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(${ext...@domain.com)
>
> [outbound]
> exten => _NXXNXXXXXX,1,Dial(DAHDI/g1/${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%3A433dc76eea085f80717687d8084654a2>
> 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-ac739a93395457cf85554d6d8810b0bd6f69;received=SIPX02-IP-ADDRESS
> Via: SIP/2.0/TCP
> SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac70fb6983d8aeacefa051a75c83ce64f8c8~0a78ca617d5b460168faf046fcaf2f1b;id=22276-565
> Via: SIP/2.0/UDP
> SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac6651daee59948275599d7b41f51a249b4d~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%7EYXMyMzQ0ZjE5MA%60%60.900_ntap%2Aid%7EMjIyNzYtNTY1%214bc4cb52b2d1e947feccb17805166d0b>
> 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-ac739a93395457cf85554d6d8810b0bd6f69
> 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;rport=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-ac739a93395457cf85554d6d8810b0bd6f69;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;rport=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;rport=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;rport=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;rport=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 ?
>
> ------------------------------------------------------------------------
>
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