[asterisk-r2] Calls dropped by asterisk with R2 Telmex

Tue Jan 19 12:56:59 CST 2010

this is what i got on the full asterisk log at the time the call got dropped

   4134 [Jan 19 10:41:54] DEBUG[11136] chan_dahdi.c: Write returned -1
(Resource temporarily unavailable) on channel 4
   4135 [Jan 19 10:41:54] DEBUG[11136] chan_dahdi.c: Write returned -1
(Resource temporarily unavailable) on channel 4
   4136 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4137 [Jan 19 10:41:54] DEBUG[11136] chan_dahdi.c: Write returned -1
(Resource temporarily unavailable) on channel 4
   4138 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4139 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4140 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4141 [Jan 19 10:41:54] DEBUG[11129] rtp.c: Got RTCP report of 176 bytes
   4142 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4143 [Jan 19 10:41:54] DEBUG[11132] audiohook.c: Audiohook 0xb6c52ef4 has
stale audio in its factories. Flushing them both
   4144 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4145 [Jan 19 10:41:54] DEBUG[11132] rtp.c: Got RTCP report of 208 bytes
   4146 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4147 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4148 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is
disabled (option_internal_timing=0 chan->timingfd=51)
   4149 [Jan 19 10:41:54] DEBUG[11129] audiohook.c: Failed to get 160
samples from write factory 0xb6c53968
   4150 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: = No match Their Call ID:
OWE1NDNkODE1OWIwMjI3Zjk0NjFlMmZlN2U0YjEzNmM. Their         Tag 007b5a64 Our
tag: as4de61d80
   4151 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: = Found Their Call ID:
NDI5ODhlNDgxYzNhODM4ZTQ3YTg3ZGNiZTkwZWU0Zjg. Their Ta        g 5573a07e Our
tag: as355bacb9
   4152 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: **** Received BYE (8) -
Command in SIP BYE
   4153 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: Setting SIP_ALREADYGONE on
dialog NDI5ODhlNDgxYzNhODM4ZTQ3YTg3ZGNiZTkwZWU0Zj        g.
   4154 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: Received bye, issuing
owner hangup
   4155 [Jan 19 10:41:54] DEBUG[11132] channel.c: Didn't get a frame from
channel: SIP/1011-0000007b
   4156 [Jan 19 10:41:54] DEBUG[11132] chan_dahdi.c: Requested indication 20
on channel DAHDI/5-1
   4157 [Jan 19 10:41:54] DEBUG[11132] channel.c: Bridge stops bridging
channels SIP/1011-0000007b and DAHDI/5-1
   4158 [Jan 19 10:41:54] DEBUG[11132] pbx.c: Launching 'Macro'
   4159 [Jan 19 10:41:54] VERBOSE[11132] logger.c:     -- Executing
[h at macro-dialout-trunk:1] Macro("SIP/1011-0000007b", "hang        upcall|")
in new stack
   4160 [Jan 19 10:41:54] DEBUG[11132] pbx.c: Expression result is '1'
   4161 [Jan 19 10:41:54] DEBUG[11132] pbx.c: Launching 'GotoIf'
   4162 [Jan 19 10:41:54] VERBOSE[11132] logger.c:     -- Executing
[s at macro-hangupcall:1] GotoIf("SIP/1011-0000007b", "1?skip        rg") in
new stack
   4163 [Jan 19 10:41:54] VERBOSE[11132] logger.c:     -- Goto
(macro-hangupcall,s,4)

2010/1/19 Moises Silva <moises.silva at gmail.com>

> On Tue, Jan 19, 2010 at 11:46 AM, blackgecko <blackgecko at gmail.com> wrote:
>
>> ive done test and the calls get hanged up after 30 seconds, i dont know if
>> this helps to identify what the problem can be.
>>
>
> The protocol file does not show any problems. You need to start debugging
> this at a higher layer (Asterisk) it seems to me Asterisk decides to hangup
> the call, and openr2 just go ahead and hangup the call using R2 signaling,
> so Asterisk logs are needed with full debug enabled in order to see where
> the hangup comes from.
>
> --
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> --
> _____________________________________________________________________
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