[asterisk-users] FAX

[Danny Nicholas]

On the extension (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 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX

 

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.

 

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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.

 

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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.

 

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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: