[asterisk-users] FAX

[Danny Nicholas]

Got too cute.  Make AST_FORMAT_H100 be AST_FORMAT_H263.

 

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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 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX

 

Dear Danny,

i got the following error during make

   [CC] rtp.c -> rtp.o
rtp.c:1390:3: error: invalid preprocessing directive #[
rtp.c:1391: error: ‘AST_FORMAT_H100’ undeclared here (not in a function)
rtp.c:1392: error: expected ‘}’ before ‘[’ token
make[1]: *** [rtp.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-1.4.22.1/main'
make: *** [main] Error 2

regards

On Thu, Jan 29, 2009 at 9:48 PM, Danny Nicholas <danny at debsinc.com> wrote:

Good new is "your'e getting somewhere".  Bad new is – you have to modify rtp.c to allow this codec.  You should be able to duplicate a line (around 1390) and change the value from 

[34] = {1, AST_FORMAT_H263},

To 

[100] = {1, AST_FORMAT_H100},

 

Then just do a make && make install on asterisk again.

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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 1:35 PM


To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX

 

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.

 

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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.

 

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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 3:09 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: