Hi;
We use Asterisk as a gateway to connect PSTN with our SIP platform with a SIP trunk. We upgraded our Asterisk gateway from version 11 to 13. After the migration, we have an issue with attended transfer. 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.
The TRANSFER_CONTEXT variable is defined but the call no execute any line of code defined in the dialplan from this context.
These are the sip.conf and dialplan configuration:
sip.conf
[general]
domain=mydomain.com
autodomain=no
fromdomain=mydomain.com
context=default
udpbindaddr=0.0.0.0:5100
tcpenable=no
transport=udp
preferred_codec_only=yes
disallow=all
allow=alaw,g729
nat=force_rport,comedia
canreinvite=no
directmedia=no
allowtransfer=no
pedantic=no
dtmfmode=rfc2833
trustrpid=yes
srvlookup=no
rtptimeout=60
cos_audio=5
cos_sip=3
tos_sip=cs3
tos_audio=ef
[ISP_SIPtrunk]
type=peer
domain=mydomain.com
autodomain=no
context=incoming
host=1.1.1.1
fromdomain=mydomain.com
insecure=port,invite
trustrpid=yes
nat=force_rport,comedia
directmedia=no
pedantic=no
allowtransfer=no
canreinvite=no
dtmfmode=rfc2833
preferred_codec_only=yes
disallow=all
allow=alaw,g729
srvlookup=no
rtptimeout=60
qualify=yes ; SIP OPTIONS (check Agent OK)
qualifyfreq=60 ; 60 seconds
[PROXY-out]
type=peer
domain=mydomain.com
autodomain=no
host=2.2.2.2
port=5070
fromdomain=mydomain.com
context=outgoing
preferred_codec_only=yes
disallow=all
;allow=alaw
allow=alaw,g729
allowtransfer=yes
canreinvite=no
nat=force_rport,comedia
;nat=yes
pedantic=no
directmedia=no
dtmfmode=rfc2833
defaultexpiry=300
trustrpid=yes
srvlookup=no
rtptimeout=60
extensions.conf
[general]
static=yes
writeprotect=yes
language=es
[globals]
TRANSFER_CONTEXT=station_transferUC
_TRANSFER_CONTEXT=station_transferUC
__TRANSFER_CONTEXT=station_transferUC
…
[station_transferUC]
; – Masquerade NAT users
exten => _uX-X.,1,Macro(transferUCAAi,${EXTEN},${CALLERID(num)},${UCLID})
; — URIs
exten => _[a-zA-Z.-@].,1,Macro(transferUCAAi,${EXTEN},${CALLERID(num)},${UCLID})
; INTERNS
exten => _X.,1,Macro(transferUCAAi,${EXTEN},${CALLERID(num)},${UCLID},${IPGW2})
…
[macro-transferUCAA]
; Transfer Active-Active with failover Gateway. 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})
…
In the Asterisk CLI we can see:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [25988@incoming:1] Macro(“SIP/ISP_SIPtrunk-00000000”, “llamadaEntranteUC,25988,37233”) in new stack
– Executing [s@macro-llamadaEntranteUC:1] NoOp(“SIP/ISP_SIPtrunk-00000000”, “CallingPress --1-- caller id --“37233” <37233>–”) in new stack
– Executing [s@macro-llamadaEntranteUC:2] Set(“SIP/ISP_SIPtrunk-00000000”, “TRANSFER_CONTEXT=station_transferUC”) in new stack
– Executing [s@macro-llamadaEntranteUC:3] Set(“SIP/ISP_SIPtrunk-00000000”, “_TRANSFER_CONTEXT=station_transferUC”) in new stack
– Executing [s@macro-llamadaEntranteUC:4] Set(“SIP/ISP_SIPtrunk-00000000”, “__TRANSFER_CONTEXT=station_transferUC”) in new stack
– Executing [s@macro-llamadaEntranteUC:5] Set(“SIP/ISP_SIPtrunk-00000000”, “NUMCALLED=25988”) in new stack
– Executing [s@macro-llamadaEntranteUC:6] Set(“SIP/ISP_SIPtrunk-00000000”, “NUMCALLER=37233”) in new stack
– Executing [s@macro-llamadaEntranteUC:7] GotoIf(“SIP/ISP_SIPtrunk-00000000”, “0?desconegut:conegut”) in new stack
– Goto (macro-llamadaEntranteUC,s,10)
– Executing [s@macro-llamadaEntranteUC:10] GotoIf(“SIP/ISP_SIPtrunk-00000000”, “0?addZeroPrefix”) in new stack
– Executing [s@macro-llamadaEntranteUC:11] Goto(“SIP/ISP_SIPtrunk-00000000”, “s,nextZeroPrefix”) in new stack
– Goto (macro-llamadaEntranteUC,s,13)
– Executing [s@macro-llamadaEntranteUC:13] Set(“SIP/ISP_SIPtrunk-00000000”, “_UCUSER=25988”) in new stack
– Executing [s@macro-llamadaEntranteUC:14] Set(“SIP/ISP_SIPtrunk-00000000”, “CDR(PRIVACY-ID)=“Privacy: off””) in new stack
– Executing [s@macro-llamadaEntranteUC:15] Set(“SIP/ISP_SIPtrunk-00000000”, “CALLERID(all)=37233 <37233>”) in new stack
– Executing [s@macro-llamadaEntranteUC:16] SIPAddHeader(“SIP/ISP_SIPtrunk-00000000”, “P-Asserted-Identity: 37233 sip:37233@mydomain.com”) in new stack
– Executing [s@macro-llamadaEntranteUC:17] GotoIf(“SIP/ISP_SIPtrunk-00000000”, “0?privacy:privacyoff”) in new stack
– Goto (macro-llamadaEntranteUC,s,22)
– Executing [s@macro-llamadaEntranteUC:22] SIPAddHeader(“SIP/ISP_SIPtrunk-00000000”, “Remote-Party-ID: 37233 sip:37233@mydomain.com”) in new stack
– Executing [s@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 1.1.1.1:10748
> 0x7f746800adc0 – Probation passed - setting RTP source address to 2.2.2.2.: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
This same scenario works fine with Asterisk 11 and with the same config files.
-- Executing [25988@incoming:1] Macro("SIP/ISP_SIPtrunk-0000000a", "llamadaEntranteUC,25988,37233") in new stack
-- Executing [s@macro-llamadaEntranteUC:1] NoOp("SIP/ISP_SIPtrunk-0000000a", "CallingPress --1-- caller id --"37233" <37233>--") in new stack
-- Executing [s@macro-llamadaEntranteUC:2] Set("SIP/ISP_SIPtrunk-0000000a", "TRANSFER_CONTEXT=station_transferUC") in new stack
-- Executing [s@macro-llamadaEntranteUC:3] Set("SIP/ISP_SIPtrunk-0000000a", "_TRANSFER_CONTEXT=station_transferUC") in new stack
-- Executing [s@macro-llamadaEntranteUC:4] Set("SIP/ISP_SIPtrunk-0000000a", "__TRANSFER_CONTEXT=station_transferUC") in new stack
-- Executing [s@macro-llamadaEntranteUC:5] Set("SIP/ISP_SIPtrunk-0000000a", "NUMCALLED=25988") in new stack
-- Executing [s@macro-llamadaEntranteUC:6] Set("SIP/ISP_SIPtrunk-0000000a", "NUMCALLER=37233") in new stack
-- Executing [s@macro-llamadaEntranteUC:7] GotoIf("SIP/ISP_SIPtrunk-0000000a", "0?desconegut:conegut") in new stack
-- Goto (macro-llamadaEntranteUC,s,10)
-- Executing [s@macro-llamadaEntranteUC:10] GotoIf("SIP/ISP_SIPtrunk-0000000a", "0?addZeroPrefix") in new stack
-- Executing [s@macro-llamadaEntranteUC:11] Goto("SIP/ISP_SIPtrunk-0000000a", "s,nextZeroPrefix") in new stack
-- Goto (macro-llamadaEntranteUC,s,13)
-- Executing [s@macro-llamadaEntranteUC:13] Set("SIP/ISP_SIPtrunk-0000000a", "_UCUSER=25988") in new stack
-- Executing [s@macro-llamadaEntranteUC:14] Set("SIP/ISP_SIPtrunk-0000000a", "CDR(PRIVACY-ID)="Privacy: off"") in new stack
-- Executing [s@macro-llamadaEntranteUC:15] Set("SIP/ISP_SIPtrunk-0000000a", "CALLERID(all)=37233 <37233>") in new stack
-- Executing [s@macro-llamadaEntranteUC:16] SIPAddHeader("SIP/ISP_SIPtrunk-0000000a", "P-Asserted-Identity: 37233 <sip:37233@mydomain.com>") in new stack
-- Executing [s@macro-llamadaEntranteUC:17] GotoIf("SIP/ISP_SIPtrunk-0000000a", "0?privacy:privacyoff") in new stack
-- Goto (macro-llamadaEntranteUC,s,22)
-- Executing [s@macro-llamadaEntranteUC:22] SIPAddHeader("SIP/ISP_SIPtrunk-0000000a", "Remote-Party-ID: 37233 <sip:37233@mydomain.com>") in new stack
-- Executing [s@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 1.1.1.1:10742
> 0x2c6cbf0 – Probation passed - setting RTP source address to 2.2.2.2:35120
– 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@station_transferUC:1] Macro(“SIP/ISP_SIPtrunk-0000000a”, “transferUCAAi,u6-676,37233,”) in new stack
– Executing [s@macro-transferUCAAi:1] NoOp(“SIP/ISP_SIPtrunk-0000000a”, “Transfer call 37233 to u6-676. Prefix . GWFailover . --UCUser: 25988–”) in new stack
– Executing [s@macro-transferUCAAi:2] Set(“SIP/ISP_SIPtrunk-0000000a”, “NUMDESTI=u6-676”) in new stack
– Executing [s@macro-transferUCAAi:3] Set(“SIP/ISP_SIPtrunk-0000000a”, “NUMORIGEN=37233”) in new stack
– Executing [s@macro-transferUCAAi:4] Set(“SIP/ISP_SIPtrunk-0000000a”, “CALLEDPREFIX=”) in new stack
– Executing [s@macro-transferUCAAi:5] Set(“SIP/ISP_SIPtrunk-0000000a”, “GWFAILOVER=”) in new stack
– Executing [s@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 2.2.2.2:40104
Any idea why Asterisk 13 don’t execute the TRASNFER_CONTEXT code and send the INVITE with Replaces?
Thanks for your help.
Jose