On 30 Dec 2009, at 19:43, vijay.go...@alliance-infotech.com wrote:
>
> Hi Sir,
>
> 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:
>
> case 1: When we call from normal skype account to skype account
> (rexesbposolutions), everything is working fine.
>
> case 2: This skype account (rexesbposolutions) has been assigned with a
> online virtual number (00 44 20 **** ****). 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:
>
> Executing [...@skypeincoming:1]
> Answer("Skype/rexesbposolutions-084159e8", "") in new stack
> -- Executing [...@skypeincoming:2]
> Wait("Skype/rexesbposolutions-084159e8", "5") in new stack
> -- Executing [...@skypeincoming:3]
> GotoIfTime("Skype/rexesbposolutions-084159e8",
> "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack
> -- Goto (sky,s,1)
> -- Executing [...@sky:1] Playback("Skype/rexesbposolutions-084159e8",
> "enter") in new stack
> -- <Skype/rexesbposolutions-084159e8> Playing 'enter' (language 'en')
> [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)
> [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore
> format back to 4
> -- Executing [...@sky:2] Queue("Skype/rexesbposolutions-084159e8",
> "markq|t|||900") in new stack
> -- Started music on hold, class 'default', on
> Skype/rexesbposolutions-084159e8
> [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)
> [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
> -- Stopped music on hold on Skype/rexesbposolutions-084159e8
> [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)
> [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release:
> Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2'
> -- Playing periodic announcement
> [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)
> -- <Skype/rexesbposolutions-084159e8> Playing 'queue' (language 'en')
> [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)
> [Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore
> format back to 2
> == Spawn extension (sky, s, 2) exited non-zero on
> 'Skype/rexesbposolutions-084159e8'
> [Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call
>
>
>
> following are output of some commands:-
>
> *CLI> core show translation
>
> Translation times between formats (in milliseconds) for one second of data
> Source Format (Rows) Destination Format (Columns)
>
> g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726
> g722
> g723 - - - - - - - - - - - -
> -
> gsm - - 2 2 2 2 1 2 6 - - 2
> -
> ulaw - 2 - 1 2 2 1 2 6 - - 2
> -
> alaw - 2 1 - 2 2 1 2 6 - - 2
> -
> g726aal2 - 2 2 2 - 2 1 2 6 - - 2
> -
> adpcm - 2 2 2 2 - 1 2 6 - - 2
> -
> slin - 1 1 1 1 1 - 1 5 - - 1
> -
> lpc10 - 2 2 2 2 2 1 - 6 - - 2
> -
> g729 - 6 6 6 6 6 5 6 - - - 6
> -
> speex - - - - - - - - - - - -
> -
> ilbc - - - - - - - - - - - -
> -
> g726 - 2 2 2 2 2 1 2 6 - - -
> -
> g722 - - - - - - - - - - - -
> -
>
>
> *CLI> help g729
> g729 show hostid Show G.729 Host-ID
> g729 show licenses Show G.729 Licenses and Usage
> g729 show version Show G.729 Module Version
>
> *CLI> g729 show hostid
> Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be
>
> *CLI> g729 show licenses
> 0/0 encoders/decoders of 1 licensed channels are currently in use
>
> Licenses Found:
> 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)
>
> *CLI> g729 show version
> Digium G.729A Module Version 1.4_3.1.4 (optimized for i686_32)
>
>
> *CLI> core show codecs
> Disclaimer: this command is for informational purposes only.
> It does not indicate anything about your configuration.
> INT BINARY HEX TYPE NAME DESC
> --------------------------------------------------------------------------------
> 1 (1 << 0) (0x1) audio g723 (G.723.1)
> 2 (1 << 1) (0x2) audio gsm (GSM)
> 4 (1 << 2) (0x4) audio ulaw (G.711 u-law)
> 8 (1 << 3) (0x8) audio alaw (G.711 A-law)
> 16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2)
> 32 (1 << 5) (0x20) audio adpcm (ADPCM)
> 64 (1 << 6) (0x40) audio slin (16 bit Signed Linear
> PCM)
> 128 (1 << 7) (0x80) audio lpc10 (LPC10)
> 256 (1 << 8) (0x100) audio g729 (G.729A)
> 512 (1 << 9) (0x200) audio speex (SpeeX)
> 1024 (1 << 10) (0x400) audio ilbc (iLBC)
> 2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551)
> 4096 (1 << 12) (0x1000) audio g722 (G722)
> 65536 (1 << 16) (0x10000) image jpeg (JPEG image)
> 131072 (1 << 17) (0x20000) image png (PNG image)
> 262144 (1 << 18) (0x40000) video h261 (H.261 Video)
> 524288 (1 << 19) (0x80000) video h263 (H.263 Video)
> 1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)
> 2097152 (1 << 21) (0x200000) video h264 (H.264 Video)
>
>
> Asterisk CLI logs:-
>
> *************************************************************************************************
>
> func_logic.so => (Logical dialplan functions)
> [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.
> [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.
> [Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:757 load_module: Please see the
> full license text s upplied by the accompanying
> [Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:758 load_module: "register"
> utility, or ask for a c opy from Digium.
> [Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:763 load_module: This product
> includes software dev eloped by the OpenSSL Project
> [Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:764 load_module: for use in the
> OpenSSL Toolkit. (h ttp://www.openssl.org/)
> [Dec 29 18:22:17] NOTICE[4282]: codec_g729a.c:765 load_module: Copyright (C)
> 1998-2006 The OpenSS L Project
>
> == Manager registered action G729LicenseStatus
> == Manager registered action G729LicenseList
> == Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be
> == Found license 'S4A-UGMS4JZXQMDE' providing 1 channels
> == Found total of 1 G.729 licenses
> == Registered translator 'g729tolin' from format g729 to slin, cost 1
> == Registered translator 'lintog729' from format slin to g729, cost 5
> codec_g729a.so => (Digium G.729 Annex A Codec (optimized for i686_32))
> == Registered application 'Flash'
> app_flash.so => (Flash channel application)
> == Registered file format iLBC, extension(s) ilbc
>
> *************************************************************************************************
>
>
> *CLI> Executing [...@skypeincoming:1]
> Answer("Skype/rexesbposolutions-084159e8", "") in new stack
> -- Executing [...@skypeincoming:2]
> Wait("Skype/rexesbposolutions-084159e8", "5") in new stack
> -- Executing [...@skypeincoming:3]
> GotoIfTime("Skype/rexesbposolutions-084159e8",
> "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack
> -- Goto (sky,s,1)
> -- Executing [...@sky:1] Playback("Skype/rexesbposolutions-084159e8",
> "enter") in new stack
> -- <Skype/rexesbposolutions-084159e8> Playing 'enter' (language 'en')
> [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)
> [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore
> format back to 4
> -- Executing [...@sky:2] Queue("Skype/rexesbposolutions-084159e8",
> "markq|t|||900") in new stack
> -- Started music on hold, class 'default', on
> Skype/rexesbposolutions-084159e8
> [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)
> [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
> -- Stopped music on hold on Skype/rexesbposolutions-084159e8
> [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)
> [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release:
> Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2'
> -- Playing periodic announcement
> [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)
> -- <Skype/rexesbposolutions-084159e8> Playing 'queue' (language 'en')
> [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)
> [Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore
> format back to 2
> == Spawn extension (sky, s, 2) exited non-zero on
> 'Skype/rexesbposolutions-084159e8'
> [Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call
>
>
> Kindly resolve this issue ASAP.
>
>
> With Regards
>
>
> Vijay Goyal (Software Engineer VOIP)
> Alliance Infotech Private Limited - Mobility,Convenience,Realization
> (An ISO 9001: 2000 certified company)
>
> 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
> Digium Select Partner | Dialogic Partner | Microsoft Certified Partner CRM
> & Computer Telephony solutions | Speech Enabled IVRS | Unified
> Communications | Voice loggers | Audio Conferencing | Web Enabled solutions
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
It looks to me as if you are running out of 729 licenses. A single call may
(sometimes) need more than one license.
You can probably avoid this problem by either:
1) buying more 729 licenses (just a few more than active channels
should do)
2) using Ulaw in chan_skype (instead of 729)
3) downloading the soundfiles in 729 (you currently only have GSM)
Do 3) anyway - gsm transcoded to 729 always sounds horrible.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
smime.p7s
Description: S/MIME cryptographic signature
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
