SIP trunk tutorial

[smith_user]

Hi there

Before someone jumps down my throat and says search the forum, i have read this forum through and through looking for examples of detailed configuation tutorial of how to connect an OXO to Asterisk but have found nothing that gives full details, just bits and pieces all over the place and im trying to connect the dots. Im pretty new to both Asterisk and Alcatel so its a pretty steep learning curve and when you dont know how either side should be configured there is a lot more to go wrong and its much more difficult to diagnose and troubleshoot.

Anyway i have got an asterisk box with some sip phones (about 5) connected to it, which each of those asterisk extensions can dial between each other. I also have an Alcatel OXO 9.1 system with an E1 (and about 30 handsets) out to the PSTN. I would like to be able to connect the alcatel and asterisk boxes, i have read numerous posts saying they have them connected but only calls going one way will work, but nothing really explaining whats going on between the two as in extension numbers and routing etc. I dont know a heck of a lot about the ARS on the alcatel system, so im trying to read up on that. Anyway, im going to keep reading but hopefully someone can help me out, im happy to make a fully documented tutorial once its done.

Ive been trying to use these :-
http://www.kic-ftp1.com/alliance_hotlin ... cew_en.pdf
https://wiki.2n.cz/pages/viewpage.action?pageId=7353370
http://www.iconvoicenetworks.com/assets ... _Guide.pdf
http://www.cadvision.com/blanchas/Aster ... trunk.html
http://tyler.anairo.com/?id=3.1.0

But i just cant get them to talk.

Running :-
Asterisk (Ver. 11.2.1)
Asterisk Now - v3.0.0 32bit

Cheers

[tot3nkopf]

If you use the search function you can get smth like (and many more):
viewtopic.php?f=227&t=18497

[smith_user]

Thanks for that tot3nkopf

I have been through this post, and a quite a few others like it, showing configuration screen shots of the setups that some people have managed to get working. I have tried some of these configurations, including this one, and no luck. Now what i was asking for was a detailed tutorial, someone describing why you should change certain settings and why not. I dont want to sound ungrateful especially since you have been very helpful around the forum from what i have read of your posts, but i was just wondering if there was an easier way for total newbies to connect and get going without wading through loads of forum posts and hugely technical manuals. Ill keep going and when i figure it out ill try to write a tut for it.

Cheers

[cavagnaro]

No... that won't happen as on SIP every configuration is because of your environment or provider. What you should do is read the documentation completely and understand what does each parameter does. Read and study some SIP stuff to also understand tthe exchange messages.
There is no magic cooking book on SIP nor OXO/OXE.

[smith_user]

Ok so i understand that you need to have a SIP trunk between the asterisk box and the alcatel box. The extension numbers setup on the alcatel box are 1000-1999. The alcatel extensions are all 8xx. I have created a sip trunk (though i dont think its configured correctly as i dont think they are connecting). I have added an outbound routes such that any number 8XX dialled on an asterisk sip phone will be sent to the alcatel sip trunk and from there hopefully the alcatel system will route it appropriately. In reading the expert documentation i havent really come across a good explaination of the ARS, the gateway and how its all connected ie how i route calls from asterisk extensions to alcatel extensions or send them out to the PSTN.
alcatel = 192.168.0.99
asterisk = 192.168.0.220

When the SIP trunk tries to register i get the following:-

<--- SIP read from UDP:192.168.0.99:10240 --->
REGISTER sip:192.168.0.220 SIP/2.0
CSeq: 45748971 REGISTER
Expires: 3600
User-Agent: OXO_GW_910/037.001
To: sip:alize@192.168.0.220
From: sip:alize@192.168.0.220;tag=930b65951e625490d7ac0571337eca52
Contact: <sip:alize@192.168.0.99:10240;transport=UDP>;audio;class="business, personal";duplex="full, half";mobility="fixed";description="<OmniPCX Office>";methods="ACK, INVITE, CANCEL, BYE, PRACK, REFER, NOTIFY, OPTIONS, UPDATE";extensions="100rel, timer, from-change";schemes="sip"
Call-ID: 72aab48a7474fcfbd0628b7f12bef5e6@192.168.0.99
Via: SIP/2.0/UDP 192.168.0.99:10240;rport;branch=z9hG4bK3b34aa7d8509246bd82d87a13678892d
Max-Forwards: 70
Content-Length: 0

<------------->
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c: --- (11 headers 0 lines) ---
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c: Sending to 192.168.0.99:10240 (NAT)
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c:
<--- Transmitting (NAT) to 192.168.0.99:10240 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.99:10240;branch=z9hG4bK3b34aa7d8509246bd82d87a13678892d;received=192.168.0.99;rport=10240
From: sip:alize@192.168.0.220;tag=930b65951e625490d7ac0571337eca52
To: sip:alize@192.168.0.220;tag=as0fcc0d4f
Call-ID: 72aab48a7474fcfbd0628b7f12bef5e6@192.168.0.99
CSeq: 45748971 REGISTER
Server: -2.11.0beta2(11.2.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29576619"
Content-Length: 0


<------------>
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c: Scheduling destruction of SIP dialog '72aab48a7474fcfbd0628b7f12bef5e6@192.168.0.99' in 32000 ms (Method: REGISTER)
[2013-12-11 15:56:28] NOTICE[3864] chan_sip.c: Registration from 'sip:alize@192.168.0.220' failed for '192.168.0.99:10240' - No matching peer found
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c: Scheduling destruction of SIP dialog '72aab48a7474fcfbd0628b7f12bef5e6@192.168.0.99' in 32000 ms (Method: REGISTER)
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c:
<--- SIP read from UDP:192.168.0.99:10240 --->
REGISTER sip:192.168.0.220 SIP/2.0
CSeq: 45748972 REGISTER
Expires: 3600
User-Agent: OXO_GW_910/037.001
To: sip:alize@192.168.0.220
From: sip:alize@192.168.0.220;tag=930b65951e625490d7ac0571337eca52
Contact: <sip:alize@192.168.0.99:10240;transport=UDP>;audio;class="business, personal";duplex="full, half";mobility="fixed";description="<OmniPCX Office>";methods="ACK, INVITE, CANCEL, BYE, PRACK, REFER, NOTIFY, OPTIONS, UPDATE";extensions="100rel, timer, from-change";schemes="sip"
Call-ID: 72aab48a7474fcfbd0628b7f12bef5e6@192.168.0.99
Max-Forwards: 70
Authorization: Digest username="",realm="asterisk",nonce="29576619",algorithm=MD5,uri="sip:192.168.0.220",response="b7500069257817cae4faa14665a0323e"
Via: SIP/2.0/UDP 192.168.0.99:10240;rport;branch=z9hG4bKffa12923bbef513aa7a19ff343d3ba5f
Content-Length: 0

<------------->
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c: --- (12 headers 0 lines) ---
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c: Sending to 192.168.0.99:10240 (NAT)
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c:
<--- Transmitting (NAT) to 192.168.0.99:10240 --->
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.0.99:10240;branch=z9hG4bKffa12923bbef513aa7a19ff343d3ba5f;received=192.168.0.99;rport=10240
From: sip:alize@192.168.0.220;tag=930b65951e625490d7ac0571337eca52
To: sip:alize@192.168.0.220;tag=as0fcc0d4f
Call-ID: 72aab48a7474fcfbd0628b7f12bef5e6@192.168.0.99
CSeq: 45748972 REGISTER
Server: -2.11.0beta2(11.2.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2013-12-11 15:56:28] NOTICE[3864] chan_sip.c: Registration from 'sip:alize@192.168.0.220' failed for '192.168.0.99:10240' - No matching peer found
[2013-12-11 15:56:28] VERBOSE[3864] chan_sip.c: Scheduling destruction of SIP dialog '72aab48a7474fcfbd0628b7f12bef5e6@192.168.0.99' in 32000 ms (Method: REGISTER)






anyway when i dial an alcatel extension from an asterisk extension i get this :-

INVITE sip:0808@192.168.0.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK59893b51;rport
Max-Forwards: 70
From: <sip:97554888@192.168.0.220>;tag=as38b533ce
To: <sip:0808@192.168.0.99>
Contact: <sip:97554888@192.168.0.220:5060>
Call-ID: 5cb0356f605eb99324116ee10466d031@192.168.0.220:5060
CSeq: 102 INVITE
User-Agent: -2.11.0beta2(11.2.1)
Date: Wed, 11 Dec 2013 04:55:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 984673639 984673639 IN IP4 192.168.0.220
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.220
t=0 0
m=audio 14502 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[2013-12-11 15:55:49] VERBOSE[14277][C-0000003a] app_dial.c: -- Called SIP/Alcatel/0808
[2013-12-11 15:55:49] VERBOSE[3864] chan_sip.c: Retransmitting #1 (NAT) to 192.168.0.99:5060:
INVITE sip:0808@192.168.0.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK59893b51;rport
Max-Forwards: 70
From: <sip:97554888@192.168.0.220>;tag=as38b533ce
To: <sip:0808@192.168.0.99>
Contact: <sip:97554888@192.168.0.220:5060>
Call-ID: 5cb0356f605eb99324116ee10466d031@192.168.0.220:5060
CSeq: 102 INVITE
User-Agent: -2.11.0beta2(11.2.1)
Date: Wed, 11 Dec 2013 04:55:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 984673639 984673639 IN IP4 192.168.0.220
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.220
t=0 0
m=audio 14502 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

I guess this means that it keeps retrying to send the request because the alcatel box is not responding through the SIP trunk.

This is my setup

siptrunk_config.jpg

outbound.jpg

gateway_params.jpg

ARS_settings.jpg

[smith_user]

whoops just realised i had a 0 in the sip trunk config which was putting a 0 in front of calls to extensions, that didnt fix my problem though.

[tot3nkopf]

SIP/2.0 401 Unauthorized

What does this tell you?

[smith_user]

guess it means that i cant authenticate, which is strange because the username and password are correct? So i turned off login ACL and cracked up the verbosity of sip debug and tried to dial again
---
[2013-12-13 08:18:50] DEBUG[23689] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.0.99:5060
[2013-12-13 08:18:51] DEBUG[23689] chan_sip.c: Header 0 [ 0]:
[2013-12-13 08:18:51] DEBUG[23689] chan_sip.c: SIP TIMER: Rescheduling retransmission #6650 (2) INVITE - 5
[2013-12-13 08:18:51] DEBUG[23689] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #6650))
[2013-12-13 08:18:51] VERBOSE[23689] chan_sip.c: Retransmitting #2 (NAT) to 192.168.0.99:5060:
INVITE sip:808@192.168.0.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK3e737e33;rport
Max-Forwards: 70
From: "Jakes Desktop" <sip:1010@192.168.0.220>;tag=as412b18f3
To: <sip:808@192.168.0.99>
Contact: <sip:1010@192.168.0.220:5060>
Call-ID: 4ca9e8f01f77a8326f3f07495183fbf0@192.168.0.220:5060
CSeq: 102 INVITE
User-Agent: -2.11.0beta2(11.2.1)
Date: Thu, 12 Dec 2013 21:18:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1633887023 1633887023 IN IP4 192.168.0.220
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.220
t=0 0
m=audio 12108 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[2013-12-13 08:18:51] DEBUG[23689] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.0.99:5060
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Allocating new SIP dialog for 022e3a3d53ab236b14689b355bbac6c1@199.101.28.130:5060 - OPTIONS (No RTP)
[2013-12-13 08:18:52] DEBUG[23689] acl.c: For destination '192.168.0.233', our source address is '192.168.0.220'.
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.0.220:5060
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Initializing initreq for method OPTIONS - callid 143e0ca459aa7a7731a3666943ff5bda@192.168.0.220:5060
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 0 [ 47]: OPTIONS sip:1004@192.168.0.233:54380;ob SIP/2.0
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK3384fb08;rport
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 3 [ 58]: From: "Unknown" <sip:Unknown@192.168.0.220>;tag=as3d4242d5
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 4 [ 37]: To: <sip:1004@192.168.0.233:54380;ob>
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 5 [ 41]: Contact: <sip:Unknown@192.168.0.220:5060>
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 6 [ 60]: Call-ID: 143e0ca459aa7a7731a3666943ff5bda@192.168.0.220:5060
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 8 [ 32]: User-Agent: -2.11.0beta2(11.2.1)
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 9 [ 35]: Date: Thu, 12 Dec 2013 21:18:52 GMT
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6652
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.0.233:54380
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.0.220:5060;rport=5060;received=192.168.0.220;branch=z9hG4bK3384fb08
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 2 [ 60]: Call-ID: 143e0ca459aa7a7731a3666943ff5bda@192.168.0.220:5060
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 3 [ 58]: From: "Unknown" <sip:Unknown@192.168.0.220>;tag=as3d4242d5
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 4 [ 51]: To: <sip:1004@192.168.0.233;ob>;tag=z9hG4bK3384fb08
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 6 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 7 [177]: Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 9 [ 46]: Allow-Events: presence, message-summary, refer
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 10 [ 37]: User-Agent: CSipSimple_jflte-17/r2330
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 12 [ 19]: Content-Length: 286
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Header 13 [ 0]:
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 0 [ 3]: v=0
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 1 [ 46]: o=- 3595875350 3595875350 IN IP4 192.168.0.233
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 2 [ 9]: s=pjmedia
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 3 [ 5]: t=0 0
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 4 [ 31]: m=audio 4000 RTP/AVP 99 0 8 101
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 5 [ 22]: c=IN IP4 192.168.0.233
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 6 [ 10]: a=sendrecv
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 7 [ 22]: a=rtpmap:99 SILK/24000
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 8 [ 24]: a=fmtp:99 useinbandfec=0
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 10 [ 20]: a=rtpmap:8 PCMA/8000
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-15
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: = Looking for Call ID: 143e0ca459aa7a7731a3666943ff5bda@192.168.0.220:5060 (Checking To) --From tag as3d4242d5 --To-tag z9hG4bK3384fb08
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6652
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Stopping retransmission on '143e0ca459aa7a7731a3666943ff5bda@192.168.0.220:5060' of Request 102: Match Found
[2013-12-13 08:18:52] DEBUG[23689] chan_sip.c: Destroying SIP dialog 143e0ca459aa7a7731a3666943ff5bda@192.168.0.220:5060
[2013-12-13 08:18:53] DEBUG[23689] chan_sip.c: SIP TIMER: Rescheduling retransmission #6650 (3) INVITE - 5
[2013-12-13 08:18:53] DEBUG[23689] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #6650))
[2013-12-13 08:18:53] VERBOSE[23689] chan_sip.c: Retransmitting #3 (NAT) to 192.168.0.99:5060:
INVITE sip:808@192.168.0.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK3e737e33;rport
Max-Forwards: 70
From: "Jakes Desktop" <sip:1010@192.168.0.220>;tag=as412b18f3
To: <sip:808@192.168.0.99>
Contact: <sip:1010@192.168.0.220:5060>
Call-ID: 4ca9e8f01f77a8326f3f07495183fbf0@192.168.0.220:5060
CSeq: 102 INVITE
User-Agent: -2.11.0beta2(11.2.1)
Date: Thu, 12 Dec 2013 21:18:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1633887023 1633887023 IN IP4 192.168.0.220
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.220
t=0 0
m=audio 12108 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

sip_reg.jpg

so now its not responding at all? I did a sip show registry on the freepbx box and it says not connected. I also ran a sip dump on the asterisk box and had a look at it through wireshark and it looks as though the request is being sent to the alcatel box, but nothing is happening with it.

[smith_user]

im doing a dump from the alcatel box and im getting errors saying icmp destination host unreachable and i have disabled the firewall?

icmp.jpg

[tot3nkopf]

Show us ARS setting in OXO with IP parameters activated (right click and tick IP parameters).