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