[asterisk-users] Telecom HU cannot callforward to external number

Wed Jul 18 06:35:11 CDT 2012

Unfortunately not, I already tried different forms callerid(num). Always
the same error.

I came across this entry in asterisk changelogs - maybe an update of
asterisk will help.

* Asterisk 1.4.36-rc1 Released.

2010-08-20 16:46 +0000 [r283048-283123]  Richard Mudgett <rmudgett at digium.com>

	* channels/chan_dahdi.c: Merged revision 278274 from
	  https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
	  | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
	  line Reference correct struct member for unlikely event
	  PRI_EVENT_CONFIG_ERR. ..........

	* channels/chan_dahdi.c: Q931 - Sending PROGRESS after sending
	  ALERTING is a protocol error The PRI layer in chan_dadhi will
	  check if a PROGRESS message has already been sent, and not allow
	  sending another (although that is technically allowed by the Q931
	  spec), however it does not protect against sending an ALERTING
	  and then sending a PROGRESS message, which is a violation of the
	  specification. Most switches don't seem to care too deeply about
	  this, but some do, and will disconnect the call when receiving
	  this invalid sequence. Protocol specification reference:
	  T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
	  protocol control (network side) point-point (sheet 3 of 8)"
	  (closes issue #17874) Reported by: nic_bellamy Patches:
	  asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
	  nic bellamy (license 299)
	  asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
	  by nic bellamy (license 299)
	  asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
	  by nic bellamy (license 299)

thx
christian

On 18 July 2012 13:07, Mitul Limbani <mitul at enterux.in> wrote:

> Mebe your operator doesnt like the CallerID(num) set as NULL just remove
> the callerid(num) statement and let the standard callerId get set by
> network.
>
> Regards,
> Mitul Limbani,
> Chief Architech & Founder,
> Enterux Solutions Pvt. Ltd.
> 110 Reena Complex, Opp. Nathani Steel,
> Vidyavihar (W), Mumbai - 400 086. India
> http://www.enterux.com/
> http://www.entvoice.com/
> email: mitul at enterux.in
> DID: +91-22-61447605
> Cell: +91-9820332422
>
>
>
>
> On Wed, Jul 18, 2012 at 4:23 PM, Christian Gansberger <
> christian.gansberger at accm.at> wrote:
>
>> Hi List!
>>
>> I have a Problem with Telecom Hungary, if I set a callforwarding on the
>> Snom, to an external number (mobile).
>> Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0,
>> snom-7.7.30
>>
>> When I call the Snom (Extension 68), it responds with "302 moved
>> temporarily", and Asterisk try to dial out over the LOCAL channel using
>> DAHDI.
>> I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup
>> request, cause 21
>>
>>
>> Here is cli output:
>>
>>       -- Accepting call from 'callerid' to '68' on channel 0/1, span 1
>>      -- Executing [s at macro-station-fallback-Q-VM:5] Dial("DAHDI/1-1",
>> "SIP/68|15|tTW") in new stack
>>     -- Called 68
>>
>>     -- Got SIP response 302 "Moved Temporarily" back from 10.70.x.xxx
>>
>>     -- Now forwarding DAHDI/1-1 to 'Local/*1mobilenr at snom68' (thanks to
>> SIP/68-000076b8)
>>     -- Executing [*1mobilenr at snom68:1]
>> Macro("Local/*1mobilenr at snom68-2fe3,2",
>> "dialout-dahdi-test|mobilenr|g1|") in new stack
>>     -- Executing [s at macro-dialout-dahdi-test:1]
>> Set("Local/*1mobilenr at snom68-2fe3,2", "CALLERID(number)=") in new stack
>>     -- Executing [s at macro-dialout-dahdi-test:2]
>> Dial("Local/*1mobilenr at snom68-2fe3,2", "DAHDI/g1/mobilenr||") in new
>> stack
>>     -- Requested transfer capability: 0x00 - SPEECH
>>     -- Called g1/mobilnr
>>     -- DAHDI/2-1 is proceeding passing it to Local/*1mobilenr at snom68-2fe3
>> ,2
>>
>>     -- Local/*1mobilenr at snom68-2fe3,1 is proceeding passing it to
>> DAHDI/1-1
>>
>>     -- Channel 0/2, span 1 got hangup request, cause 21
>>
>>     -- DAHDI/2-1 is circuit-busy
>>
>>     -- Hungup 'DAHDI/2-1'
>>
>>   == Everyone is busy/congested at this time (1:0/1/0)
>>
>>
>> I have also an output from "pri intense debug" - But I think the Telecom
>> is just not accepting the outgoing call.
>> What do you think?
>>
>>
>> thanks
>> yours
>> christian
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> --
>> _____________________________________________________________________
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
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