# Asterisk 13 don't send INVITE with Replaces after accepted REFER **URL:** \ **Category:** Asterisk Support\ **Created:** [May 25, 2016, 9:37am UTC](https://community.asterisk.org/t/asterisk-13-dont-send-invite-with-replaces-after-accepted-refer/66801 "2016-05-25T09:37:58Z")\ **Posts on this page:** 1\ **Page:** 1
**Author:** ![jose.molina](https://avatars.discourse-cdn.com/v4/letter/j/a183cd/32.png) [@jose.molina](https://community.asterisk.org/u/jose.molina)\ **Post date:** [May 25, 2016, 9:37am UTC](https://community.asterisk.org/t/asterisk-13-dont-send-invite-with-replaces-after-accepted-refer/66801/1 "2016-05-25T09:37:58Z")
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](http://domain=mydomain.com) autodomain=no [fromdomain=mydomain.com](http://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](http://domain=mydomain.com) autodomain=no context=incoming host=1.1.1.1 [fromdomain=mydomain.com](http://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](http://domain=mydomain.com) autodomain=no host=2.2.2.2 port=5070 [fromdomain=mydomain.com](http://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 ") 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 ") 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