|
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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