[Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

[Freddy Setiawan]

Thanks. Gonna try today. 

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Neill
Wilkinson
Sent: Monday, June 26, 2006 3:54 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

 

 

In the quintum also check you have a codec profile:

 

Example (below has alaw and G729 configure in the codec profile):

CodecProfile-default :

  name                         : (Not Set)           name                

  VoiceCodecAttached[1]        : VoiceCodec-1

  VoiceCodecAttached[2]        : VoiceCodec-2

  VoiceCodecAttached[3..8]     : (unspecified)

 

CodecProfile-default :

  name                         : (Not Set)           name                

  VoiceCodecAttached[1]        : VoiceCodec-1

  VoiceCodecAttached[2]        : VoiceCodec-2

  VoiceCodecAttached[3..8]     : (unspecified)

 

config-VoiceCodec-2* show

 

VoiceCodec-2 :

  name                         : (Not Set)           name                

  CodecVoiceCoding             : 8                   G.711 A-Law         

  CodecPayloadSize             : 1280                bits                

 

config-VoiceCodec-2*

 

 

this profile should be attached to you IP Routing Group (IPRG).

 

Neill..;o)

====================================

Try Ulaw.

 

Found description format h723

Capabilities: us - 0x100 (h723), peer - audio=0x100 (h723)/video=0x0
(nothing), combined - 0x100 (h723) Non-codec capabilities: us - 0x1
(telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)

 

 

 

> Hello,

> 

> I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk 

> trunk as followed:

> 

> [SIP_BD1]

> type=peer

> qualify=yes

> host=192.168.0.254

> disallow=all

> context=from-pstn

> allow=h723

> 

> and inside the quantum I change the config sip useragent to 5060. Up 

> to this part if I run sip show peers, I got:

> 

> asterisk1*CLI> sip show peers

> Name/username              Host            Dyn Nat ACL Port     Status

> SIP_BD1                    192.168.0.254               5060     OK (56 ms)

> 

> Which seems that I can connect to the quantum A800, but when ever I 

> tried to call I can_t get the phone connected. I mean the destination 

> phone was ring and picked up, but on the pap2 device I didn_t hear any 

> voice, as the destination phone also doesn_t heard any voice.

> 

> Followed are my sip debug for the SIP_BD1:

 

 

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