<div dir="ltr">Dear,<br> there is a problem in codec translation..so change the ulaw codec to g729. .if problem persist then u must have same codex on asterisk server and clients (skype)...<br><br><div class="gmail_quote">
On Mon, Jan 4, 2010 at 11:24 AM, Tim Panton <span dir="ltr"><<a href="mailto:thp@westhawk.co.uk">thp@westhawk.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style="word-wrap: break-word;"><br><div><div>On 30 Dec 2009, at 19:43, <a href="mailto:vijay.goyal@alliance-infotech.com" target="_blank">vijay.goyal@alliance-infotech.com</a> wrote:</div><br><blockquote type="cite">
<div>
<br>
<font color="#000000">Hi Sir,</font><br>
<br>
<font color="#000000">We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases:</font><br>
<br>
<font color="#000000">case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine.</font><br>
<br>
<font color="#000000">case 2: This skype account (rexesbposolutions) has been assigned with a online virtual number (00 44 20 *</font>***<font color="#000000"> ****). If somebody dial this number from their landline/cellphone, call is transfered to Asterisk queue but it shows some problem related to G729 codecs. following are Asterisk CLI log:</font><br>
<br>
<font color="#333399"> </font><font color="#008080">Executing [s@skypeincoming:1] Answer("Skype/rexesbposolutions-084159e8", "") in new stack</font><br>
<font color="#008080"> -- Executing [s@skypeincoming:2] Wait("Skype/rexesbposolutions-084159e8", "5") in new stack</font><br>
<font color="#008080"> -- Executing [s@skypeincoming:3] GotoIfTime("Skype/rexesbposolutions-084159e8", "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack</font><br>
<font color="#008080"> -- Goto (sky,s,1)</font><br>
<font color="#008080"> -- Executing [s@sky:1] Playback("Skype/rexesbposolutions-084159e8", "enter") in new stack</font><br>
<font color="#008080"> -- <Skype/rexesbposolutions-084159e8> Playing 'enter' (language 'en')</font><br>
<font color="#008080">[Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw)</font><br>
<font color="#008080">[Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 4</font><br>
<font color="#008080"> -- Executing [s@sky:2] Queue("Skype/rexesbposolutions-084159e8", "markq|t|||900") in new stack</font><br>
<font color="#008080"> -- Started music on hold, class 'default', on Skype/rexesbposolutions-084159e8</font><br>
<font color="#008080">[Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)</font><br>
<font color="#008080">[Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:251 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/manolo_camp-morning_coffee': No such file or directory</font><br>
<font color="#008080"> -- Stopped music on hold on Skype/rexesbposolutions-084159e8</font><br>
<font color="#008080">[Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)</font><br>
<font color="#008080">[Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release: Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2'</font><br>
<font color="#008080"> -- Playing periodic announcement</font><br>
<font color="#008080">[Dec 29 17:27:34] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)</font><br>
<font color="#008080"> -- <Skype/rexesbposolutions-084159e8> Playing 'queue' (language 'en')</font><br>
<font color="#008080">[Dec 29 17:27:43] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)</font><br>
<font color="#008080">[Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 2</font><br>
<font color="#008080"> == Spawn extension (sky, s, 2) exited non-zero on 'Skype/rexesbposolutions-084159e8'</font><br>
<font color="#008080">[Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call</font><br>
<br>
<br>
<br>
following are output of some commands:-<br>
<br>
*CLI> core show translation<br>
<br>
Translation times between formats (in milliseconds) for one second of data<br>
Source Format (Rows) Destination Format (Columns)<br>
<br>
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722<br>
g723 - - - - - - - - - - - - -<br>
gsm - - 2 2 2 2 1 2 6 - - 2 -<br>
ulaw - 2 - 1 2 2 1 2 6 - - 2 -<br>
alaw - 2 1 - 2 2 1 2 6 - - 2 -<br>
g726aal2 - 2 2 2 - 2 1 2 6 - - 2 -<br>
adpcm - 2 2 2 2 - 1 2 6 - - 2 -<br>
slin - 1 1 1 1 1 - 1 5 - - 1 -<br>
lpc10 - 2 2 2 2 2 1 - 6 - - 2 -<br>
g729 - 6 6 6 6 6 5 6 - - - 6 -<br>
speex - - - - - - - - - - - - -<br>
ilbc - - - - - - - - - - - - -<br>
g726 - 2 2 2 2 2 1 2 6 - - - -<br>
g722 - - - - - - - - - - - - -<br>
<br>
<br>
*CLI> help g729<br>
g729 show hostid Show G.729 Host-ID<br>
g729 show licenses Show G.729 Licenses and Usage<br>
g729 show version Show G.729 Module Version<br>
<br>
*CLI> g729 show hostid<br>
Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be<br>
<br>
*CLI> g729 show licenses<br>
0/0 encoders/decoders of 1 licensed channels are currently in use<br>
<br>
Licenses Found:<br>
File: ***-*************.lic -- Key: ***-************* -- Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be -- Channels: 1 (Expires: 2029-11-30) (OK)<br>
<br>
*CLI> g729 show version<br>
Digium G.729A Module Version 1.4_3.1.4 (optimized for i686_32)<br>
<br>
<br>
*CLI> core show codecs<br>
Disclaimer: this command is for informational purposes only.<br>
It does not indicate anything about your configuration.<br>
INT BINARY HEX TYPE NAME DESC<br>
--------------------------------------------------------------------------------<br>
1 (1 << 0) (0x1) audio g723 (G.723.1)<br>
2 (1 << 1) (0x2) audio gsm (GSM)<br>
4 (1 << 2) (0x4) audio ulaw (G.711 u-law)<br>
8 (1 << 3) (0x8) audio alaw (G.711 A-law)<br>
16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2)<br>
32 (1 << 5) (0x20) audio adpcm (ADPCM)<br>
64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM)<br>
128 (1 << 7) (0x80) audio lpc10 (LPC10)<br>
256 (1 << 8) (0x100) audio g729 (G.729A)<br>
512 (1 << 9) (0x200) audio speex (SpeeX)<br>
1024 (1 << 10) (0x400) audio ilbc (iLBC)<br>
2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551)<br>
4096 (1 << 12) (0x1000) audio g722 (G722)<br>
65536 (1 << 16) (0x10000) image jpeg (JPEG image)<br>
131072 (1 << 17) (0x20000) image png (PNG image)<br>
262144 (1 << 18) (0x40000) video h261 (H.261 Video)<br>
524288 (1 << 19) (0x80000) video h263 (H.263 Video)<br>
1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)<br>
2097152 (1 << 21) (0x200000) video h264 (H.264 Video)<br>
<br>
<br>
Asterisk CLI logs:-<br>
<br>
*************************************************************************************************<br>
<br>
func_logic.so => (Logical dialplan functions)<br>
[Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:755 load_module: G.729A transcoding module version 1.4_3.1.4, Copyright (C) 1999-2009 Digium, Inc.<br>
[Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:756 load_module: This module is supplied under a co mmercial license granted by Digium, Inc.<br>
[Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:757 load_module: Please see the full license text s upplied by the accompanying<br>
[Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:758 load_module: "register" utility, or ask for a c opy from Digium.<br>
[Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:763 load_module: This product includes software dev eloped by the OpenSSL Project<br>
[Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:764 load_module: for use in the OpenSSL Toolkit. (h ttp://<a href="http://www.openssl.org/%29" target="_blank">www.openssl.org/)</a><br>
[Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:765 load_module: Copyright (C) 1998-2006 The OpenSS L Project<br>
<br>
== Manager registered action G729LicenseStatus<br>
== Manager registered action G729LicenseList<br>
== Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be<br>
== Found license 'S4A-UGMS4JZXQMDE' providing 1 channels<br>
== Found total of 1 G.729 licenses<br>
== Registered translator 'g729tolin' from format g729 to slin, cost 1<br>
== Registered translator 'lintog729' from format slin to g729, cost 5<br>
codec_g729a.so => (Digium G.729 Annex A Codec (optimized for i686_32))<br>
== Registered application 'Flash'<br>
app_flash.so => (Flash channel application)<br>
== Registered file format iLBC, extension(s) ilbc<br>
<br>
*************************************************************************************************<br>
<br>
<br>
*CLI> Executing [s@skypeincoming:1] Answer("Skype/rexesbposolutions-084159e8", "") in new stack<br>
-- Executing [s@skypeincoming:2] Wait("Skype/rexesbposolutions-084159e8", "5") in new stack<br>
-- Executing [s@skypeincoming:3] GotoIfTime("Skype/rexesbposolutions-084159e8", "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack<br>
-- Goto (sky,s,1)<br>
-- Executing [s@sky:1] Playback("Skype/rexesbposolutions-084159e8", "enter") in new stack<br>
-- <Skype/rexesbposolutions-084159e8> Playing 'enter' (language 'en')<br>
[Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw)<br>
[Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 4<br>
-- Executing [s@sky:2] Queue("Skype/rexesbposolutions-084159e8", "markq|t|||900") in new stack<br>
-- Started music on hold, class 'default', on Skype/rexesbposolutions-084159e8<br>
[Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)<br>
[Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:251 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/manolo_camp-morning_coffee': No such file or directory<br>
-- Stopped music on hold on Skype/rexesbposolutions-084159e8<br>
[Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)<br>
[Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release: Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2'<br>
-- Playing periodic announcement<br>
[Dec 29 17:27:34] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)<br>
-- <Skype/rexesbposolutions-084159e8> Playing 'queue' (language 'en')<br>
[Dec 29 17:27:43] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)<br>
[Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 2<br>
== Spawn extension (sky, s, 2) exited non-zero on 'Skype/rexesbposolutions-084159e8'<br>
[Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call<br>
<br>
<br>
<font color="#000000">Kindly resolve this issue ASAP.</font><br>
<br>
<br>
<table cellpadding="0" cellspacing="0" width="100%">
<tbody><tr>
<td>
<font size="2"><font color="#000000">With Regards</font></font><br>
<br>
<br>
<b><font size="2"><font color="#000080">Vijay Goyal (Software Engineer VOIP)</font></font></b><br>
<font size="2"><font color="#000080">Alliance Infotech Private Limited - Mobility,Convenience,Realization</font></font><br>
<font size="2"><font color="#000080">(An ISO 9001: 2000 certified company)</font></font> <br>
<br>
<font size="2"><font color="#000080">B 254 First Floor, Okhla Industrial Area-I, New Delhi 110 020 (India) | Tel: +91 11 40517731, 2637 1851 | Fax: +91 11 2637 1852, 2981 0953</font></font><br>
<font size="2"><font color="#000080">Digium Select Partner | Dialogic Partner | Microsoft Certified Part</font></font><font color="#000080">ner CRM & </font><font color="#000080"><font size="2">Computer Telephony solutions | Speech Enabled IVRS | Unified Communications | Voice loggers | Audio Conferencing | Web Enabled sol</font></font><font color="#000080">utions </font>
</td>
</tr>
</tbody></table>
</div>
_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><div><br></div><div><br></div><div>It looks to me as if you are running out of 729 licenses. A single call may (sometimes) need more than one license.</div>
<div>You can probably avoid this problem by either:</div><div><span style="white-space: pre;">        </span>1) buying more 729 licenses (just a few more than active channels should do)</div><div><span style="white-space: pre;">        </span>2) using Ulaw in chan_skype (instead of 729)</div>
<div><span style="white-space: pre;">        </span>3) downloading the soundfiles in 729 (you currently only have GSM)</div><div><br></div><div>Do 3) anyway - gsm transcoded to 729 always sounds horrible.</div><div><br></div><div>
Tim.</div><div><br></div><font color="#888888"><br><div>
<span style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;"><div>
Tim Panton - Web/VoIP consultant and implementor</div><div><a href="http://www.westhawk.co.uk" target="_blank">www.westhawk.co.uk</a></div><div><br></div></span><br>
</div>
<br></font></div><br>_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Best Regards<br><br>Yawar Hadi Noshahi<br>
Consultant/Software Engineer<br> NGI Islamabad<br><br>MS Computer Science<br> Linkoping University<br> Sweden<br> +46700-445479<br><br>
</div>