It also appears that YaTE is the same way. that one was a little easier to set up, but it's the same old song and dance: REFER trips it up every time.
I really wish sipXbridge was stable for me. Even with patch20 I drop to one way audio on my local LAN after about 5 minutes and locations that are just a couple of milliseconds ping away from the bridge software drop calls completely after a couple of minutes, and it's not my Adtran router, even when I'm bridging to other SIP devices this happens. I've sent Ranga a snapshot before but because we thought it was my Adtran router it never went anywhere. perhaps I should submit a bug with , or do you still have my original snapshots Ranga? 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/
