SIP Trunk transfer error

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marcin99
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Posts: 6
Joined: 20 Jan 2016 04:29

SIP Trunk transfer error

Post by marcin99 » 31 Aug 2016 03:55

hello,

I have sip trunk alcatel (OmniPCX Enterprise R11.0.1 k1.520.22.b) - asterisk and everything works well except transfers. When sip call to alcatel and try transfer from sip to other number - work ok.
Problem is when I try transfer sip call from alcatel to alcatel, then alcatel send REFER without INVITE and call ends. It is possible change transfer REFER method to another, is some option for that on alcatel? Or anyone does it work?

Code: Select all

<------------>
    -- SIP/12500-00000140 is ringing

<--- Transmitting (no NAT) to 192.168.1.35:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bKb1362a732295805c0a0e694e2c463742;received=192.168.1.35
From: "Marcin" <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297867 INVITE
Server: FPBX-13.0.124(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12500@192.168.1.31:5060>
Content-Length: 0


<------------>
    -- SIP/12500-00000140 answered SIP/alcatel-0000013f
Audio is at 11796
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.1.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bKb1362a732295805c0a0e694e2c463742;received=192.168.1.35
From: "Marcin" <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297867 INVITE
Server: FPBX-13.0.124(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12500@192.168.1.31:5060>
Content-Type: application/sdp
Content-Length: 249

v=0
o=root 1631329348 1631329348 IN IP4 192.168.1.31
s=Asterisk PBX 13.9.1
c=IN IP4 192.168.1.31
t=0 0
m=audio 11796 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/12500-00000140 joined 'simple_bridge' basic-bridge <30b14ac0-24e9-4b8f-8b7e-a6007fd66d3c>
    -- Channel SIP/alcatel-0000013f joined 'simple_bridge' basic-bridge <30b14ac0-24e9-4b8f-8b7e-a6007fd66d3c>

<--- SIP read from UDP:192.168.1.35:5060 --->
ACK sip:12500@192.168.1.31:5060 SIP/2.0
Contact: <sip:10883@192.168.1.35;transport=UDP>
User-Agent: OmniPCX Enterprise R11.0.1 k1.520.22.b
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
From: "Marcin" <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297867 ACK
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bK44a4e43e41a64a839457d0cbf0fe5e6a
Max-Forwards: 70
Content-Length: 0

<--- SIP read from UDP:192.168.1.35:5060 --->
INVITE sip:12500@192.168.1.31:5060 SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: <sip:10883@192.168.1.35;transport=UDP>
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R11.0.1 k1.520.22.b
Session-Expires: 1800;refresher=uac
Min-SE: 900
Content-Type: application/sdp
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
From: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297868 INVITE
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bKa452302f90d4bcf5d81a98a7b7913263
Max-Forwards: 70
Content-Length: 214

v=0
o=OXE 1472626645 1472626646 IN IP4 192.168.1.35
s=abs
c=IN IP4 192.168.1.28
t=0 0
m=audio 32648 RTP/AVP 8 101
a=sendonly
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:101 telephone-event/8000
<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.1.35:5060 (no NAT)
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|g726|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.28:32648

<--- Transmitting (no NAT) to 192.168.1.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bKa452302f90d4bcf5d81a98a7b7913263;received=192.168.1.35
From: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297868 INVITE
Server: FPBX-13.0.124(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12500@192.168.1.31:5060>
Content-Length: 0


<------------>
Audio is at 11796
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.1.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bKa452302f90d4bcf5d81a98a7b7913263;received=192.168.1.35
From: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297868 INVITE
Server: FPBX-13.0.124(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12500@192.168.1.31:5060>
Content-Type: application/sdp
Content-Length: 249

v=0
o=root 1631329348 1631329349 IN IP4 192.168.1.31
s=Asterisk PBX 13.9.1
c=IN IP4 192.168.1.31
t=0 0
m=audio 11796 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=recvonly

<------------>
    -- Started music on hold, class 'default', on channel 'SIP/12500-00000140'

<--- SIP read from UDP:192.168.1.35:5060 --->
ACK sip:12500@192.168.1.31:5060 SIP/2.0
Contact: <sip:10883@192.168.1.35;transport=UDP>
User-Agent: OmniPCX Enterprise R11.0.1 k1.520.22.b
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
From: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297868 ACK
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bK5b7f6c254d7312874ff73b77edbf6e14
Max-Forwards: 70
Content-Length: 0


<--- SIP read from UDP:192.168.1.35:5060 --->
REFER sip:12500@192.168.1.31:5060 SIP/2.0
Contact: <sip:10883@192.168.1.35;transport=UDP>
Supported: timer,path,100rel
User-Agent: OmniPCX Enterprise R11.0.1 k1.520.22.b
Refer-To: <sip:10885@192.168.1.35;user=phone?REPLACES=ed93f89d919b02568c1fc100a618870e%40192.168.1.35%3bto-tag%3das6d338b8b%3bfrom-tag%3d2b62d24cba1d15e50545de8e489afe86>
Referred-By: <sip:10883@192.168.1.35;user=phone>
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
From: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297869 REFER
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bKa89c11139c2e0cc342eb71f716740490
Max-Forwards: 70
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Call ed93f89d919b02568c1fc100a618870e@192.168.1.35 got a SIP call transfer from caller: (REFER)!
[2016-08-31 08:57:36] WARNING[7721][C-000000b4]: chan_sip.c:18563 get_refer_info: Got an attempt to replace own Call-ID on ed93f89d919b02568c1fc100a618870e@192.168.1.35

<--- Transmitting (no NAT) to 192.168.1.35:5060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bKa89c11139c2e0cc342eb71f716740490;received=192.168.1.35
From: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297869 REFER
Server: FPBX-13.0.124(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12500@192.168.1.31:5060>
Content-Length: 0


<------------>
set_destination: Parsing <sip:10883@192.168.1.35;transport=UDP> for address/port to send to
set_destination: set destination to 192.168.1.35:5060
Reliably Transmitting (no NAT) to 192.168.1.35:5060:
NOTIFY sip:10883@192.168.1.35;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK408173bc
Max-Forwards: 70
From: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
To: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
Contact: <sip:12500@192.168.1.31:5060>
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 102 NOTIFY
User-Agent: FPBX-13.0.124(13.9.1)
Event: refer;id=605297869
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 23

SIP/2.0 404 Not found

---

<--- SIP read from UDP:192.168.1.35:5060 --->
SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: <sip:10883@192.168.1.35;transport=UDP>
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R11.0.1 k1.520.22.b
To: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
From: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK408173bc
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
[2016-08-31 08:57:36] NOTICE[7721][C-000000b4]: chan_sip.c:23978 handle_response_notify: Got OK on REFER Notify message

<--- SIP read from UDP:192.168.1.35:5060 --->
BYE sip:12500@192.168.1.31:5060 SIP/2.0
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R11.0.1 k1.520.22.b
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
From: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297870 BYE
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bK621a29c5f7d39018c6c8fbdb6f26228e
Max-Forwards: 70
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.35:5060 (no NAT)
Scheduling destruction of SIP dialog 'ed93f89d919b02568c1fc100a618870e@192.168.1.35' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.1.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.35;branch=z9hG4bK621a29c5f7d39018c6c8fbdb6f26228e;received=192.168.1.35
From: <sip:10883@192.168.1.35;user=phone>;tag=2b62d24cba1d15e50545de8e489afe86
To: <sip:12500@192.168.1.31;user=phone>;tag=as6d338b8b
Call-ID: ed93f89d919b02568c1fc100a618870e@192.168.1.35
CSeq: 605297870 BYE
Server: FPBX-13.0.124(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

User avatar
dryhouse
Member
Posts: 164
Joined: 06 Apr 2010 06:11
Location: Madrid,Spain

Re: SIP Trunk transfer error

Post by dryhouse » 31 Aug 2016 06:02

Hi marcin99,

Try this:

System > Other System Param > SIP Parameters > Transfer : Refer using single step. This paramter is set by default at True and to obtain Referred-by in such case, it must be checked at False.

Regards.
may the force be with you....

marcin99
Member
Posts: 6
Joined: 20 Jan 2016 04:29

Re: SIP Trunk transfer error

Post by marcin99 » 31 Aug 2016 06:24

Hi dryhorse,

I checked this options, True/False nothing changed in sip debug.
I read many topics and changed trunk type from ABC-F to ISDN All Countries. In this mode transfer works, but I can't add Routing No. or OpenRouting No. so I add Network No. and is ok, but in Network No. I'm able add only one single number on remote network per entry. Is it possible routing all 10xxx numbers without using prefix to remote network?

User avatar
tot3nkopf
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Alcatel Unleashed Certified Guru
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Joined: 02 Feb 2006 10:41
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Re: SIP Trunk transfer error

Post by tot3nkopf » 01 Sep 2016 16:35

ARS instead of routing nb

Sent from my SM-G920F using Tapatalk

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KIKA
Member
Posts: 44
Joined: 31 Oct 2013 10:07

Re: SIP Trunk transfer error

Post by KIKA » 02 Sep 2016 01:12

Have you try to check changing whit or without compression in the domains?
Works for me.
Instead that REFER message has been a problem for me, I tried to manage with the SBC but OXE doesn't respond as I expected.

Sent from my XT1575 using Tapatalk
Best Regards,
K

ACSE OXE R11
ACSE OXO R8

User avatar
tot3nkopf
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Re: SIP Trunk transfer error

Post by tot3nkopf » 02 Sep 2016 08:01

Or looking back also at some past posts:
viewtopic.php?f=264&t=24063&hilit=oxe+asterisk

Guest

Re: RE: Re: SIP Trunk transfer error

Post by Guest » 02 Sep 2016 10:15

[quote="tot3nkopf"]Or looking back also at some past posts:
viewtopic.php?f=264&t=24063&hilit=oxe+asterisk[/quote]
Thank you. I already have everything with IP address not names.

User avatar
tgn
Member
Posts: 802
Joined: 30 Dec 2009 17:59
Location: Germany

Re: SIP Trunk transfer error

Post by tgn » 02 Sep 2016 13:23

there is a bwps knowledgebase article about refer:

Code: Select all

To allow this method on SIP-ISDN, a new parameter is added on the SIP > SIP External Gateway > Attended Transfer:
If True, the REFER method applies for SIP offnet/offnet attended transfer and the OXE leaves the signalling path.
When two off-net calls connected through the same external gateway and the call is transferred. REFER method enables the external to reroute the destination number and makes OXE to release its resources
If False, the RE-INVITE method applies for SIP offnet/offnet attended transfer and OXE remains in the signalling path.
When user A calls an external user X and goes into conversation. Then A makes another enquiry call to external Y and transfers the call. In such a case the call is retained in OXE side for signaling purpose even after OXE initiates transfer. Only the RTP re-negotiation occurs enabling direct RTP.

Parameter can be set to TRUE, only if SIP > SIP External Gateway > Support Re-invite without SDP is set to TRUE
http://alcatel-lucent-enterprise.force. ... r&nrows=20

also i think there where several issues with the "refer". better also upgrade to latest patch in k1520.

regards...


--- back to basics... focus your eyes on the essential things... ---
--- back to basics... focus your eyes to the essential things... ---

marcin99
Member
Posts: 6
Joined: 20 Jan 2016 04:29

Re: SIP Trunk transfer error

Post by marcin99 » 05 Sep 2016 02:12

@tgn
in Asterisk 13 with option: Support Re-invite without SDP is set to TRUE is problem with all incoming calls, I tested it.
@ tot3nkopf
I used ip address not name. I can use ARS, but then I have to user prefix like for external calls and isn't good solution for my local numbering plan, so it seems that the only solution is to add each separately number from asterisk to Network No.

thanks for yours help

User avatar
blupsy
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Posts: 6
Joined: 27 Feb 2012 13:52

Re: SIP Trunk transfer error

Post by blupsy » 03 Jul 2019 06:19

With the SIP-Trunk from the german Telekom (Deutsche Telekom-DeutschlandLAN SIP Trunk) can be the same problem.
1. external call (ext) to internal subscriber (sub-1): The sub-1 off hook and everthing is fine. :)
2. the sub-1 call the number of a second internal subscriber (sub-2): The ext hear the MOH :)
3. the sub-2 off hook and speak to sub-1: The ext hear the MOH :)
4. the sub-1 on hook and transfer the call to sub-2: The display on sub-2 show ten number of the ext and the ext and the sub-2 hear nothing. :?
5. the sub-2 hook on and the ext become a disconnect. :shock:

The solution is in the external SIP-Gateway.

SIP > SIP Ext Gateway > 'number of the SIP-Gateway'

unmark the

<Support Re-invite without SDP >
and
<Sendonly for hold>

they are normaly marked in the default.
Kind regards
Michael (blupsy)

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