[Asterisk-Dev] AGI still falls over
Iain Stevenson
iain at iainstevenson.com
Fri Jan 2 15:17:11 MST 2004
--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
>
More information about the asterisk-dev
mailing list