[asterisk-users] FAX

michel freiha michofr at gmail.com
Thu Jan 29 13:34:44 CST 2009


I'm getting now the below notice:

rtp.c: Unknown RTP codec 100 received from 'GW address'

On Thu, Jan 29, 2009 at 9:18 PM, michel freiha <michofr at gmail.com> wrote:

> Do you mean call limit on the extension or on the outgoing gateway? Kindly
> note that my outbound dialpeer has meeb defined as follow:
>
> [outbound]
> exten => _X.,1,Dial(SIP/${EXTEN}@Outbound_GW,60)
> Regards
>
>
> On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas <danny at debsinc.com> wrote:
>
>>  Doesn't matter – the call-limit is important because 1 call can actually
>> be 2-N hops.
>>
>>
>>  ------------------------------
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *michel freiha
>> *Sent:* Thursday, January 29, 2009 12:45 PM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] FAX
>>
>>
>>
>> Dear Danny,
>>
>>
>>
>> This is the only call on asterisk...:)
>>
>>
>>
>> Regards
>>
>> On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas <danny at debsinc.com>
>> wrote:
>>
>> Try increasing (or adding) call-limit on sip.conf.
>>
>>
>>  ------------------------------
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *michel freiha
>> *Sent:* Thursday, January 29, 2009 12:27 PM
>>
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] FAX
>>
>>
>>
>> Dear Sir,
>>
>>
>>
>> When trying to send a FAX with T.38I got the following error message
>>
>>
>>
>>
>> [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on
>> transmission 3058f601-475045fb at 14.14.14.49 for seqno 102 (Critical
>> Response) -- See doc/sip-retransmit.txt.
>> [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call
>> 3058f601-475045fb at 14.14.14.49 - no reply to our critical packet (see
>> doc/sip-retransmit.txt).
>>
>>
>>
>>
>>
>> Regards
>>
>> On Thu, Jan 29, 2009 at 12:04 AM, michel freiha <michofr at gmail.com>
>> wrote:
>>
>> Dear Danny,
>>
>>
>>
>> Thanks a lot for the help...I'll try and let you know
>>
>>
>>
>> Regards
>>
>> On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas <danny at debsinc.com>
>> wrote:
>>
>> You need to determine what codecs are expected (sip set debug on from
>> CLI).  Commenting out the disallow=all lets * use any available codecs, but
>> may slow down the process or cause undesirable results by using/accounting
>> for unneeded or unwanted codecs.
>>
>>
>>  ------------------------------
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *michel freiha
>> *Sent:* Wednesday, January 28, 2009 3:32 PM
>>
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] FAX
>>
>>
>>
>> Dear Sir,
>>
>>
>>
>> What do you mean by manual fax? I need to offer the ability for each
>> extension to use voice and FAX...MAybe the voice will use G729 and the FAX
>> ulaw for the same extension...If I configure the device in a manner that use
>> ulaw for FAX and G729 for voice then this should work smoothly with an
>> extension where G729,ulaw, alaw are allowed?
>>
>>
>>
>> Regards
>>
>> On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas <danny at debsinc.com>
>> wrote:
>>
>> The codecs should only be needed for a "manual" fax, where a voice
>> interaction might be expected or anticipated.
>>
>>
>>  ------------------------------
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *michel freiha
>> *Sent:* Wednesday, January 28, 2009 3:09 PM
>>
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] FAX
>>
>>
>>
>> Dear Sir,
>>
>> If I commant all codecs including disallow=all, then which codec should I
>> define on the extensions from where I'm trying to send FAX?
>>
>> Regards
>>
>> On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas <danny at debsinc.com>
>> wrote:
>>
>> From your sip.conf, you are only allowing ulaw and alaw codes.  I'd try
>> adding gsm or just comment out the disallow and the 2 allows.  (your
>> recipient is using a codec that isn't ulaw or alaw).
>>
>>
>>  ------------------------------
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *michel freiha
>> *Sent:* Wednesday, January 28, 2009 2:21 PM
>>
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>
>> *Subject:* Re: [asterisk-users] FAX
>>
>>
>>
>> Dear SIr,
>>
>> please find attached my sip.conf file
>>
>> Regards
>>
>> On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com>
>> wrote:
>>
>> Show us your sip.conf
>>
>>
>>  ------------------------------
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *michel freiha
>> *Sent:* Wednesday, January 28, 2009 9:30 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] FAX
>>
>>
>>
>> Hi all,
>>
>> When trying to send a FAX I got the following error:
>>
>> Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/
>> 003228949469 at 80.169.210.181|60") in new stack
>> [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio
>> format found to offer. Cancelling call to 003228949469
>>     -- Couldn't call 0032234534534 at 1.1.1.1.1
>>
>> Where I should define the codec other than the extension in order to
>> succeed the call?
>>
>> Regards
>>
>>
>> _______________________________________________
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>>
>>
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>
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