[asterisk-bugs] [JIRA] (ASTERISK-26072) Asterisk 13 don't send INVITE with Replaces after accepted REFER
Jose Molina (JIRA)
noreply at issues.asterisk.org
Tue May 31 01:42:57 CDT 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-26072?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230842#comment-230842 ]
Jose Molina commented on ASTERISK-26072:
----------------------------------------
This is the console output for Asterisk 13.9.1:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [25988 at incoming:1] Macro("SIP/ISP_SIPtrunk-00000000", "llamadaEntranteUC,25988,37233") in new stack
-- Executing [s at macro-llamadaEntranteUC:1] NoOp("SIP/ISP_SIPtrunk-00000000", "CallingPress --1-- caller id --"37233" <37233>--") in new stack
-- Executing [s at macro-llamadaEntranteUC:2] Set("SIP/ISP_SIPtrunk-00000000", "TRANSFER_CONTEXT=station_transferUC") in new stack
-- Executing [s at macro-llamadaEntranteUC:3] Set("SIP/ISP_SIPtrunk-00000000", "TRANSFERCONTEXT=station_transferUC") in new stack
-- Executing [s at macro-llamadaEntranteUC:4] Set("SIP/ISP_SIPtrunk-00000000", "_TRANSFERCONTEXT=station_transferUC") in new stack
-- Executing [s at macro-llamadaEntranteUC:5] Set("SIP/ISP_SIPtrunk-00000000", "NUMCALLED=25988") in new stack
-- Executing [s at macro-llamadaEntranteUC:6] Set("SIP/ISP_SIPtrunk-00000000", "NUMCALLER=37233") in new stack
-- Executing [s at macro-llamadaEntranteUC:7] GotoIf("SIP/ISP_SIPtrunk-00000000", "0?desconegut:conegut") in new stack
-- Goto (macro-llamadaEntranteUC,s,10)
-- Executing [s at macro-llamadaEntranteUC:10] GotoIf("SIP/ISP_SIPtrunk-00000000", "0?addZeroPrefix") in new stack
-- Executing [s at macro-llamadaEntranteUC:11] Goto("SIP/ISP_SIPtrunk-00000000", "s,nextZeroPrefix") in new stack
-- Goto (macro-llamadaEntranteUC,s,13)
-- Executing [s at macro-llamadaEntranteUC:13] Set("SIP/ISP_SIPtrunk-00000000", "_UCUSER=25988") in new stack
-- Executing [s at macro-llamadaEntranteUC:14] Set("SIP/ISP_SIPtrunk-00000000", "CDR(PRIVACY-ID)="Privacy: off"") in new stack
-- Executing [s at macro-llamadaEntranteUC:15] Set("SIP/ISP_SIPtrunk-00000000", "CALLERID(all)=37233 <37233>") in new stack
-- Executing [s at macro-llamadaEntranteUC:16] SIPAddHeader("SIP/ISP_SIPtrunk-00000000", "P-Asserted-Identity: 37233 sip:37233 at mydomain.com") in new stack
-- Executing [s at macro-llamadaEntranteUC:17] GotoIf("SIP/ISP_SIPtrunk-00000000", "0?privacy:privacyoff") in new stack
-- Goto (macro-llamadaEntranteUC,s,22)
-- Executing [s at macro-llamadaEntranteUC:22] SIPAddHeader("SIP/ISP_SIPtrunk-00000000", "Remote-Party-ID: 37233 sip:37233 at mydomain.com") in new stack
-- Executing [s at macro-llamadaEntranteUC:23] Dial("SIP/ISP_SIPtrunk-00000000", "SIP/PROXY-out/25988,,t") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/PROXY-out/25988
-- SIP/PROXY-out-00000001 is ringing
-- SIP/PROXY-out-00000001 answered SIP/ISP_SIPtrunk-00000000
-- Channel SIP/PROXY-out-00000001 joined 'simple_bridge' basic-bridge
-- Channel SIP/ISP_SIPtrunk-00000000 joined 'simple_bridge' basic-bridge
0x7f747800b0b0 -- Probation passed - setting RTP source address to 1.1.1.3:10748
0x7f746800adc0 -- Probation passed - setting RTP source address to 1.1.1.1:30122
-- Started music on hold, class 'default', on channel 'SIP/ISP_SIPtrunk-00000000'
[May 25 11:27:50] NOTICE[19960][C-00000000]: chan_sip.c:23978 handle_response_notify: Got OK on REFER Notify message
And this is the console output for Asterisk 11 where the transfer works fine. As you can see in this case the call enters in the station_transferUC context to do the attendant transfer after receives the REFER message.
-- Executing [25988 at incoming:1] Macro("SIP/ISP_SIPtrunk-0000000a", "llamadaEntranteUC,25988,37233") in new stack
-- Executing [s at macro-llamadaEntranteUC:1] NoOp("SIP/ISP_SIPtrunk-0000000a", "CallingPress --1-- caller id --"37233" <37233>--") in new stack
-- Executing [s at macro-llamadaEntranteUC:2] Set("SIP/ISP_SIPtrunk-0000000a", "TRANSFER_CONTEXT=station_transferUC") in new stack
-- Executing [s at macro-llamadaEntranteUC:3] Set("SIP/ISP_SIPtrunk-0000000a", "_TRANSFER_CONTEXT=station_transferUC") in new stack
-- Executing [s at macro-llamadaEntranteUC:4] Set("SIP/ISP_SIPtrunk-0000000a", "__TRANSFER_CONTEXT=station_transferUC") in new stack
-- Executing [s at macro-llamadaEntranteUC:5] Set("SIP/ISP_SIPtrunk-0000000a", "NUMCALLED=25988") in new stack
-- Executing [s at macro-llamadaEntranteUC:6] Set("SIP/ISP_SIPtrunk-0000000a", "NUMCALLER=37233") in new stack
-- Executing [s at macro-llamadaEntranteUC:7] GotoIf("SIP/ISP_SIPtrunk-0000000a", "0?desconegut:conegut") in new stack
-- Goto (macro-llamadaEntranteUC,s,10)
-- Executing [s at macro-llamadaEntranteUC:10] GotoIf("SIP/ISP_SIPtrunk-0000000a", "0?addZeroPrefix") in new stack
-- Executing [s at macro-llamadaEntranteUC:11] Goto("SIP/ISP_SIPtrunk-0000000a", "s,nextZeroPrefix") in new stack
-- Goto (macro-llamadaEntranteUC,s,13)
-- Executing [s at macro-llamadaEntranteUC:13] Set("SIP/ISP_SIPtrunk-0000000a", "_UCUSER=25988") in new stack
-- Executing [s at macro-llamadaEntranteUC:14] Set("SIP/ISP_SIPtrunk-0000000a", "CDR(PRIVACY-ID)="Privacy: off"") in new stack
-- Executing [s at macro-llamadaEntranteUC:15] Set("SIP/ISP_SIPtrunk-0000000a", "CALLERID(all)=37233 <37233>") in new stack
-- Executing [s at macro-llamadaEntranteUC:16] SIPAddHeader("SIP/ISP_SIPtrunk-0000000a", "P-Asserted-Identity: 37233 <sip:37233 at mydomain.com>") in new stack
-- Executing [s at macro-llamadaEntranteUC:17] GotoIf("SIP/ISP_SIPtrunk-0000000a", "0?privacy:privacyoff") in new stack
-- Goto (macro-llamadaEntranteUC,s,22)
-- Executing [s at macro-llamadaEntranteUC:22] SIPAddHeader("SIP/ISP_SIPtrunk-0000000a", "Remote-Party-ID: 37233 <sip:37233 at mydomain.com>") in new stack
-- Executing [s at macro-llamadaEntranteUC:23] Dial("SIP/ISP_SIPtrunk-0000000a", "SIP/PROXY-out/25988,,t") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/PROXY-out/25988
-- SIP/PROXY-out-0000000b is ringing
-- SIP/PROXY-out-0000000b answered SIP/ISP_SIPtrunk-0000000a
0x7f2b3c006da0 -- Probation passed - setting RTP source address to 1.1.1.3:10742
0x2c6cbf0 -- Probation passed - setting RTP source address to 1.1.1.1:35120
-- Started music on hold, class 'default', on SIP/ISP_SIPtrunk-0000000a
-- Stopped music on hold on SIP/ISP_SIPtrunk-0000000a
== Spawn extension (station_transferUC, u6-676, 1) exited non-zero on 'SIP/ISP_SIPtrunk-0000000a' in macro 'llamadaEntranteUC'
== Spawn extension (station_transferUC, u6-676, 1) exited non-zero on 'SIP/ISP_SIPtrunk-0000000a'
-- Executing [u6-676 at station_transferUC:1] Macro("SIP/ISP_SIPtrunk-0000000a", "transferUCAAi,u6-676,37233,") in new stack
-- Executing [s at macro-transferUCAAi:1] NoOp("SIP/ISP_SIPtrunk-0000000a", "Transfer call 37233 to u6-676. Prefix . GWFailover . --UCUser: 25988--") in new stack
-- Executing [s at macro-transferUCAAi:2] Set("SIP/ISP_SIPtrunk-0000000a", "NUMDESTI=u6-676") in new stack
-- Executing [s at macro-transferUCAAi:3] Set("SIP/ISP_SIPtrunk-0000000a", "NUMORIGEN=37233") in new stack
-- Executing [s at macro-transferUCAAi:4] Set("SIP/ISP_SIPtrunk-0000000a", "CALLEDPREFIX=") in new stack
-- Executing [s at macro-transferUCAAi:5] Set("SIP/ISP_SIPtrunk-0000000a", "GWFAILOVER=") in new stack
-- Executing [s at macro-transferUCAAi:6] Dial("SIP/ISP_SIPtrunk-0000000a", "SIP/NB-out/u6-676,,t") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/NB-out/u6-676
-- SIP/NB-out-0000000c answered SIP/ISP_SIPtrunk-0000000a
0x2c79f50 -- Probation passed - setting RTP source address to 1.1.1.1:40104
Here you can see the dialplan. To summarize, I pasted the contexts involved in the transfer and I masqueraded the IP address for security
[general]
static=yes
writeprotect=yes
language=es
[globals]
DOMINI=mydomain.com
TRANSFER_CONTEXT=station_transferUC
_TRANSFER_CONTEXT=station_transferUC
__TRANSFER_CONTEXT=station_transferUC
[incoming]
exten => s,1,NoOp(OPTION ping)
exten => _[1235679]XXXX,1,Macro(llamadaEntranteUC,${EXTEN:-5},${CALLERID(num)})
exten => _4[45]XXX,1,Macro(llamadaEntranteUC,${EXTEN:-5},${CALLERID(num)})
exten => _0[67]XXXXXXXX,1,Macro(llamadaEntranteUC,${EXTEN},${CALLERID(num)})
exten => _0XX.,1,Macro(llamadaEntranteUC,${EXTEN},${CALLERID(num)})
;---[FI] e-Connect
[macro-llamadaEntranteUC]
exten => s,1,NoOp(CallingPress --${CALLINGPRES}-- caller id --${CALLERID(all)}--)
exten => s,n,Set(TRANSFER_CONTEXT=station_transferUC)
exten => s,n,Set(_TRANSFER_CONTEXT=station_transferUC)
exten => s,n,Set(__TRANSFER_CONTEXT=station_transferUC)
exten => s,n,Set(NUMCALLED=${ARG1})
exten => s,n,Set(NUMCALLER=${ARG2})
exten => s,n,GotoIf($["${NUMCALLER}" = ""]?desconegut:conegut)
exten => s,n(desconegut),Set(NUMCALLER=desconegut)
exten => s,n,Goto(s,nextZeroPrefix)
exten => s,n(conegut),GotoIf($[ $["${NUMCALLER:0:1}" != "0"] & $[${LEN(${NUMCALLER})} != 5] ]?addZeroPrefix)
exten => s,n,Goto(s,nextZeroPrefix)
exten => s,n(addZeroPrefix),Set(NUMCALLER=0${NUMCALLER})
exten => s,n(nextZeroPrefix),Set(_UCUSER=${NUMCALLED})
exten => s,n,Set(CDR(PRIVACY-ID)="Privacy: off")
exten => s,n,Set(CALLERID(all)=${CALLERID(name)} <${NUMCALLER}>)
exten => s,n,SIPAddHeader(P-Asserted-Identity: ${CALLERID(name)} <sip:${NUMCALLER}@${DOMINI}>)
exten => s,n,GotoIf($[${MATH(${CALLINGPRES}>30)}=TRUE]?privacy:privacyoff)
exten => s,n(privacy),SIPAddHeader(Remote-Party-ID: ${CALLERID(name)} <sip:${NUMCALLER}@${DOMINI}>\;privacy=full)
exten => s,n,SIPAddHeader(Privacy: Id)
exten => s,n,Set(CDR(PRIVACY-ID)="Privacy: full")
exten => s,n,GoTo(s,end-privacy)
exten => s,n(privacyoff),SIPAddHeader(Remote-Party-ID: ${CALLERID(name)} <sip:${NUMCALLER}@${DOMINI}>)
exten => s,n(end-privacy),Dial(SIP/PROXY-out/${NUMCALLED},,t); ;t - Callee can transfer
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-CHANUNAVAIL,1,Congestion
exten => s-BUSY,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-VMNOEXIST,1,Goto(s-VMNO-${DIALSTATUS},1)
exten => s-VMNO-BUSY,1,Hangup
exten => s-VMNO-CONGESTION,1,Congestion
exten => s-CANCEL,1,Hangup
exten => s-CONGESTION,1,Congestion
exten => _s-.,1,Congestion
exten => s-.,1,Congestion
[station_transferUC]
; -- Masquerade NAT users
exten => _uX-X.,1,Macro(transferUCAAi,${EXTEN},${CALLERID(num)},${UCLID})
[macro-transferUCAAi]
; Transfer Active-Active, internal. Variable ${UCUSER} ve del INVITE original
exten => s,1,NoOp(Transfer call ${ARG2} to ${ARG1}. Prefix ${ARG3}. GWFailover ${ARG4}. --UCUser: ${UCUSER}--)
exten => s,n,Set(NUMDESTI=${ARG1})
exten => s,n,Set(NUMORIGEN=${ARG2})
exten => s,n,Set(CALLEDPREFIX=${ARG3})
exten => s,n,Set(GWFAILOVER=${ARG4})
exten => s,n(outDial),Dial(SIP/NB-out/${NUMDESTI},,t)
exten => s,n,Goto(s1-${DIALSTATUS},1)
;exten => s,n,Goto(s1-BUSY,1)
; Si la trucada dona un BUSY ho intentem per l'altre GW
exten => s1-BUSY,1,Goto(s2,1)
exten => s1-CHANUNAVAIL,1,Goto(s2,1)
exten => s1-CONGESTION,1,Goto(s2,1)
exten => s1-NOANSWER,1,Hangup
exten => s1-VMNOEXIST,1,Goto(s-VMNO-${DIALSTATUS},1)
exten => s1-VMNO-BUSY,1,Hangup
exten => s1-VMNO-CONGESTION,1,Congestion
exten => s1-CANCEL,1,Hangup
exten => _s1-.,1,Congestion
exten => s2,1(gwasterisk),Dial(SIP/PROXY-out/${CALLEDPREFIX}${NUMDESTI},,t)
exten => s2,n,Goto(s-${DIALSTATUS},1)
exten => s-CHANUNAVAIL,1,Congestion
exten => s-BUSY,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-VMNOEXIST,1,Goto(s-VMNO-${DIALSTATUS},1)
exten => s-VMNO-BUSY,1,Hangup
exten => s-VMNO-CONGESTION,1,Congestion
exten => s-CANCEL,1,Hangup
exten => s-CONGESTION,1,Congestion
exten => _s-.,1,Congestion
exten => s-.,1,Congestion
Thanks for your help
> Asterisk 13 don't send INVITE with Replaces after accepted REFER
> ----------------------------------------------------------------
>
> Key: ASTERISK-26072
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-26072
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General
> Affects Versions: 13.9.1
> Environment: Ubuntu 12.04
> Reporter: Jose Molina
> Assignee: Jose Molina
>
> Hi;
> We use Asterisk as a gateway to connect PSTN with our SIP platform (an opensips proxy) with a SIP trunk. We upgraded our Asterisk gateway from version 11 to 13. After the migration, we have an issue with attended transfer with the chan_sip driver. This is the scenario:
> 1- PSTN number (user A) calls any extension of our SIP platform (user B). B answer the call
> 2- user B holds the incoming call and makes a second call to other SIP extension (user C) to initiate an attended transfer.
> 3- user B transfers the call for try to join A and C users but this transfer fails.
> Analyzing the signalling traffic, we can see that the REFER message arrives from B to the gateway Asterisk with the Replaces header. Asterisk accepts the REFER message with a 202 Accepted response but doesn't send the INVITE with the replaces message to do the transfer. Asterisk send a NOTIFY with a sipfrag reporting 503 Service Unavaiable.
> U 2016/05/30 08:20:02.3871052 1.1.1.2:5070 --> 1.1.1.1:5100
> REFER sip:37233 at 1.1.1.1:5100 SIP/2.0.
> Record-Route: <sip:xuser at 1.1.1.2;lr=on;ftag=4FC4B456-84A45F87>.
> Via: SIP/2.0/UDP 1.1.1.2:5070;branch=z9hG4bKa4e7.68946b95.0.
> Via: SIP/2.0/UDP 1.1.1.2:5070;branch=z9hG4bKa4e7.58946b95.0.
> Via: SIP/2.0/UDP 1.1.1.2;branch=z9hG4bKa4e7.5a17f5d2.0.
> Via: SIP/2.0/UDP 1.1.1.3;rport=5060;branch=z9hG4bK3b060ecd72DB1B2E.
> From: "userB" <sip:25988 at mydomain.com:5070>;tag=4FC4B456-84A45F87.
> To: "userA" <sip:37233 at mydomian.com:5100>;tag=as04e841f5.
> CSeq: 2 REFER.
> Call-ID: 26fc844416113667695f1f3f06d22826 at mydomain.com.
> Contact: <sip:u8-3 at 1.1.1.2>.
> Accept-Language: pl-pl,pl;q=0.9,en;q=0.8.
> Refer-To: <sip:u6-676 at 1.1.1.2?Replaces=4827f2a1-e2ca8142-5416e633%401.1.1.3%3Bto-tag%3D1422695821%3Bfrom-tag%3D107F133D-7BEB429E>.
> Referred-By: <sip:userB at mydomain.com>.
> Max-Forwards: 11.
> Content-Length: 0.
> .
> U 2016/05/30 08:20:02.387120 1.1.1.1:5100 -> 1.1.1.2:5070
> SIP/2.0 202 Accepted.
> Via: SIP/2.0/UDP 1.1.1.2:5070;branch=z9hG4bKa4e7.68946b95.0;received=1.1.1.2;rport=5070.
> Via: SIP/2.0/UDP 1.1.1.2:5070;branch=z9hG4bKa4e7.58946b95.0.
> Via: SIP/2.0/UDP 1.1.1.2;branch=z9hG4bKa4e7.5a17f5d2.0.
> Via: SIP/2.0/UDP 1.1.1.3;rport=5060;branch=z9hG4bK3b060ecd72DB1B2E.
> Record-Route: <sip:xuser at 1.1.1.2;lr=on;ftag=4FC4B456-84A45F87>.
> From: "userB" <sip:25988 at mydomain.com:5070>;tag=4FC4B456-84A45F87.
> To: "userA" <sip:37233 at mydomain.com:5100>;tag=as04e841f5.
> Call-ID: 26fc844416113667695f1f3f06d22826 at mydomain.com.
> CSeq: 2 REFER.
> Server: Asterisk PBX 13.9.1.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
> Supported: replaces, timer.
> Contact: <sip:37233 at 1.1.1.1:5100>.
> Content-Length: 0.
> .
> U 2016/05/30 08:20:02.387228 1.1.1.1:5100 -> 1.1.1.2:5070
> NOTIFY sip:u8-3 at 1.1.1.2 SIP/2.0.
> Via: SIP/2.0/UDP 1,1,1,1:5100;branch=z9hG4bK285dcfd6;rport.
> Route: <sip:1.1.1.2:5070;lr=on;ftag=as04e841f5>,<sip:1.1.1.2:5070;lr=on;ftag=as04e841f5>,<sip:xuser at 1.1.1.2;lr=on;ftag=as04e841f5>.
> Max-Forwards: 70.
> From: "userA" <sip:37233 at mydomain.com:5100>;tag=as04e841f5.
> To: "userB" <sip:25988 at mydomain.com:5070>;tag=4FC4B456-84A45F87.
> Contact: <sip:37233 at 1.1.1.1:5100>.
> Call-ID: 26fc844416113667695f1f3f06d22826 at mydomain.com.
> CSeq: 103 NOTIFY.
> User-Agent: Asterisk PBX 13.9.1.
> Event: refer;id=2.
> 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, timer.
> Content-Length: 33.
> .
> SIP/2.0 503 Service Unavailable.
> The TRANSFER_CONTEXT variable is defined but the call no execute any line of code defined in the dialplan from this context.
> This same scenario works fine with Asterisk 11 and with the same config files. Asterisk sends a NOTIFY with a sifrag 200 OK and sent and INVITE with Replaces and the trasnfer is performed.
> Any idea why Asterisk 13 don't execute the TRASNFER_CONTEXT code and send the INVITE with Replaces? Is a possible bug in this release?
> Thanks for your help.
> Jose
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