[asterisk-users] FAX

Thu Jan 29 14:11:58 CST 2009

[3] = {1, AST_FORMAT_GSM},
        [4] = {1, AST_FORMAT_G723_1},
        [5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
        [6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
        [7] = {1, AST_FORMAT_LPC10},
        [8] = {1, AST_FORMAT_ALAW},
        [9] = {1, AST_FORMAT_G722},
        [10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
        [11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
        [13] = {0, AST_RTP_CN},
        [16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
        [17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
        [18] = {1, AST_FORMAT_G729A},
        [19] = {0, AST_RTP_CN},         /* Also used for CN */
        [26] = {1, AST_FORMAT_JPEG},
        [31] = {1, AST_FORMAT_H261},
#       [34] = {1, AST_FORMAT_H263},
        [100] = {1, AST_FORMAT_H100}
        [103] = {1, AST_FORMAT_H263_PLUS},
        [97] = {1, AST_FORMAT_ILBC},
        [99] = {1, AST_FORMAT_H264},
        [101] = {0, AST_RTP_DTMF},
        [110] = {1, AST_FORMAT_SPEEX},
        [111] = {1, AST_FORMAT_G726},
        [112] = {1, AST_FORMAT_G726_AAL2},
        [121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */

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.
>  ------------------------------
>
> *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.
>
>
>  ------------------------------
>
> *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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