Well, I spent all day setting up FreeSWITCH on PFSense and I can make calls in and out of it just fine however, just like EVERYTHING else I've tried, it chokes on REFER when doing an attended transfer, with or without MoH. Freeswitch does, however, play nicely with forwarding rules unlike Asterisk. Here is the weird error I was getting in freeswitch when I tried to do an attended transfer (emphasis mine):
2009-11-14 07:17:31.195282 [DEBUG] sofia_glue.c:2292 AUDIO RTP CHANGING DEST TO: [0.0.0.0:2260] 2009-11-14 07:17:31.195282 [DEBUG] sofia.c:3576 Processing Reinvite 2009-11-14 07:17:31.195282 [DEBUG] sofia.c:3210 Channel sofia/lan/4...@ippbx.co.brazos.tx.us entering state [completed][200] 2009-11-14 07:17:31.221308 [DEBUG] sofia.c:3210 Channel sofia/lan/4...@ippbx.co.brazos.tx.us entering state [ready][200] 2009-11-14 07:17:54.441411 [DEBUG] sofia.c:3911 Process REFER to [4...@ippbx.co.brazos.tx.us] *2009-11-14 07:17:54.441411 [DEBUG] sofia.c:3938 Replaces: [95daff7b-93ef1bd0-38a1b...@10.200.25.170;to-tag=E7AE08F0-C76ABE3D;from-tag=32361EDE-33E55EF7&X-sipX-Authidentity=<sip:4...@ippbx.co.brazos.tx.us;signature=4AFE0542%3A%3A2e17bc28f9ba044d84efa0ec63478513>] 2009-11-14 07:17:54.441411 [DEBUG] sofia.c:4114 Good Luck, you'll need it......* 2009-11-14 07:17:54.441411 [NOTICE] switch_channel.c:602 New Channel sofia/lan/sip:4...@ippbx.co.brazos.tx.us [16320c89-bbd0-de11-82f8-0002a56bdd24] 2009-11-14 07:17:54.441411 [DEBUG] mod_sofia.c:2741 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State Change CS_NEW -> CS_INIT 2009-11-14 07:17:54.441411 [DEBUG] switch_core_session.c:933 Send signal sofia/lan/sip:4...@ippbx.co.brazos.tx.us [BREAK] 2009-11-14 07:17:54.444226 [DEBUG] switch_core_state_machine.c:397 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) Running State Change CS_INIT 2009-11-14 07:17:54.444226 [DEBUG] switch_core_state_machine.c:480 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State INIT 2009-11-14 07:17:54.444226 [DEBUG] mod_sofia.c:83 sofia/lan/sip:4...@ippbx.co.brazos.tx.us SOFIA INIT 2009-11-14 07:17:54.444226 [DEBUG] mod_sofia.c:111 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State Change CS_INIT -> CS_ROUTING 2009-11-14 07:17:54.444226 [DEBUG] switch_core_session.c:933 Send signal sofia/lan/sip:4...@ippbx.co.brazos.tx.us [BREAK] 2009-11-14 07:17:54.444226 [DEBUG] switch_core_state_machine.c:480 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State INIT going to sleep 2009-11-14 07:17:54.444226 [DEBUG] switch_core_state_machine.c:397 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) Running State Change CS_ROUTING 2009-11-14 07:17:54.444226 [DEBUG] switch_core_state_machine.c:483 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State ROUTING 2009-11-14 07:17:54.444226 [DEBUG] mod_sofia.c:130 sofia/lan/sip:4...@ippbx.co.brazos.tx.us SOFIA ROUTING 2009-11-14 07:17:54.444226 [DEBUG] switch_ivr_originate.c:63 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-14 07:17:54.444226 [DEBUG] switch_core_session.c:933 Send signal sofia/lan/sip:4...@ippbx.co.brazos.tx.us [BREAK] 2009-11-14 07:17:54.444226 [DEBUG] switch_core_state_machine.c:483 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State ROUTING going to sleep 2009-11-14 07:17:54.444226 [DEBUG] switch_core_state_machine.c:397 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) Running State Change CS_CONSUME_MEDIA 2009-11-14 07:17:54.444226 [DEBUG] switch_core_state_machine.c:502 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State CONSUME_MEDIA 2009-11-14 07:17:54.446695 [DEBUG] sofia.c:3210 Channel sofia/lan/sip:4...@ippbx.co.brazos.tx.us entering state [calling][0] 2009-11-14 07:17:54.591412 [DEBUG] sofia.c:3210 Channel sofia/lan/sip:4...@ippbx.co.brazos.tx.us entering state [terminated][481] 2009-11-14 07:17:54.591412 [NOTICE] sofia.c:3770 Hangup sofia/lan/sip:4...@ippbx.co.brazos.tx.us [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-11-14 07:17:54.591412 [DEBUG] switch_channel.c:1683 Send signal sofia/lan/sip:4...@ippbx.co.brazos.tx.us [KILL] 2009-11-14 07:17:54.591412 [DEBUG] switch_core_session.c:933 Send signal sofia/lan/sip:4...@ippbx.co.brazos.tx.us [BREAK] 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:502 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State CONSUME_MEDIA going to sleep 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:397 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) Running State Change CS_HANGUP 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:433 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State HANGUP 2009-11-14 07:17:54.600062 [DEBUG] mod_sofia.c:306 sofia/lan/sip:4...@ippbx.co.brazos.tx.us Overriding SIP cause 503 with 481 from the other leg 2009-11-14 07:17:54.600062 [DEBUG] mod_sofia.c:338 Channel sofia/lan/sip:4...@ippbx.co.brazos.tx.us hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:46 sofia/lan/sip:4...@ippbx.co.brazos.tx.us Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:433 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State HANGUP going to sleep 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:475 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State Change CS_HANGUP -> CS_REPORTING 2009-11-14 07:17:54.600062 [DEBUG] switch_core_session.c:933 Send signal sofia/lan/sip:4...@ippbx.co.brazos.tx.us [BREAK] 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:397 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) Running State Change CS_REPORTING 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:607 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State REPORTING 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:53 sofia/lan/sip:4...@ippbx.co.brazos.tx.us Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:607 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State REPORTING going to sleep 2009-11-14 07:17:54.600062 [DEBUG] switch_core_state_machine.c:410 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State Change CS_REPORTING -> CS_DESTROY 2009-11-14 07:17:54.600062 [DEBUG] switch_core_session.c:1067 Session 19 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) Locked, Waiting on external entities 2009-11-14 07:17:55.259441 [DEBUG] switch_ivr_originate.c:2101 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2009-11-14 07:17:55.259441 [ERR] sofia.c:3824 Cannot Create Outgoing Channel! [sofia/lan/sip:4...@ippbx.co.brazos.tx.us] 2009-11-14 07:17:55.259441 [NOTICE] switch_core_session.c:1085 Session 19 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) Ended 2009-11-14 07:17:55.259441 [NOTICE] switch_core_session.c:1087 Close Channel sofia/lan/sip:4...@ippbx.co.brazos.tx.us [CS_DESTROY] 2009-11-14 07:17:55.259441 [DEBUG] switch_core_state_machine.c:559 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State DESTROY 2009-11-14 07:17:55.259441 [DEBUG] mod_sofia.c:255 sofia/lan/sip:4...@ippbx.co.brazos.tx.us SOFIA DESTROY 2009-11-14 07:17:55.259441 [DEBUG] switch_core_state_machine.c:60 sofia/lan/sip:4...@ippbx.co.brazos.tx.us Standard DESTROY 2009-11-14 07:17:55.259441 [DEBUG] switch_core_state_machine.c:559 (sofia/lan/sip:4...@ippbx.co.brazos.tx.us) State DESTROY going to sleep Picher, Michael wrote: > > I played with it a little bit… did a little write-up here: > http://sipxecs.blogspot.com/2009/09/pfsense-with-freeswitch-for-sip-trunks.html > > Mike > > *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net] > *Sent:* Friday, November 13, 2009 8:57 AM > *To:* Picher, Michael > *Cc:* Josh Patten; sipx-users@list.sipfoundry.org; > gca...@franklinamerican.com > *Subject:* Re: [sipx-users] Call Forwarding: Sipxecs with Asterisk > MediaGateway > > Good point. Never tried it, but once you get pfSense up and running > (it aint hard!), installing freeswitch is 2 clicks. I don;t know about > configuring it, but at least the effort involved in getting it to that > point is painfully easy. > > On Fri, Nov 13, 2009 at 8:54 AM, Picher, Michael > <mpic...@cmctechgroup.com <mailto:mpic...@cmctechgroup.com>> wrote: > > pfSense has a Freeswitch add-in that will give you a simplistic GUI to > freeswitch if you want to go that route. > > Mike > > > -----Original Message----- > From: sipx-users-boun...@list.sipfoundry.org > <mailto:sipx-users-boun...@list.sipfoundry.org> > [mailto:sipx-users-boun...@list.sipfoundry.org > <mailto:sipx-users-boun...@list.sipfoundry.org>] On Behalf Of Josh Patten > Sent: Thursday, November 12, 2009 1:27 PM > To: sipx-users@list.sipfoundry.org > <mailto:sipx-users@list.sipfoundry.org>; gca...@franklinamerican.com > <mailto:gca...@franklinamerican.com> > Subject: Re: [sipx-users] Call Forwarding: Sipxecs with Asterisk > MediaGateway > > Unfortunately there is no fix for this other than submitting a bug to > Digium and having it ignored. The SIP stack in asterisk is pretty > shoddy, even in 1.6 as I currently see this problem as well. What you > can try to do is run it through the SBC (sipXbridge) as that will water > the SIP down enough for asterisk to work with however I could never make > > sipXbridge work reliably on my internal network (lots of call drops and > one-way audio for reasons no one could figure out, though you may have > better luck.) At this point you might try callweaver, yate, or > freeswitch instead of Asterisk as their SIP stacks are more complete. If > > you've learned asterisk to a point where you have it making and > receiving calls, you can learn the other 3 with ease. > > I am planning on purchasing a quad-PRI audiocodes mediant 1000 soon > though; The price sucks but it is "certified" to work. > > sipx-users-requ...@list.sipfoundry.org > <mailto:sipx-users-requ...@list.sipfoundry.org> wrote: > > > > Subject: > > [sipx-users] Call Forwarding: Sipxecs with Asterisk Media Gateway > > From: > > Gabe Casey <gca...@franklinamerican.com > <mailto:gca...@franklinamerican.com>> > > Date: > > Thu, 12 Nov 2009 11:15:13 -0600 (CST) > > To: > > sipx-users@list.sipfoundry.org <mailto:sipx-users@list.sipfoundry.org> > > > > To: > > sipx-users@list.sipfoundry.org <mailto:sipx-users@list.sipfoundry.org> > > > > > > I am having some issues using Asterisk as a PRI gateway with Sipxecs. > > For the most part it works for inbound and outbound calling however > > when a call is received on a PRI channel and then send to a SipXecs > > extension which has a forwarding rule to ring the extension and a > > mobile device at the same time > > asterisk quickly cancels the call to the extension while allowing the > > mobile to ring. > > > > I have 2 media gateways and 2 sipxecs proxies this behavior is not > > happening when the call comes from GW2 then gets forwarded out GW1 (or > > > vice versa) > > > > Call --> PRI ---> Asterisk PRI GW 1 ---> Sipxecs (Forward Rule > > "simultaneous ring") ---> Asterisk PRI GW 1 ----> (SipX Exten + > > Mobile) SipX Exten Rings One Time while mobile rings as expected. > > > > Some of my calls come in another gateway and when this happens the > > call is handled properly: > > > > Call --> PRI ---> Asterisk PRI GW 2 ---> Sipxecs (Forward Rule > > "simultaneous ring") ---> Asterisk PRI GW 1 ----> (SipX Exten + > > Mobile) Expected result both extensions ring > > > > Both Asterisk PRI GWs are set up as unmanaged gateways in sipxecs. > > > > Peer Def in asterisk look like this: > > > > [general] > > trustrpid = yes > > sendrpid = yes > > progressinband=never > > srvlookup=yes > > > > [GW01] > > type=friend > > port=5060 > > insecure=invite,port > > host=GW01.domain.com <http://GW01.domain.com> > > context=default > > dtmfmode=rfc2833 > > > > > > [GW02] > > type=friend > > port=5060 > > insecure=invite,port > > host=GW02.domain.com <http://GW02.domain.com> > > context=default > > dtmfmode=rfc2833 > > > > > > Dialplan is basically > > > > [inbound] > > exten => _XXXX,1,AGI(route.php) > > exten => _XXXX,2, Dial(${ext...@domain.com <mailto:exten...@domain.com>) > > > > [outbound] > > exten => _NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN} > > > > [default] > > include => inbound > > include => outbound > > > > > > here is the sip debug from server --- calling my did which routes to > > exten 2945 on sipxecs > > > > > > Content-Length: 316 > > Expires: 60 > > X-Sipx-Authidentity: > > > <sip:2...@domain.com;signature=4AFC3F27%3A433dc76eea085f80717687d8084654 > a2> > > X-Sipx-Handled: XSIPX02-IP-ADDRESS-67.107.93.2 > > > > v=0 > > o=root 1085943255 1085943255 IN IP4 GW01-IP-ADDRESS > > s=Asterisk PBX 1.6.2.0-rc4 > > c=IN IP4 GW01-IP-ADDRESS > > t=0 0 > > m=audio 15766 RTP/AVP 0 3 8 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > > > <-------------> > > --- (21 headers 14 lines) --- > > > > <--- Transmitting (no NAT) to SIPX02-IP-ADDRESS:5060 ---> > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP > > > SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac739a93395457cf85554d6d8810b0b > d6f69;received=SIPX02-IP-ADDRESS > > Via: SIP/2.0/TCP > > > SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac70fb6983d8aeacefa051a75c83ce6 > 4f8c8~0a78ca617d5b460168faf046fcaf2f1b;id=22276-565 > > Via: SIP/2.0/UDP > > > SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac6651daee59948275599d7b41f51a2 > 49b4d~1bfa448fba164a3d273549fca4a8a79d > > Via: SIP/2.0/UDP > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060 > > Record-Route: > > > <sip:SIPX02-IP-ADDRESS:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EYXMyMzQ0Z > jE5MA%60%60.900_ntap%2Aid%7EMjIyNzYtNTY1%214bc4cb52b2d1e947feccb17805166 > d0b> > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com> > > Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > CSeq: 102 INVITE > > Server: Asterisk PBX 1.6.2.0-rc4 > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > > Supported: replaces, timer > > Contact: <sip:6185591...@gw01-ip-address> > > Content-Length: 0 > > > > > > <------------> > > * -- Now forwarding DAHDI/11-1 to 'Local/6932...@default' (thanks > > to SIP/DOMAIN.com-00001844)* > > Scheduling destruction of SIP dialog > > '44c969f1000b2c574b3245e779126...@gw01-ip-address' in 32000 ms > > (Method: INVITE) > > Reliably Transmitting (no NAT) to SIPX02-IP-ADDRESS:5060: > > CANCEL sip:2...@domain.com SIP/2.0 > > Via: SIP/2.0/UDP GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport > > Max-Forwards: 70 > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com> > > Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > CSeq: 102 CANCEL > > User-Agent: Asterisk PBX 1.6.2.0-rc4 > > Content-Length: 0 > > > > > > --- > > Scheduling destruction of SIP dialog > > '44c969f1000b2c574b3245e779126...@gw01-ip-address' in 32000 ms > > (Method: INVITE) > > -- Executing [6932...@default:1] > > Dial("Local/6932...@default-e585;2", "DAHDI/g2/6932833") in new stack > > -- Requested transfer capability: 0x00 - SPEECH > > -- Called g2/6932833 > > plastmg01*CLI> > > <--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 ---> > > SIP/2.0 200 OK > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=a70a3f79 > > Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > Cseq: 102 CANCEL > > Via: SIP/2.0/UDP > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060 > > Content-Length: 0 > > > > > > <-------------> > > --- (7 headers 0 lines) --- > > plastmg01*CLI> > > <--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 ---> > > CANCEL > > sip:6932...@gw01-ip-address;sipx-noroute=Voicemail;transport=udp > SIP/2.0 > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com> > > Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > Cseq: 102 CANCEL > > Max-Forwards: 20 > > Via: SIP/2.0/UDP > > > SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac739a93395457cf85554d6d8810b0b > d6f69 > > Content-Length: 0 > > > > > > <-------------> > > --- (8 headers 0 lines) --- > > Sending to SIPX02-IP-ADDRESS : 5060 (no NAT) > > Scheduling destruction of SIP dialog > > '44c969f1000b2c574b3245e779126...@gw01-ip-address' in 32000 ms > > (Method: CANCEL) > > plastmg01*CLI> > > <--- Reliably Transmitting (no NAT) to SIPX02-IP-ADDRESS:5060 ---> > > SIP/2.0 487 Request Terminated > > Via: SIP/2.0/UDP > > > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;r > port=5060 > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=a70a3f79 > > Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > CSeq: 102 INVITE > > Server: Asterisk PBX 1.6.2.0-rc4 > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > > Supported: replaces, timer > > Content-Length: 0 > > > > > > <------------> > > plastmg01*CLI> > > <--- Transmitting (no NAT) to SIPX02-IP-ADDRESS:5060 ---> > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > > > SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac739a93395457cf85554d6d8810b0b > d6f69;received=SIPX02-IP-ADDRESS > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=a70a3f79 > > Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > CSeq: 102 CANCEL > > Server: Asterisk PBX 1.6.2.0-rc4 > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > > Supported: replaces, timer > > Content-Length: 0 > > > > > > <------------> > > plastmg01*CLI> > > <--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 ---> > > SIP/2.0 408 Request timeout > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=023e4750 > > Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > Cseq: 102 INVITE > > Via: SIP/2.0/UDP > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060 > > Server: sipXecs/4.0.2 sipXecs/sipXproxy (Linux) > > Content-Length: 0 > > > > > > <-------------> > > --- (8 headers 0 lines) --- > > plastmg01*CLI> > > <--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 ---> > > SIP/2.0 408 Request timeout > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=023e4750 > > Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > Cseq: 102 INVITE > > Via: SIP/2.0/UDP > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060 > > Server: sipXecs/4.0.2 sipXecs/sipXproxy (Linux) > > Content-Length: 0 > > > > > > <-------------> > > --- (8 headers 0 lines) --- > > -- DAHDI/30-1 is proceeding passing it to > Local/6932...@default-e585;2 > > -- Local/6932...@default-e585;1 is proceeding passing it to > DAHDI/11-1 > > plastmg01*CLI> > > <--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 ---> > > SIP/2.0 408 Request timeout > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=023e4750 > > Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > Cseq: 102 INVITE > > Via: SIP/2.0/UDP > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060 > > Server: sipXecs/4.0.2 sipXecs/sipXproxy (Linux) > > Content-Length: 0 > > > > > > <-------------> > > --- (8 headers 0 lines) --- > > plastmg01*CLI> > > <--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 ---> > > SIP/2.0 408 Request timeout > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=023e4750 > > Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > Cseq: 102 INVITE > > Via: SIP/2.0/UDP > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060 > > Server: sipXecs/4.0.2 sipXecs/sipXproxy (Linux) > > Content-Length: 0 > > > > > > <-------------> > > --- (8 headers 0 lines) --- > > Retransmitting #1 (no NAT) to SIPX02-IP-ADDRESS:5060: > > SIP/2.0 487 Request Terminated > > Via: SIP/2.0/UDP > > > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;r > port=5060 > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=a70a3f79 > > Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > CSeq: 102 INVITE > > Server: Asterisk PBX 1.6.2.0-rc4 > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > > Supported: replaces, timer > > Content-Length: 0 > > > > > > --- > > -- DAHDI/30-1 is making progress passing it to > > Local/6932...@default-e585;2 > > -- DAHDI/30-1 is making progress passing it to > > Local/6932...@default-e585;2 > > -- Local/6932...@default-e585;1 is making progress passing it to > > DAHDI/11-1 > > -- Local/6932...@default-e585;1 is making progress passing it to > > DAHDI/11-1 > > Retransmitting #2 (no NAT) to SIPX02-IP-ADDRESS:5060: > > SIP/2.0 487 Request Terminated > > Via: SIP/2.0/UDP > > > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;r > port=5060 > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=a70a3f79 > > Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > CSeq: 102 INVITE > > Server: Asterisk PBX 1.6.2.0-rc4 > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > > Supported: replaces, timer > > Content-Length: 0 > > > > > > --- > > -- DAHDI/30-1 answered Local/6932...@default-e585;2 > > -- Local/6932...@default-e585;1 answered DAHDI/11-1 > > -- Native bridging DAHDI/11-1 and DAHDI/30-1 > > == Spawn extension (default, 6932833, 1) exited non-zero on > > 'Local/6932...@default-e585;2' > > Retransmitting #3 (no NAT) to SIPX02-IP-ADDRESS:5060: > > SIP/2.0 487 Request Terminated > > Via: SIP/2.0/UDP > > > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;r > port=5060 > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=a70a3f79 > > Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > CSeq: 102 INVITE > > Server: Asterisk PBX 1.6.2.0-rc4 > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > > Supported: replaces, timer > > Content-Length: 0 > > > > > > --- > > Retransmitting #4 (no NAT) to SIPX02-IP-ADDRESS:5060: > > SIP/2.0 487 Request Terminated > > Via: SIP/2.0/UDP > > > GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;r > port=5060 > > From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190 > > To: <sip:2...@domain.com>;tag=a70a3f79 > > Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address > > CSeq: 102 INVITE > > Server: Asterisk PBX 1.6.2.0-rc4 > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > > Supported: replaces, timer > > Content-Length: 0 > > > > finally asterisk will report something like the following .... (note > > this is not from the above call so the call-id is different) > > > > [Nov 12 11:02:57] WARNING[6378]: chan_sip.c:3782 retrans_pkt: Maximum > > retries exceeded on transmission > > 1e88918e19216bcb4a4d43fb7793c...@gw01-ip-address for seqno 102 > > (Critical Response) -- See doc/sip-retransmit.txt. > > Really destroying SIP dialog > > '1e88918e19216bcb4a4d43fb7793c...@gw01-ip-address' Method: CANCEL > > > > > > It seem that asterisk just wants to forward the call to the mobile > > device and cancel the extens call > > Can anyone advise me on a working config for this ? > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > sipx-users mailing list sipx-users@list.sipfoundry.org > <mailto:sipx-users@list.sipfoundry.org> > > List Archive: http://list.sipfoundry.org/archive/sipx-users > > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > > sipXecs IP PBX -- http://www.sipfoundry.org/ > > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > <mailto:sipx-users@list.sipfoundry.org> > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > <mailto:sipx-users@list.sipfoundry.org> > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
