vijay.goyal at alliance-infotech.com
2009-Dec-30 19:43 UTC
[asterisk-users] Skype for Asterisk
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 [s at skypeincoming:1]
Answer("Skype/rexesbposolutions-084159e8", "") in new stack
-- Executing [s at skypeincoming:2]
Wait("Skype/rexesbposolutions-084159e8", "5") in new stack
-- Executing [s at skypeincoming:3]
GotoIfTime("Skype/rexesbposolutions-084159e8",
"9:00-18:00|mon-fri|*|
*?sky|s|1") in new stack
-- Goto (sky,s,1)
-- Executing [s at 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 [s at 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 [s at skypeincoming:1]
Answer("Skype/rexesbposolutions-084159e8", "") in new stack
-- Executing [s at skypeincoming:2]
Wait("Skype/rexesbposolutions-084159e8", "5") in new stack
-- Executing [s at skypeincoming:3]
GotoIfTime("Skype/rexesbposolutions-084159e8",
"9:00-18:00|mon-fri|*|
*?sky|s|1") in new stack
-- Goto (sky,s,1)
-- Executing [s at 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 [s at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20091230/47b16e9c/attachment.htm
On 30 Dec 2009, at 19:43, vijay.goyal at 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 [s at skypeincoming:1] Answer("Skype/rexesbposolutions-084159e8", "") in new stack > -- Executing [s at skypeincoming:2] Wait("Skype/rexesbposolutions-084159e8", "5") in new stack > -- Executing [s at skypeincoming:3] GotoIfTime("Skype/rexesbposolutions-084159e8", "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack > -- Goto (sky,s,1) > -- Executing [s at 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 [s at 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 [s at skypeincoming:1] Answer("Skype/rexesbposolutions-084159e8", "") in new stack > -- Executing [s at skypeincoming:2] Wait("Skype/rexesbposolutions-084159e8", "5") in new stack > -- Executing [s at skypeincoming:3] GotoIfTime("Skype/rexesbposolutions-084159e8", "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack > -- Goto (sky,s,1) > -- Executing [s at 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 [s at 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-usersIt 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100104/5336824c/attachment-0001.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2419 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100104/5336824c/attachment-0001.bin
On Wed, 30 Dec 2009 11:43:59 -0800, vijay.goyal at alliance-infotech.com wrote:>>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:I had the same problem. I contacted Digium support and this was the answer: You can download the G.729 codec from the following link: http://www.digium.com/en/docs/G729/g729-download.php Install the codec_g729a.so binary in /usr/lib/asterisk/modules/ and restart the asterisk service. I followed the advice and the problem is resolved. You can try. Bye