Re: [Asterisk-Users] I don't want ilbc, i just want G.711

2005-12-15 Thread Elton Machado
And if, for some very strange reason, it doesn't work, use noload at modules.conf ;)
 
Regards,
 
 
 
2005/12/15, Umair Bari <[EMAIL PROTECTED]>:

in your sip.cong [general] contexts
 
put 
disallow=all
allow=ulaw
allow=alaw
 
and in your sip user, use disallow only ONCE, that is 
disallow=all
allow=ulaw
allow=alawhope this helps.
 
regards,
 
Umair bari
 

On 12/15/05, Jason Chan (jasonOfficial) <
[EMAIL PROTECTED]> wrote: 


   Hi there,I am writing to ask about how to fix the codec to G.711 ONLY.Actually what I am doing is, try to use DTMF when the POTS phone call hasdirected to Asterisk via Planet VIP-450 FXO Port, but this gateway just 
simply doesn't support RFC2833 nor SIP-INFO. The only method I can use isInband DTMF. I know it only support G.711, but I DID disallow others andmake it work only with G.711. But the problem is, although I disallow all 
other codecs, ilbc still itching me...[extensions.conf][852]username=HKGWserect=blahtype=friendhost=dynamicnat =yescanreinvite=nodisallow=alldisallow=ilbcallow=ulawdtmfmode=inband 
(P.S. I don't use REINVITE simply because I need the asterisk to be amedia gateway cause the gateway is inside NAT behind the Asterisk)Whenever I try to pass DTMF from phone to Asterisk via that gateway, I got 
such messages:Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF isnot supported on codec ilbc. Use RFC2833Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh? 
An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?How come!? I DID DISALLOW them, but it keeps bugging me=
192.168.2.3  852 79f9e0-c0a8  00101/1  ulaw  No   Rx:ACK1 active SIP channel*CLI> sip show channel 79  * SIP Call  Direction:  Incoming  Call-ID:   
[EMAIL PROTECTED]  Our Codec Capability:   4 
  Non-Codec Capability:   0  Their Codec Capability:   261  Joint Codec Capability:   4  Format  ulaw  Theoretical Address:    
192.168.2.3:5060  Received Address:   192.168.2.3:5060  NAT Support:    Always  Audio IP:   
192.168.2.1 (local)  Our Tag:    as737358ce  Their Tag:  3a53f3e1-bbfcafe6d5c  SIP User agent: 
  Username:   852  Peername:   852  Original uri:   sip:[EMAIL PROTECTED]:5060  Caller-ID:  elite  Need Destroy:   0  Last Message:   Rx: ACK 
  Promiscuous Redir:  No  Route:  sip:[EMAIL PROTECTED]:5060  DTMF Mode:  inband  SIP Options:    (none)==Previously I installed 1.0.3 in same machine, but I overwrite all files 
with 1.2.1.. does it cause a trouble?Can anyone figure out what is the problem? ==Thanks very much for your help!Best regards, 
Jason Chan, Hong KongNo virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005___ 
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Re: [Asterisk-Users] I don't want ilbc, i just want G.711

2005-12-15 Thread Umair Bari
in your sip.cong [general] contexts
 
put 
disallow=all
allow=ulaw
allow=alaw
 
and in your sip user, use disallow only ONCE, that is 
disallow=all
allow=ulaw
allow=alawhope this helps.
 
regards,
 
Umair bari
 
On 12/15/05, Jason Chan (jasonOfficial) <[EMAIL PROTECTED]> wrote:

   Hi there,I am writing to ask about how to fix the codec to G.711 ONLY.Actually what I am doing is, try to use DTMF when the POTS phone call hasdirected to Asterisk via Planet VIP-450 FXO Port, but this gateway just
simply doesn't support RFC2833 nor SIP-INFO. The only method I can use isInband DTMF. I know it only support G.711, but I DID disallow others andmake it work only with G.711. But the problem is, although I disallow all
other codecs, ilbc still itching me...[extensions.conf][852]username=HKGWserect=blahtype=friendhost=dynamicnat =yescanreinvite=nodisallow=alldisallow=ilbcallow=ulawdtmfmode=inband
(P.S. I don't use REINVITE simply because I need the asterisk to be amedia gateway cause the gateway is inside NAT behind the Asterisk)Whenever I try to pass DTMF from phone to Asterisk via that gateway, I got
such messages:Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF isnot supported on codec ilbc. Use RFC2833Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh? 
An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?How come!? I DID DISALLOW them, but it keeps bugging me=
192.168.2.3  852 79f9e0-c0a8  00101/1  ulaw  No   Rx:ACK1 active SIP channel*CLI> sip show channel 79  * SIP Call  Direction:  Incoming  Call-ID:   
[EMAIL PROTECTED]  Our Codec Capability:   4
  Non-Codec Capability:   0  Their Codec Capability:   261  Joint Codec Capability:   4  Format  ulaw  Theoretical Address:    
192.168.2.3:5060  Received Address:   192.168.2.3:5060  NAT Support:    Always  Audio IP:   
192.168.2.1 (local)  Our Tag:    as737358ce  Their Tag:  3a53f3e1-bbfcafe6d5c  SIP User agent:
  Username:   852  Peername:   852  Original uri:   sip:[EMAIL PROTECTED]:5060  Caller-ID:  elite  Need Destroy:   0  Last Message:   Rx: ACK
  Promiscuous Redir:  No  Route:  sip:[EMAIL PROTECTED]:5060  DTMF Mode:  inband  SIP Options:    (none)==Previously I installed 1.0.3 in same machine, but I overwrite all files
with 1.2.1.. does it cause a trouble?Can anyone figure out what is the problem? ==Thanks very much for your help!Best regards,
Jason Chan, Hong KongNo virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005___
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[Asterisk-Users] I don't want ilbc, i just want G.711

2005-12-14 Thread Jason Chan \(jasonOfficial\)



   Hi there,I am writing to ask 
about how to fix the codec to G.711 ONLY.Actually what I am doing is, try to 
use DTMF when the POTS phone call hasdirected to Asterisk via Planet VIP-450 
FXO Port, but this gateway justsimply doesn't support RFC2833 nor SIP-INFO. 
The only method I can use isInband DTMF. I know it only support G.711, but I 
DID disallow others andmake it work only with G.711. But the problem is, 
although I disallow allother codecs, ilbc still itching 
me...[extensions.conf][852]username=HKGWserect=blahtype=friendhost=dynamicnat 
=yescanreinvite=nodisallow=alldisallow=ilbcallow=ulawdtmfmode=inband(P.S. 
I don't use REINVITE simply because I need the asterisk to be amedia gateway 
cause the gateway is inside NAT behind the Asterisk)Whenever I try to pass 
DTMF from phone to Asterisk via that gateway, I gotsuch messages:Dec 
14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF isnot 
supported on codec ilbc. Use RFC2833Dec 14 23:35:32 WARNING[10958]: 
codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple 
of 50 bytes long from RTP (4)?How come!? I DID DISALLOW them, but it 
keeps bugging me=192.168.2.3  
852 79f9e0-c0a8  
00101/1  ulaw  No   
Rx:ACK1 active SIP channel*CLI> sip show channel 79  
* SIP Call  
Direction:  
Incoming  
Call-ID:   
[EMAIL PROTECTED]  Our Codec Capability:   4  Non-Codec 
Capability:   0  Their Codec Capability:   
261  Joint Codec Capability:   4  
Format  
ulaw  Theoretical Address:    192.168.2.3:5060  
Received Address:   192.168.2.3:5060  
NAT Support:    
Always  Audio 
IP:   
192.168.2.1 (local)  Our 
Tag:    
as737358ce  Their 
Tag:  
3a53f3e1-bbfcafe6d5c  SIP User agent:  
Username:   
852  
Peername:   
852  Original 
uri:   
sip:[EMAIL PROTECTED]:5060  
Caller-ID:  
elite  Need 
Destroy:   0  
Last Message:   Rx: 
ACK  Promiscuous Redir:  No  
Route:  
sip:[EMAIL PROTECTED]:5060  DTMF 
Mode:  
inband  SIP 
Options:    
(none)==Previously I installed 1.0.3 in same machine, but I 
overwrite all fileswith 1.2.1.. does it cause a trouble?Can 
anyone figure out what is the problem? 
==Thanks 
very much for your help!Best regards,Jason Chan, Hong 
Kong
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005
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