I played with it a little bit... did a little write-up here: http://sipxecs.blogspot.com/2009/09/pfsense-with-freeswitch-for-sip-trun ks.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> 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] On Behalf Of Josh Patten Sent: Thursday, November 12, 2009 1:27 PM To: sipx-users@list.sipfoundry.org; 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 wrote: > > Subject: > [sipx-users] Call Forwarding: Sipxecs with Asterisk Media Gateway > From: > Gabe Casey <gca...@franklinamerican.com> > Date: > Thu, 12 Nov 2009 11:15:13 -0600 (CST) > To: > sipx-users@list.sipfoundry.org > > To: > 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 > context=default > dtmfmode=rfc2833 > > > [GW02] > type=friend > port=5060 > insecure=invite,port > host=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 > 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/ _______________________________________________ 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/
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