become can't use. I try #1 to call to Alcatel the error message is "All circuit is busy now"
I've no Idea about this.. after try to check almost 2 months
If I delete the Ext gateway on Alcatel, the Elastix can use again.
I'm really need help to create call from Alcatel to Elastix
sorry I'm Newbie on Alcatel, just learn this system for a few months.. also can't find good Alcatel training school in my country philippine.
I'm also have one problem, If I create a new ABC-F SIP trunk, the state always in hs
-
tot3nkopf
- Alcatel Unleashed Certified Guru
- Posts: 4058
- Joined: 02 Feb 2006 10:41
- Location: Germany & Romania
- Contact:
Re: connecting elastix (freebpx based, witch is based on ast
Post your complete configuration and traces. We cannot guess...
It looks like you are a newbie to Asterisk also. Translation configured in Asterisk? Do you have licenses for additional SIP channels in Alcatel in order to create another TG? Channels should be up when external gateway is associated and gateway alive (registration ok if configured - workaround is to disable registration-->registration timer=0 --> see my example).
Why don't you try my example first and then customize it for your use?
It looks like you are a newbie to Asterisk also. Translation configured in Asterisk? Do you have licenses for additional SIP channels in Alcatel in order to create another TG? Channels should be up when external gateway is associated and gateway alive (registration ok if configured - workaround is to disable registration-->registration timer=0 --> see my example).
Why don't you try my example first and then customize it for your use?
Re: connecting elastix (freebpx based, witch is based on ast
Thanks for your reply
Here is my configuration setup
SIP status for Elastix
+==============================================================================+
| S I P T R U N K S T A T E Trunk group number : 20 |
| Trunk group name : sip int |
| Number of Trunks : 62 |
+------------------------------------------------------------------------------+
| Index : 1 2 3 4 5 6 7 8 9 10 11 12 13 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 14 15 16 17 18 19 20 21 22 23 24 25 26 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 27 28 29 30 31 32 33 34 35 36 37 38 39 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 40 41 42 43 44 45 46 47 48 49 50 51 52 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 53 54 55 56 57 58 59 60 61 62 |
| State : F F F F F F F F F F |
+------------------------------------------------------------------------------+
| F: Free | B: Busy | Ct: busy Comp trunk | Cl: busy Comp link |
| WB: Busy Without B Channel| Cr: busy Comp trunk for RLIO inter-ACT link |
| WBD: Data Transparency without chan.| WBM: Modem transparency without chan. |
| D: Data Transparency | M: Modem transparency |
+------------------------------------------------------------------------------+
Trunk Group ID 20
┌─Review/Modify: Trunk Groups───────────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Trunk Group ID : 20 │
│ │
│ Trunk Group Type + T2 │
│ Trunk Group Name : Elastix Sip │
│ UTF-8 Trunk Group Name : ------------------------------------------- │
│ Number Compatible With : -1 │
│ Remote Network : 11 │
│ Shared Trunk Group + False │
│ Special Services + Nothing │
│ Node number : 1 │
│ Transcom Trunk Group + False │
│ Auto.reserv.by Attendant + False │
│ Overflow trunk group No. : -1 │
│ Tone on seizure + False │
│ Private Trunk Group + False │
│ Q931 Signal variant + ISDN all countries │
│ SS7 Signal variant + No variant │
│ Number Of Digits To Send : 0 │
│ Channel selection type + Quantified │
│ Auto.DTMF dialing on outgoing call + NO │
│ T2 Specification + SIP │
│ Homogenous network for direct RTP + NO │
│ Public Network COS : 31 │
│ DID transcoding + False │
│ Can support UUS in SETUP + True │
│ │
│ Implicit Priority │
│ │
│ Activation mode : 0 │
│ Priority Level : 0 │
│ │
│ Preempter + NO │
│ Incoming calls Restriction COS : 10 │
│ Outgoing calls Restriction COS : 10 │
│ Callee number mpt1343 + NO │
│ Overlap dialing + YES │
│ Call diversion in ISDN + NO │
│ │
└───────────────────────────────────────────────────────────────────────────────────┘
SIP gateway
┌─Review/Modify: SIP Gateway────────────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ │
│ SIP Subnetwork : 11 │
│ SIP Trunk Group : 20 │
│ IP Address : 10.1.0.1 │
│ Machine name - Host : host │
│ SIP Proxy Port Number : 5060 │
│ SIP Subscribe Min Duration : 1800 │
│ SIP Subscribe Max Duration : 86400 │
│ Session Timer : 1800 │
│ Min Session Timer : 900 │
│ Session Timer Method + RE_INVITE │
│ DNS local domain name : -------------------------------------------- │
│ DNS type + DNS A │
│ SIP DNS1 IP Address : -------------------------------------------- │
│ SIP DNS2 IP Address : -------------------------------------------- │
│ SDP in 18x + True │
│ Cac SIP-SIP + False │
│ INFO method for remote extension + False │
│ Dynamic Payload type for DTMF : 97 │
│ │
└───────────────────────────────────────────────────────────────────────────────────┘
SP admin SIP license status
SIP Users : 29/31
For the SIP external gateway and the next step haven't create it If I create external gateway same as ur configuration my trunk 20 will become HS.
And I want try using ABC-F method, but can't create new trunk don't know what happen become HS, note :trunk 20 is on productivity
Elastix Setup
SIP Extension setup 100
SIP trunk OXE SIP
Peer Details
host=10.1.0.1
type=peer
insecure=very
context=from-pstn
transfer=yes
immediate=no
dtmfmode=outofband
dtmf=rfc2833
nat=no
canreinvite=no ;yes
qualify=no; yes
Outbound route = #2
Dial Pattern = #2|.
Here is my configuration setup
SIP status for Elastix
+==============================================================================+
| S I P T R U N K S T A T E Trunk group number : 20 |
| Trunk group name : sip int |
| Number of Trunks : 62 |
+------------------------------------------------------------------------------+
| Index : 1 2 3 4 5 6 7 8 9 10 11 12 13 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 14 15 16 17 18 19 20 21 22 23 24 25 26 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 27 28 29 30 31 32 33 34 35 36 37 38 39 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 40 41 42 43 44 45 46 47 48 49 50 51 52 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 53 54 55 56 57 58 59 60 61 62 |
| State : F F F F F F F F F F |
+------------------------------------------------------------------------------+
| F: Free | B: Busy | Ct: busy Comp trunk | Cl: busy Comp link |
| WB: Busy Without B Channel| Cr: busy Comp trunk for RLIO inter-ACT link |
| WBD: Data Transparency without chan.| WBM: Modem transparency without chan. |
| D: Data Transparency | M: Modem transparency |
+------------------------------------------------------------------------------+
Trunk Group ID 20
┌─Review/Modify: Trunk Groups───────────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Trunk Group ID : 20 │
│ │
│ Trunk Group Type + T2 │
│ Trunk Group Name : Elastix Sip │
│ UTF-8 Trunk Group Name : ------------------------------------------- │
│ Number Compatible With : -1 │
│ Remote Network : 11 │
│ Shared Trunk Group + False │
│ Special Services + Nothing │
│ Node number : 1 │
│ Transcom Trunk Group + False │
│ Auto.reserv.by Attendant + False │
│ Overflow trunk group No. : -1 │
│ Tone on seizure + False │
│ Private Trunk Group + False │
│ Q931 Signal variant + ISDN all countries │
│ SS7 Signal variant + No variant │
│ Number Of Digits To Send : 0 │
│ Channel selection type + Quantified │
│ Auto.DTMF dialing on outgoing call + NO │
│ T2 Specification + SIP │
│ Homogenous network for direct RTP + NO │
│ Public Network COS : 31 │
│ DID transcoding + False │
│ Can support UUS in SETUP + True │
│ │
│ Implicit Priority │
│ │
│ Activation mode : 0 │
│ Priority Level : 0 │
│ │
│ Preempter + NO │
│ Incoming calls Restriction COS : 10 │
│ Outgoing calls Restriction COS : 10 │
│ Callee number mpt1343 + NO │
│ Overlap dialing + YES │
│ Call diversion in ISDN + NO │
│ │
└───────────────────────────────────────────────────────────────────────────────────┘
SIP gateway
┌─Review/Modify: SIP Gateway────────────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ │
│ SIP Subnetwork : 11 │
│ SIP Trunk Group : 20 │
│ IP Address : 10.1.0.1 │
│ Machine name - Host : host │
│ SIP Proxy Port Number : 5060 │
│ SIP Subscribe Min Duration : 1800 │
│ SIP Subscribe Max Duration : 86400 │
│ Session Timer : 1800 │
│ Min Session Timer : 900 │
│ Session Timer Method + RE_INVITE │
│ DNS local domain name : -------------------------------------------- │
│ DNS type + DNS A │
│ SIP DNS1 IP Address : -------------------------------------------- │
│ SIP DNS2 IP Address : -------------------------------------------- │
│ SDP in 18x + True │
│ Cac SIP-SIP + False │
│ INFO method for remote extension + False │
│ Dynamic Payload type for DTMF : 97 │
│ │
└───────────────────────────────────────────────────────────────────────────────────┘
SP admin SIP license status
SIP Users : 29/31
For the SIP external gateway and the next step haven't create it
If I create external gateway same as ur configuration my trunk 20 will become HS.And I want try using ABC-F method, but can't create new trunk don't know what happen become HS
, note :trunk 20 is on productivityElastix Setup
SIP Extension setup 100
SIP trunk OXE SIP
Peer Details
host=10.1.0.1
type=peer
insecure=very
context=from-pstn
transfer=yes
immediate=no
dtmfmode=outofband
dtmf=rfc2833
nat=no
canreinvite=no ;yes
qualify=no; yes
Outbound route = #2
Dial Pattern = #2|.
Re: connecting elastix (freebpx based, witch is based on ast
Hi anyone can help me? 
Re: connecting elastix (freebpx based, witch is based on ast
Hi
Now From Elastix can call to Alcatel Extension , working well
From Alcatel also can call to Elastix Extension
The problem is from Elastix can't use the Alcatel trunk
the error message is all circuit are busy now
this one the log
[KElastix*CLI>
[0K -- SIP/oxe-00000016 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Please help me to fix this, I'm almost finnish the setup, left this issue
thanks
Now From Elastix can call to Alcatel Extension , working well
From Alcatel also can call to Elastix Extension
The problem is from Elastix can't use the Alcatel trunk
the error message is all circuit are busy now
this one the log
[KElastix*CLI>
[0K -- SIP/oxe-00000016 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Please help me to fix this, I'm almost finnish the setup, left this issue
thanks
-
tot3nkopf
- Alcatel Unleashed Certified Guru
- Posts: 4058
- Joined: 02 Feb 2006 10:41
- Location: Germany & Romania
- Contact:
Re: connecting elastix (freebpx based, witch is based on ast
Post traces of a call. Is external gateway in OXE up?
Re: connecting elastix (freebpx based, witch is based on ast
i can call from elastix to oxe but 1 to 10 calls could be established the other gives an infernal beep !!
Re: connecting elastix (freebpx based, witch is based on ast
Yes the External gateway in OXE is uptot3nkopf wrote:Post traces of a call. Is external gateway in OXE up?
This is traces of a call from Elastix to -> Alcatel
Note : Alcatel External gateway using ABC-F SIP trunk
Dial out number is 998564551 << first 9 is Dial Rule at asterisk for go to Alcatel, second 9 is Alcatel Prefix number for call out and the rest is local number at here
Code: Select all
[KElastix*CLI> sip set debug on
Elastix*CLI> [0KSIP Debugging enabled
[KElastix*CLI> [0KReliably Transmitting (NAT) to 10.100.0.68:5060:
OPTIONS sip:0288@10.100.0.68:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK0c92970f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.100.0.151>;tag=as278dbb8d
To: <sip:0288@10.100.0.68:5060;user=phone>
Contact: <sip:Unknown@10.100.0.151>
Call-ID: 2c031bcb5f2ab84837a1de4055725129@10.100.0.151
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Tue, 13 Sep 2011 00:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
[KElastix*CLI> [0K
<--- SIP read from UDP:10.100.0.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK0c92970f;rport
From: "Unknown"<sip:Unknown@10.100.0.151>;tag=as278dbb8d
To: <sip:0288@10.100.0.68:5060;user=phone>;tag=c0a80101-4dcc993
Call-ID: 2c031bcb5f2ab84837a1de4055725129@10.100.0.151
CSeq: 102 OPTIONS
Contact: <sip:0288@10.100.0.68:5060;user=phone>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Supported: timer, replaces
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 270
v=0
o=0288 81578386 81578386 IN IP4 10.100.0.68
s=-
c=IN IP4 10.100.0.68
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
<------------->
[KElastix*CLI> [0K--- (12 headers 13 lines) ---
Really destroying SIP dialog '2c031bcb5f2ab84837a1de4055725129@10.100.0.151' Method: OPTIONS
[KElastix*CLI> [0K
<--- SIP read from UDP:10.100.0.68:5060 --->
INVITE sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142020860769754754
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4dcd9b6
To: <sip:998564551@Elastix:5060;user=phone>
Call-ID: 1353ff46-c0a80101-0-5@10.100.0.68
CSeq: 1 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:0288@10.100.0.68:5060;user=phone>
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 270
v=0
o=0288 81582519 81582519 IN IP4 10.100.0.68
s=-
c=IN IP4 10.100.0.68
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
<------------->
--- (15 headers 13 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 10.100.0.68 : 5060 (no NAT)
Using INVITE request as basis request - 1353ff46-c0a80101-0-5@10.100.0.68
Found peer '0288' for '0288' from 10.100.0.68:5060
<--- Reliably Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142020860769754754;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4dcd9b6
To: <sip:998564551@Elastix:5060;user=phone>;tag=as2eb7cdcd
Call-ID: 1353ff46-c0a80101-0-5@10.100.0.68
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7cdd2046"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1353ff46-c0a80101-0-5@10.100.0.68' in 6400 ms (Method: INVITE)
[KElastix*CLI> [0K
<--- SIP read from UDP:10.100.0.68:5060 --->
ACK sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142020860769754754
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4dcd9b6
To: <sip:998564551@Elastix:5060;user=phone>;tag=as2eb7cdcd
Call-ID: 1353ff46-c0a80101-0-5@10.100.0.68
CSeq: 1 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Content-Length: 0
<------------->
[KElastix*CLI> [0K--- (10 headers 0 lines) ---
[KElastix*CLI> [0K
<--- SIP read from UDP:10.100.0.68:5060 --->
INVITE sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK252531915214765865
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4dcd9b6
To: <sip:998564551@Elastix:5060;user=phone>
Call-ID: 1353ff46-c0a80101-0-5@10.100.0.68
CSeq: 2 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Authorization: Digest username="0288", realm="asterisk", nonce="7cdd2046", uri="sip:998564551@Elastix:5060;user=phone", response="c0680dc3cb927cbcbb2d7c9bd2663392", algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:0288@10.100.0.68:5060;user=phone>
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 270
v=0
o=0288 81582519 81582519 IN IP4 10.100.0.68
s=-
c=IN IP4 10.100.0.68
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
<------------->
--- (16 headers 13 lines) ---
Sending to 10.100.0.68 : 5060 (NAT)
Using INVITE request as basis request - 1353ff46-c0a80101-0-5@10.100.0.68
Found peer '0288' for '0288' from 10.100.0.68:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 97
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.100.0.68:41000
Looking for 998564551 in from-internal (domain Elastix)
list_route: hop: <sip:0288@10.100.0.68:5060;user=phone>
<--- Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK252531915214765865;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4dcd9b6
To: <sip:998564551@Elastix:5060;user=phone>
Call-ID: 1353ff46-c0a80101-0-5@10.100.0.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:998564551@10.100.0.151>
Content-Length: 0
<------------>
-- Executing [998564551@from-internal:1] [1;36mMacro[0m("[1;35mSIP/0288-00000006[0m", "[1;35muser-callerid,SKIPTTL,[0m") in new stack
-- Executing [s@macro-user-callerid:1] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mAMPUSER=0288[0m") in new stack
[KElastix*CLI> [0K -- Executing [s@macro-user-callerid:2] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?report[0m") in new stack
-- Executing [s@macro-user-callerid:3] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?Set(REALCALLERIDNUM=0288)[0m") in new stack
-- Executing [s@macro-user-callerid:4] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mAMPUSER=0288[0m") in new stack
-- Executing [s@macro-user-callerid:5] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mAMPUSERCIDNAME=jonkoo[0m") in new stack
-- Executing [s@macro-user-callerid:6] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?report[0m") in new stack
-- Executing [s@macro-user-callerid:7] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mAMPUSERCID=0288[0m") in new stack
-- Executing [s@macro-user-callerid:8] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mCALLERID(all)="jonkoo" <0288>[0m") in new stack
-- Executing [s@macro-user-callerid:9] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Set(CHANNEL(language)=)[0m") in new stack
-- Executing [s@macro-user-callerid:10] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?continue[0m") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] [1;36mNoOp[0m("[1;35mSIP/0288-00000006[0m", "[1;35mUsing CallerID "jonkoo" <0288>[0m") in new stack
-- Executing [998564551@from-internal:2] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35m_NODEST=[0m") in new stack
-- Executing [998564551@from-internal:3] [1;36mMacro[0m("[1;35mSIP/0288-00000006[0m", "[1;35mrecord-enable,0288,OUT,[0m") in new stack
-- Executing [s@macro-record-enable:1] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?check[0m") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?MacroExit()[0m") in new stack
-- Executing [s@macro-record-enable:5] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Group:OUT[0m") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?IN[0m") in new stack
-- Executing [s@macro-record-enable:16] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?MacroExit()[0m") in new stack
-- Executing [998564551@from-internal:4] [1;36mMacro[0m("[1;35mSIP/0288-00000006[0m", "[1;35mdialout-trunk,1,98564551,,[0m") in new stack
-- Executing [s@macro-dialout-trunk:1] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mDIAL_TRUNK=1[0m") in new stack
-- Executing [s@macro-dialout-trunk:2] [1;36mGosubIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?sub-pincheck,s,1[0m") in new stack
-- Executing [s@macro-dialout-trunk:3] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?disabletrunk,1[0m") in new stack
-- Executing [s@macro-dialout-trunk:4] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mDIAL_NUMBER=98564551[0m") in new stack
-- Executing [s@macro-dialout-trunk:5] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mDIAL_TRUNK_OPTIONS=tr[0m") in new stack
-- Executing [s@macro-dialout-trunk:6] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mOUTBOUND_GROUP=OUT_1[0m") in new stack
-- Executing [s@macro-dialout-trunk:7] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?nomax[0m") in new stack
-- Executing [s@macro-dialout-trunk:8] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?chanfull[0m") in new stack
-- Executing [s@macro-dialout-trunk:9] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?skipoutcid[0m") in new stack
-- Executing [s@macro-dialout-trunk:10] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mDIAL_TRUNK_OPTIONS=[0m") in new stack
-- Executing [s@macro-dialout-trunk:11] [1;36mMacro[0m("[1;35mSIP/0288-00000006[0m", "[1;35moutbound-callerid,1[0m") in new stack
-- Executing [s@macro-outbound-callerid:1] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Set(CALLERPRES()=)[0m") in new stack
-- Executing [s@macro-outbound-callerid:2] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Set(REALCALLERIDNUM=0288)[0m") in new stack
-- Executing [s@macro-outbound-callerid:3] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?normcid[0m") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mUSEROUTCID=[0m") in new stack
-- Executing [s@macro-outbound-callerid:7] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mEMERGENCYCID=[0m") in new stack
-- Executing [s@macro-outbound-callerid:8] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mTRUNKOUTCID=[0m") in new stack
-- Executing [s@macro-outbound-callerid:9] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?trunkcid[0m") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Set(CALLERID(all)=)[0m") in new stack
-- Executing [s@macro-outbound-callerid:13] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Set(CALLERID(all)=)[0m") in new stack
-- Executing [s@macro-outbound-callerid:14] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Set(CALLERID(all)=)[0m") in new stack
-- Executing [s@macro-outbound-callerid:15] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Set(CALLERPRES()=prohib_passed_screen)[0m") in new stack
-- Executing [s@macro-dialout-trunk:12] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?AGI(fixlocalprefix)[0m") in new stack
-- Executing [s@macro-dialout-trunk:13] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mOUTNUM=98564551[0m") in new stack
-- Executing [s@macro-dialout-trunk:14] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mcustom=SIP/alcatel[0m") in new stack
-- Executing [s@macro-dialout-trunk:15] [1;36mExecIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))[0m") in new stack
-- Executing [s@macro-dialout-trunk:16] [1;36mMacro[0m("[1;35mSIP/0288-00000006[0m", "[1;35mdialout-trunk-predial-hook,[0m") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] [1;36mMacroExit[0m("[1;35mSIP/0288-00000006[0m", "[1;35m[0m") in new stack
-- Executing [s@macro-dialout-trunk:17] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?bypass,1[0m") in new stack
-- Executing [s@macro-dialout-trunk:18] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?customtrunk[0m") in new stack
-- Executing [s@macro-dialout-trunk:19] [1;36mDial[0m("[1;35mSIP/0288-00000006[0m", "[1;35mSIP/alcatel/98564551,300,[0m") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 10.100.0.151 port 17636
A[KElastix*CLI> [0Kdding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.100.0.1:5060:
INVITE sip:98564551@10.100.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK393aad72;rport
Max-Forwards: 70
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as7a5629e1
To: <sip:98564551@10.100.0.1>
Contact: <sip:0288@10.100.0.151>
Call-ID: 21a040921b5d9f5e3a2f653d7390b313@10.100.0.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Tue, 13 Sep 2011 00:46:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 1970262818 1970262818 IN IP4 10.100.0.151
s=Asterisk PBX 1.6.2.13
c=IN IP4 10.100.0.151
t=0 0
m=audio 17636 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv
---
-- Called alcatel/98564551
[KElastix*CLI> [0K
<--- SIP read from UDP:10.100.0.1:5060 --->
SIP/2.0 100 Trying
To: <sip:98564551@10.100.0.1>
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as7a5629e1
Call-ID: 21a040921b5d9f5e3a2f653d7390b313@10.100.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.100.0.151:5060;received=10.100.0.151;branch=z9hG4bK393aad72;rport=5060
Content-Length: 0
<------------->
[KElastix*CLI> [0K--- (7 headers 0 lines) ---
[KElastix*CLI> [0K
<--- SIP read from UDP:10.100.0.1:5060 --->
SIP/2.0 488 Not Acceptable Here
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
User-Agent: OmniPCX Enterprise R9.0 h1.301.37
To: <sip:98564551@10.100.0.1>;tag=97b75ac251a78650503c8f934ad98512
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as7a5629e1
Call-ID: 21a040921b5d9f5e3a2f653d7390b313@10.100.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.100.0.151:5060;received=10.100.0.151;branch=z9hG4bK393aad72;rport=5060
Content-Length: 0
[KElastix*CLI> [0K
<------------->
[KElastix*CLI> [0K--- (9 headers 0 lines) ---
Transmitting (no NAT) to 10.100.0.1:5060:
ACK sip:98564551@10.100.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK393aad72;rport
Max-Forwards: 70
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as7a5629e1
To: <sip:98564551@10.100.0.1>;tag=97b75ac251a78650503c8f934ad98512
Contact: <sip:0288@10.100.0.151>
Call-ID: 21a040921b5d9f5e3a2f653d7390b313@10.100.0.151
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
---
-- SIP/alcatel-00000007 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] [1;36mNoOp[0m("[1;35mSIP/0288-00000006[0m", "[1;35mDial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 58[0m") in new stack
-- Executing [s@macro-dialout-trunk:21] [1;36mGoto[0m("[1;35mSIP/0288-00000006[0m", "[1;35ms-CONGESTION,1[0m") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mRC=58[0m") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] [1;36mGoto[0m("[1;35mSIP/0288-00000006[0m", "[1;35m58,1[0m") in new stack
-- Goto (macro-dialout-trunk,58,1)
-- Executing [58@macro-dialout-trunk:1] [1;36mGoto[0m("[1;35mSIP/0288-00000006[0m", "[1;35mcontinue,1[0m") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?noreport[0m") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] [1;36mNoOp[0m("[1;35mSIP/0288-00000006[0m", "[1;35mTRUNK Dial failed due to CONGESTION HANGUPCAUSE: 58 - failing through to other trunks[0m") in new stack
-- Executing [continue@macro-dialout-trunk:4] [1;36mSet[0m("[1;35mSIP/0288-00000006[0m", "[1;35mCALLERID(number)=0288[0m") in new stack
-- Executing [998564551@from-internal:5] [1;36mMacro[0m("[1;35mSIP/0288-00000006[0m", "[1;35moutisbusy,[0m") in new stack
-- Executing [s@macro-outisbusy:1] [1;36mProgress[0m("[1;35mSIP/0288-00000006[0m", "[1;35m[0m") in new stack
Audio is at 10.100.0.151 port 12832
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK252531915214765865;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4dcd9b6
To: <sip:998564551@Elastix:5060;user=phone>;tag=as39aadcde
Call-ID: 1353ff46-c0a80101-0-5@10.100.0.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:998564551@10.100.0.151>
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 2080498336 2080498336 IN IP4 10.100.0.151
s=Asterisk PBX 1.6.2.13
c=IN IP4 10.100.0.151
t=0 0
m=audio 12832 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv
<------------>
-- Executing [s@macro-outisbusy:2] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?emergency,1[0m") in new stack
-- Executing [s@macro-outisbusy:3] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m0?intracompany,1[0m") in new stack
-- Executing [s@macro-outisbusy:4] [1;36mPlayback[0m("[1;35mSIP/0288-00000006[0m", "[1;35mall-circuits-busy-now&pls-try-call-later, noanswer[0m") in new stack
-- <SIP/0288-00000006> Playing 'all-circuits-busy-now.gsm' (language 'en')
[KElastix*CLI> [0KReally destroying SIP dialog '21a040921b5d9f5e3a2f653d7390b313@10.100.0.151' Method: INVITE
[KElastix*CLI> [0K -- <SIP/0288-00000006> Playing 'pls-try-call-later.gsm' (language 'en')
[KElastix*CLI> [0K -- Executing [s@macro-outisbusy:5] [1;36mCongestion[0m("[1;35mSIP/0288-00000006[0m", "[1;35m20[0m") in new stack
[KElastix*CLI> [0K
<--- Reliably Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK252531915214765865;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4dcd9b6
To: <sip:998564551@Elastix:5060;user=phone>;tag=as39aadcde
Call-ID: 1353ff46-c0a80101-0-5@10.100.0.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0
<------------>
[KElastix*CLI> [0K == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/0288-00000006' in macro 'outisbusy'
[KElastix*CLI> [0K == Spawn extension (from-internal, 998564551, 5) exited non-zero on 'SIP/0288-00000006'
[KElastix*CLI> [0K -- Executing [h@from-internal:1] [1;36mMacro[0m("[1;35mSIP/0288-00000006[0m", "[1;35mhangupcall[0m") in new stack
[KElastix*CLI> [0K -- Executing [s@macro-hangupcall:1] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?noautomon[0m") in new stack
[KElastix*CLI> [0K -- Goto (macro-hangupcall,s,3)
[KElastix*CLI> [0K -- Executing [s@macro-hangupcall:3] [1;36mNoOp[0m("[1;35mSIP/0288-00000006[0m", "[1;35mTOUCH_MONITOR_OUTPUT=[0m") in new stack
[KElastix*CLI> [0K -- Executing [s@macro-hangupcall:4] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?skiprg[0m") in new stack
[KElastix*CLI> [0K -- Goto (macro-hangupcall,s,7)
[KElastix*CLI> [0K -- Executing [s@macro-hangupcall:7] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?skipblkvm[0m") in new stack
[KElastix*CLI> [0K -- Goto (macro-hangupcall,s,10)
[KElastix*CLI> [0K -- Executing [s@macro-hangupcall:10] [1;36mGotoIf[0m("[1;35mSIP/0288-00000006[0m", "[1;35m1?theend[0m") in new stack
[KElastix*CLI> [0K -- Goto (macro-hangupcall,s,12)
[KElastix*CLI> [0K -- Executing [s@macro-hangupcall:12] [1;36mHangup[0m("[1;35mSIP/0288-00000006[0m", "[1;35m[0m") in new stack
[KElastix*CLI> [0K == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/0288-00000006' in macro 'hangupcall'
[KElastix*CLI> [0K == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/0288-00000006'
[KElastix*CLI> [0K
<--- SIP read from UDP:10.100.0.68:5060 --->
ACK sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK252531915214765865
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4dcd9b6
To: <sip:998564551@Elastix:5060;user=phone>;tag=as39aadcde
Call-ID: 1353ff46-c0a80101-0-5@10.100.0.68
CSeq: 2 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Authorization: Digest username="0288", realm="asterisk", nonce="7cdd2046", uri="sip:998564551@Elastix:5060;user=phone", response="809a960b05b082450fcfe70670c86103", algorithm=MD5
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
[KElastix*CLI> If I'm using ISDN trunk, the result is I can call out, the target phone is ring but my elastix extension suddenly dropped and my trunk status on Alcatel is stuck in Busy mode.
here is the traces
Code: Select all
[KElastix*CLI> sip set debug on
Elastix*CLI>
[0KSIP Debugging re-enabled
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.68:5060 --->
INVITE sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK8631929769758603653
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 1 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:0288@10.100.0.68:5060;user=phone>
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 270
v=0
o=0288 81686075 81686075 IN IP4 10.100.0.68
s=-
c=IN IP4 10.100.0.68
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
<------------->
--- (15 headers 13 lines) ---
[KElastix*CLI>
[0K == Using SIP RTP TOS bits 184
[KElastix*CLI>
[0K == Using SIP RTP CoS mark 5
[KElastix*CLI>
[0KSending to 10.100.0.68 : 5060 (no NAT)
[KElastix*CLI>
[0KUsing INVITE request as basis request - 18326d80-c0a80101-0-6@10.100.0.68
[KElastix*CLI>
[0KFound peer '0288' for '0288' from 10.100.0.68:5060
[KElastix*CLI>
[0K
<--- Reliably Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK8631929769758603653;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>;tag=as5a89f322
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2fdc981b"
Content-Length: 0
<------------>
[KElastix*CLI>
[0KScheduling destruction of SIP dialog '18326d80-c0a80101-0-6@10.100.0.68' in 6400 ms (Method: INVITE)
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.68:5060 --->
ACK sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK8631929769758603653
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>;tag=as5a89f322
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 1 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Content-Length: 0
<------------->
[KElastix*CLI>
[0K--- (10 headers 0 lines) ---
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.68:5060 --->
INVITE sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142030814203754754
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 2 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Authorization: Digest username="0288", realm="asterisk", nonce="2fdc981b", uri="sip:998564551@Elastix:5060;user=phone", response="363f9767666552d90d56af283c3b2a33", algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:0288@10.100.0.68:5060;user=phone>
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 270
v=0
o=0288 81686075 81686075 IN IP4 10.100.0.68
s=-
c=IN IP4 10.100.0.68
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
<------------->
--- (16 headers 13 lines) ---
Sending to 10.100.0.68 : 5060 (NAT)
Using INVITE request as basis request - 18326d80-c0a80101-0-6@10.100.0.68
Found peer '0288' for '0288' from 10.100.0.68:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 97
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.100.0.68:41000
Looking for 998564551 in from-internal (domain Elastix)
list_route: hop: <sip:0288@10.100.0.68:5060;user=phone>
<--- Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142030814203754754;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
[KElastix*CLI>
[0KAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:998564551@10.100.0.151>
Content-Length: 0
<------------>
[KElastix*CLI>
[0K -- Executing [998564551@from-internal:1] [1;36mMacro[0m("[1;35mSIP/0288-00000008[0m", "[1;35muser-callerid,SKIPTTL,[0m") in new stack
-- Executing [s@macro-user-callerid:1] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mAMPUSER=0288[0m") in new stack
-- Executing [s@macro-user-callerid:2] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?report[0m") in new stack
-- Executing [s@macro-user-callerid:3] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?Set(REALCALLERIDNUM=0288)[0m") in new stack
-- Executing [s@macro-user-callerid:4] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mAMPUSER=0288[0m") in new stack
-- Executing [s@macro-user-callerid:5] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mAMPUSERCIDNAME=jonkoo[0m") in new stack
-- Executing [s@macro-user-callerid:6] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?report[0m") in new stack
[KElastix*CLI>
[0K -- Executing [s@macro-user-callerid:7] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mAMPUSERCID=0288[0m") in new stack
-- Executing [s@macro-user-callerid:8] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mCALLERID(all)="jonkoo" <0288>[0m") in new stack
-- Executing [s@macro-user-callerid:9] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Set(CHANNEL(language)=)[0m") in new stack
-- Executing [s@macro-user-callerid:10] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?continue[0m") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] [1;36mNoOp[0m("[1;35mSIP/0288-00000008[0m", "[1;35mUsing CallerID "jonkoo" <0288>[0m") in new stack
-- Executing [998564551@from-internal:2] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35m_NODEST=[0m") in new stack
-- Executing [998564551@from-internal:3] [1;36mMacro[0m("[1;35mSIP/0288-00000008[0m", "[1;35mrecord-enable,0288,OUT,[0m") in new stack
-- Executing [s@macro-record-enable:1] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?check[0m") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?MacroExit()[0m") in new stack
-- Executing [s@macro-record-enable:5] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Group:OUT[0m") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?IN[0m") in new stack
-- Executing [s@macro-record-enable:16] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?MacroExit()[0m") in new stack
-- Executing [998564551@from-internal:4] [1;36mMacro[0m("[1;35mSIP/0288-00000008[0m", "[1;35mdialout-trunk,1,98564551,,[0m") in new stack
-- Executing [s@macro-dialout-trunk:1] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mDIAL_TRUNK=1[0m") in new stack
[KElastix*CLI>
[0K -- Executing [s@macro-dialout-trunk:2] [1;36mGosubIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?sub-pincheck,s,1[0m") in new stack
-- Executing [s@macro-dialout-trunk:3] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?disabletrunk,1[0m") in new stack
-- Executing [s@macro-dialout-trunk:4] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mDIAL_NUMBER=98564551[0m") in new stack
-- Executing [s@macro-dialout-trunk:5] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mDIAL_TRUNK_OPTIONS=tr[0m") in new stack
-- Executing [s@macro-dialout-trunk:6] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mOUTBOUND_GROUP=OUT_1[0m") in new stack
-- Executing [s@macro-dialout-trunk:7] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?nomax[0m") in new stack
-- Executing [s@macro-dialout-trunk:8] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?chanfull[0m") in new stack
-- Executing [s@macro-dialout-trunk:9] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?skipoutcid[0m") in new stack
-- Executing [s@macro-dialout-trunk:10] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mDIAL_TRUNK_OPTIONS=[0m") in new stack
-- Executing [s@macro-dialout-trunk:11] [1;36mMacro[0m("[1;35mSIP/0288-00000008[0m", "[1;35moutbound-callerid,1[0m") in new stack
-- Executing [s@macro-outbound-callerid:1] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Set(CALLERPRES()=)[0m") in new stack
-- Executing [s@macro-outbound-callerid:2] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Set(REALCALLERIDNUM=0288)[0m") in new stack
-- Executing [s@macro-outbound-callerid:3] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?normcid[0m") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mUSEROUTCID=[0m") in new stack
-- Executing [s@macro-outbound-callerid:7] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mEMERGENCYCID=[0m") in new stack
-- Executing [s@macro-outbound-callerid:8] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mTRUNKOUTCID=[0m") in new stack
-- Executing [s@macro-outbound-callerid:9] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?trunkcid[0m") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Set(CALLERID(all)=)[0m") in new stack
-- Executing [s@macro-outbound-callerid:13] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Set(CALLERID(all)=)[0m") in new stack
-- Executing [s@macro-outbound-callerid:14] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Set(CALLERID(all)=)[0m") in new stack
-- Executing [s@macro-outbound-callerid:15] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Set(CALLERPRES()=prohib_passed_screen)[0m") in new stack
-- Executing [s@macro-dialout-trunk:12] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?AGI(fixlocalprefix)[0m") in new stack
-- Executing [s@macro-dialout-trunk:13] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mOUTNUM=98564551[0m") in new stack
-- Executing [s@macro-dialout-trunk:14] [1;36mSet[0m("[1;35mSIP/0288-00000008[0m", "[1;35mcustom=SIP/alcatel[0m") in new stack
-- Executing [s@macro-dialout-trunk:15] [1;36mExecIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))[0m") in new stack
-- Executing [s@macro-dialout-trunk:16] [1;36mMacro[0m("[1;35mSIP/0288-00000008[0m", "[1;35mdialout-trunk-predial-hook,[0m") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] [1;36mMacroExit[0m("[1;35mSIP/0288-00000008[0m", "[1;35m[0m") in new stack
-- Executing [s@macro-dialout-trunk:17] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?bypass,1[0m") in new stack
-- Executing [s@macro-dialout-trunk:18] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m0?customtrunk[0m") in new stack
-- Executing [s@macro-dialout-trunk:19] [1;36mDial[0m("[1;35mSIP/0288-00000008[0m", "[1;35mSIP/alcatel/98564551,300,[0m") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 10.100.0.151 port 14254
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.100.0.1:5060:
INVITE sip:98564551@10.100.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK55a64fcd;rport
Max-Forwards: 70
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as52480f93
To: <sip:98564551@10.100.0.1>
Contact: <sip:0288@10.100.0.151>
Call-ID: 291db8172479340f1f639eec3ff37478@10.100.0.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Tue, 13 Sep 2011 00:50:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 930504372 930504372 IN IP4 10.100.0.151
s=Asterisk PBX 1.6.2.13
c=IN IP4 10.100.0.151
t=0 0
m=audio 14254 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv
---
-- Called alcatel/98564551
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.1:5060 --->
SIP/2.0 100 Trying
To: <sip:98564551@10.100.0.1>
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as52480f93
Call-ID: 291db8172479340f1f639eec3ff37478@10.100.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.100.0.151:5060;received=10.100.0.151;branch=z9hG4bK55a64fcd;rport=5060
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.1:5060 --->
SIP/2.0 183 Session Progress
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Contact: sip:10.100.0.1
User-Agent: OmniPCX Enterprise R9.0 h1.301.37
Content-Type: application/sdp
To: <sip:98564551@10.100.0.1>;tag=29a3f80c713cc058bb639f3097023b9e
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as52480f93
Call-ID: 291db8172479340f1f639eec3ff37478@10.100.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.100.0.151:5060;received=10.100.0.151;branch=z9hG4bK55a64fcd;rport=5060
Content-Length: 173
v=0
o=OXE 1315875946 1315875946 IN IP4 10.100.0.1
s=abs
c=IN IP4 10.100.0.2
t=0 0
m=audio 32584 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.100.0.2:32584
[KElastix*CLI>
[0K -- SIP/alcatel-00000009 is making progress passing it to SIP/0288-00000008
[KElastix*CLI>
[0KAudio is at 10.100.0.151 port 16042
[KElastix*CLI>
[0KAdding codec 0x4 (ulaw) to SDP
[KElastix*CLI>
[0KAdding codec 0x8 (alaw) to SDP
[KElastix*CLI>
[0KAdding non-codec 0x1 (telephone-event) to SDP
[KElastix*CLI>
[0K
<--- Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142030814203754754;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>;tag=as5f4ac8a4
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:998564551@10.100.0.151>
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 997377924 997377924 IN IP4 10.100.0.151
s=Asterisk PBX 1.6.2.13
c=IN IP4 10.100.0.151
t=0 0
m=audio 16042 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv
<------------>
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.1:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Contact: sip:10.100.0.1
User-Agent: OmniPCX Enterprise R9.0 h1.301.37
Content-Type: application/sdp
To: <sip:98564551@10.100.0.1>;tag=29a3f80c713cc058bb639f3097023b9e
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as52480f93
Call-ID: 291db8172479340f1f639eec3ff37478@10.100.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.100.0.151:5060;received=10.100.0.151;branch=z9hG4bK55a64fcd;rport=5060
Content-Length: 173
v=0
o=OXE 1315875946 1315875947 IN IP4 10.100.0.1
s=abs
c=IN IP4 10.100.0.3
t=0 0
m=audio 32696 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
[KElastix*CLI>
[0KFound RTP audio format 8
[KElastix*CLI>
[0KFound audio description format PCMA for ID 8
[KElastix*CLI>
[0KCapabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[KElastix*CLI>
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[KElastix*CLI>
[0KPeer audio RTP is at port 10.100.0.3:32696
[KElastix*CLI>
[0K -- SIP/alcatel-00000009 is ringing
<--- Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142030814203754754;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>;tag=as5f4ac8a4
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:998564551@10.100.0.151>
Content-Length: 0
<------------>
-- SIP/alcatel-00000009 is making progress passing it to SIP/0288-00000008
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.68:5060 --->
CANCEL sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142030814203754754
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 2 CANCEL
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Authorization: Digest username="0288", realm="asterisk", nonce="2fdc981b", uri="sip:998564551@Elastix:5060;user=phone", response="52a176abf7ab533c0a42be650ed1bf5a", algorithm=MD5
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 10.100.0.68 : 5060 (NAT)
[KElastix*CLI>
[0K
<--- Reliably Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142030814203754754;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>;tag=as5f4ac8a4
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[KElastix*CLI>
[0K
<--- Transmitting (NAT) to 10.100.0.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142030814203754754;received=10.100.0.68
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>;tag=as5f4ac8a4
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 2 CANCEL
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[KElastix*CLI>
[0KScheduling destruction of SIP dialog '291db8172479340f1f639eec3ff37478@10.100.0.151' in 6400 ms (Method: INVITE)
[KElastix*CLI>
[0KReliably Transmitting (no NAT) to 10.100.0.1:5060:
CANCEL sip:98564551@10.100.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK55a64fcd;rport
Max-Forwards: 70
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as52480f93
To: <sip:98564551@10.100.0.1>
Call-ID: 291db8172479340f1f639eec3ff37478@10.100.0.151
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
---
[KElastix*CLI>
[0KScheduling destruction of SIP dialog '291db8172479340f1f639eec3ff37478@10.100.0.151' in 6400 ms (Method: INVITE)
[KElastix*CLI>
[0K == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/0288-00000008' in macro 'dialout-trunk'
[KElastix*CLI>
[0K == Spawn extension (from-internal, 998564551, 4) exited non-zero on 'SIP/0288-00000008'
-- Executing [h@from-internal:1] [1;36mMacro[0m("[1;35mSIP/0288-00000008[0m", "[1;35mhangupcall[0m") in new stack
-- Executing [s@macro-hangupcall:1] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?noautomon[0m") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] [1;36mNoOp[0m("[1;35mSIP/0288-00000008[0m", "[1;35mTOUCH_MONITOR_OUTPUT=[0m") in new stack
-- Executing [s@macro-hangupcall:4] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?skiprg[0m") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?skipblkvm[0m") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] [1;36mGotoIf[0m("[1;35mSIP/0288-00000008[0m", "[1;35m1?theend[0m") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] [1;36mHangup[0m("[1;35mSIP/0288-00000008[0m", "[1;35m[0m") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/0288-00000008' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/0288-00000008'
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.68:5060 --->
ACK sip:998564551@Elastix:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.68:5060;branch=z9hG4bK9142030814203754754
From: "0288"<sip:0288@Elastix:5060;user=phone>;tag=c0a80101-4de6e3a
To: <sip:998564551@Elastix:5060;user=phone>;tag=as5f4ac8a4
Call-ID: 18326d80-c0a80101-0-6@10.100.0.68
CSeq: 2 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Authorization: Digest username="0288", realm="asterisk", nonce="2fdc981b", uri="sip:998564551@Elastix:5060;user=phone", response="0a51c6dfc875eb490c8bc0a022c7fe41", algorithm=MD5
User-Agent: THOMSON ST2022 hw2 fw3.54 00-18-F6-B5-26-DA
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.1:5060 --->
SIP/2.0 200 OK
Supported: replaces,timer,100rel
User-Agent: OmniPCX Enterprise R9.0 h1.301.37
To: <sip:98564551@10.100.0.1>
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as52480f93
Call-ID: 291db8172479340f1f639eec3ff37478@10.100.0.151
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 10.100.0.151:5060;received=10.100.0.151;branch=z9hG4bK55a64fcd;rport=5060
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.1:5060 --->
SIP/2.0 487 Request Terminated
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
User-Agent: OmniPCX Enterprise R9.0 h1.301.37
To: <sip:98564551@10.100.0.1>;tag=29a3f80c713cc058bb639f3097023b9e
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as52480f93
Call-ID: 291db8172479340f1f639eec3ff37478@10.100.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.100.0.151:5060;received=10.100.0.151;branch=z9hG4bK55a64fcd;rport=5060
Content-Length: 0
[KElastix*CLI>
[0K
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 10.100.0.1:5060:
ACK sip:98564551@10.100.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK55a64fcd;rport
Max-Forwards: 70
From: "jonkoo" <sip:0288@10.100.0.151>;tag=as52480f93
To: <sip:98564551@10.100.0.1>;tag=29a3f80c713cc058bb639f3097023b9e
Contact: <sip:0288@10.100.0.151>
Call-ID: 291db8172479340f1f639eec3ff37478@10.100.0.151
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
---
Really destroying SIP dialog '291db8172479340f1f639eec3ff37478@10.100.0.151' Method: INVITE
[KElastix*CLI>
[0KReliably Transmitting (NAT) to 10.100.0.68:5060:
OPTIONS sip:0288@10.100.0.68:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK4c23141c;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.100.0.151>;tag=as1fe9d4ab
To: <sip:0288@10.100.0.68:5060;user=phone>
Contact: <sip:Unknown@10.100.0.151>
Call-ID: 18e51d4f42c9c519324ffbc92852cff7@10.100.0.151
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Tue, 13 Sep 2011 00:50:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
[KElastix*CLI>
[0K
<--- SIP read from UDP:10.100.0.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.151:5060;branch=z9hG4bK4c23141c;rport
From: "Unknown"<sip:Unknown@10.100.0.151>;tag=as1fe9d4ab
To: <sip:0288@10.100.0.68:5060;user=phone>;tag=c0a80101-4de9e71
Call-ID: 18e51d4f42c9c519324ffbc92852cff7@10.100.0.151
CSeq: 102 OPTIONS
Contact: <sip:0288@10.100.0.68:5060;user=phone>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Supported: timer, replaces
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 270
v=0
o=0288 81698417 81698417 IN IP4 10.100.0.68
s=-
c=IN IP4 10.100.0.68
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
Really destroying SIP dialog '18e51d4f42c9c519324ffbc92852cff7@10.100.0.151' Method: OPTIONS
Re: connecting elastix (freebpx based, witch is based on ast
hi bro, can post your configuration at here for share.. may be I could help younperalta wrote:i can call from elastix to oxe but 1 to 10 calls could be established the other gives an infernal beep !!
