[Asterisk-Dev] AGI still falls over

[Steven Critchfield]

Looks like you GS won't ack the file transfer and things break down from
there. Since you know the file transfer is useless on a GS, so is the
sendtext and sendimage, why don't you comment them out and see where you
get from there.

On Fri, 2004-01-02 at 16:17, Iain Stevenson wrote:
> --On Friday, January 2, 2004 3:54 pm -0600 Mark Spencer 
> <markster at digium.com> wrote:
> 
> > Can you use sip debug and look at what is getting retransmitted?
> >
> > Mark
> >
> 
> It seems to try and retransmit each digit until it finally breaks.
> 
>   Iain
> 
> 
> 
> Sending text hello world on SIP/grandstream-8d3c
> Really sending text hello world on SIP/grandstream-8d3c
> set_destination: Parsing <sip:grandstream at 192.168.1.242> for address/port 
> to send to
> set_destination: set destination to 192.168.1.242, port 5060
> Reliably Transmitting:
> MESSAGE sip:grandstream at 192.168.1.242 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1d4ed43b
> From: <sip:35 at 192.168.1.254>;tag=as57e4ccaf
> To: "Iain" 
> <sip:grandstream at 192.168.1.254>;tag=6af1a8f1-28fd-2feb-515b-d4a937d0e2f0
> Contact: <sip:35 at 192.168.1.254>
> Call-ID: 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242
> CSeq: 102 MESSAGE
> User-Agent: Asterisk PBX
> Content-Type: text/plain
> Content-Length: 11
> 
> hello world (no NAT) to 192.168.1.242:5060
> PASS (0)
> 3.  Testing 'sendimage'...PASS (0)
>     -- Playing 'digits/1' (language 'en')
>     -- Playing 'digits/hundred' (language 'en')
> Retransmitting #1 (no NAT):
> MESSAGE sip:grandstream at 192.168.1.242 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1d4ed43b
> From: <sip:35 at 192.168.1.254>;tag=as57e4ccaf
> To: "Iain" 
> <sip:grandstream at 192.168.1.254>;tag=6af1a8f1-28fd-2feb-515b-d4a937d0e2f0
> Contact: <sip:35 at 192.168.1.254>
> Call-ID: 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242
> CSeq: 102 MESSAGE
> User-Agent: Asterisk PBX
> Content-Type: text/plain
> Content-Length: 11
> 
> hello world
>  to 192.168.1.242:5060
>     -- Playing 'digits/90' (language 'en')
> Retransmitting #2 (no NAT):
> MESSAGE sip:grandstream at 192.168.1.242 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1d4ed43b
> From: <sip:35 at 192.168.1.254>;tag=as57e4ccaf
> To: "Iain" 
> <sip:grandstream at 192.168.1.254>;tag=6af1a8f1-28fd-2feb-515b-d4a937d0e2f0
> Contact: <sip:35 at 192.168.1.254>
> Call-ID: 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242
> CSeq: 102 MESSAGE
> User-Agent: Asterisk PBX
> Content-Type: text/plain
> Content-Length: 11
> 
> hello world
>  to 192.168.1.242:5060
>     -- Playing 'digits/2' (language 'en')
>     -- Playing 'digits/million' (language 'en')
> Retransmitting #3 (no NAT):
> MESSAGE sip:grandstream at 192.168.1.242 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1d4ed43b
> From: <sip:35 at 192.168.1.254>;tag=as57e4ccaf
> To: "Iain" 
> <sip:grandstream at 192.168.1.254>;tag=6af1a8f1-28fd-2feb-515b-d4a937d0e2f0
> Contact: <sip:35 at 192.168.1.254>
> Call-ID: 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242
> CSeq: 102 MESSAGE
> User-Agent: Asterisk PBX
> Content-Type: text/plain
> Content-Length: 11
> 
> hello world
>  to 192.168.1.242:5060
>     -- Playing 'digits/8' (language 'en')
>     -- Playing 'digits/hundred' (language 'en')
> Retransmitting #4 (no NAT):
> MESSAGE sip:grandstream at 192.168.1.242 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1d4ed43b
> From: <sip:35 at 192.168.1.254>;tag=as57e4ccaf
> To: "Iain" 
> <sip:grandstream at 192.168.1.254>;tag=6af1a8f1-28fd-2feb-515b-d4a937d0e2f0
> Contact: <sip:35 at 192.168.1.254>
> Call-ID: 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242
> CSeq: 102 MESSAGE
> User-Agent: Asterisk PBX
> Content-Type: text/plain
> Content-Length: 11
> 
> hello world
>  to 192.168.1.242:5060
>     -- Playing 'digits/30' (language 'en')
> Retransmitting #5 (no NAT):
> MESSAGE sip:grandstream at 192.168.1.242 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1d4ed43b
> From: <sip:35 at 192.168.1.254>;tag=as57e4ccaf
> To: "Iain" 
> <sip:grandstream at 192.168.1.254>;tag=6af1a8f1-28fd-2feb-515b-d4a937d0e2f0
> Contact: <sip:35 at 192.168.1.254>
> Call-ID: 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242
> CSeq: 102 MESSAGE
> User-Agent: Asterisk PBX
> Content-Type: text/plain
> Content-Length: 11
> 
> hello world
>  to 192.168.1.242:5060
>     -- Playing 'digits/7' (language 'en')
>     -- Playing 'digits/thousand' (language 'en')
> WARNING[6151]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries 
> exceeded on call 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242 for 
> seqno 102 (Request)
>   == Spawn extension (voip-sip, 35, 2) exited non-zero on 
> 'SIP/grandstream-8d3c'
> PASS (-1)
> 5.  Testing 'waitdtmf'...FAIL (unexpected result '')
> 6.  Testing 'record'...FAIL (unexpected result '')
> 6a.  Testing 'record' playback...FAIL (unexpected result '')
> ================== Complete ======================
> 7 tests completed, 4 passed, 3 failed
> ==================================================
> set_destination: Parsing <sip:grandstream at 192.168.1.242> for address/port 
> to send to
> set_destination: set destination to 192.168.1.242, port 5060
> Reliably Transmitting:
> BYE sip:grandstream at 192.168.1.242 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1d4ed43b
> From: <sip:35 at 192.168.1.254>;tag=as57e4ccaf
> To: "Iain" 
> <sip:grandstream at 192.168.1.254>;tag=6af1a8f1-28fd-2feb-515b-d4a937d0e2f0
> Contact: <sip:35 at 192.168.1.254>
> Call-ID: 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Content-Length: 0
> 
>  (no NAT) to 192.168.1.242:5060
> Sip read:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK1d4ed43b
> From: <sip:35 at 192.168.1.254>;tag=as57e4ccaf
> To: "Iain" 
> <sip:grandstream at 192.168.1.254>;tag=6af1a8f1-28fd-2feb-515b-d4a937d0e2f0
> Call-ID: 93dd1b14-c3a7-dbe9-b4e1-d3e7e86bd3f5 at 192.168.1.242
> CSeq: 103 BYE
> User-Agent: Grandstream SIP UA 1.0.4.17
> Contact: <sip:grandstream at 192.168.1.242>
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
> Content-Length: 0
> 
> 
> 10 headers, 0 lines
> Message is BYE
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> > On Fri, 2 Jan 2004, Iain Stevenson wrote:
> >
> >>
> >>
> >> --On Friday, January 2, 2004 2:29 pm -0600 Steven Critchfield
> >> <critch at basesys.com> wrote:
> >>
> >> > Did you answer() the line first? I'm guessing not since the SIP errors
> >> > is that it can't determine a seqno.
> >> >
> >>
> >> No - thanks for the advice.  it's not fixed the problem but eliminated
> >> some of the error messages, as below.  This is with the agi_app from the
> >> current CVS.  I've had no success with going back to versions of app_agi
> >> from way back in Novemver.
> >>
> >>   Iain
> >>
> >>
> >>
> >>
> >>  AGI Environment Dump:
> >>  -- accountcode =
> >>  -- callerid = "Iain" <15>
> >>  -- channel = SIP/grandstream-78fe
> >>  -- context = voip-sip
> >>  -- dnid = unknown
> >>  -- enhanced = 0.0
> >>  -- extension = 35
> >>  -- language = en
> >>  -- priority = 2
> >>  -- rdnis = unknown
> >>  -- request = agi-test.agi
> >>  -- type = sip
> >>  -- uniqueid = 1073079582.1
> >> 1.  Testing 'sendfile'...PASS (0)
> >> 2.  Testing 'sendtext'...PASS (0)
> >> 3.  Testing 'sendimage'...PASS (0)
> >> 4.  Testing 'saynumber'...WARNING[6151]: File chan_sip.c, Line 464
> >> (retrans_pkt): Maximum retries exceeded on call
> >> d3f5b4e1-a8f1-e86b-2feb-6af1d4a928fd at 192.168.1.242 for seqno 102
> >> (Request) == Spawn extension (voip-sip, 35, 2) exited non-zero on
> >> 'SIP/grandstream-78fe'
> >> PASS (-1)
> >> 5.  Testing 'waitdtmf'...FAIL (unexpected result '')
> >> 6.  Testing 'record'...FAIL (unexpected result '')
> >> 6a.  Testing 'record' playback...FAIL (unexpected result '')
> >> ================== Complete ======================
> >> 7 tests completed, 4 passed, 3 failed
> >>
> >> _______________________________________________
> >> Asterisk-Dev mailing list
> >> Asterisk-Dev at lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-dev
> >>
> >
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> 
> 
> 
> 
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-- 
Steven Critchfield  <critch at basesys.com>