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 <da...@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.
> ------------------------------
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 <mich...@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/${ext...@outbound_gw,60)
> Regards
>
>
>
> On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas <da...@debsinc.com> wrote:
>
> Doesn't matter – the call-limit is important because 1 call can actually be
> 2-N hops.
>
>
> ------------------------------
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 <da...@debsinc.com> wrote:
>
> Try increasing (or adding) call-limit on sip.conf.
>
>
> ------------------------------
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 3058f601-47504...@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
> 3058f601-47504...@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 <mich...@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 <da...@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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 <da...@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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 <da...@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-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 <da...@debsinc.com> wrote:
>
> Show us your sip.conf
>
>
> ------------------------------
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@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 [003228949...@micho:1] Dial("SIP/028949469-08466918", "SIP/
> 003228949...@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 0032234534...@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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