|
Thanks. Gonna try today. From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Neill Wilkinson 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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