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SIP Compatibility between Vigorphone350/Asterisk with MOH
- sppintec
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02 Dec 2010 04:49 #65114
by sppintec
SIP Compatibility between Vigorphone350/Asterisk with MOH was created by sppintec
Hello everybody,
I use the firmware v12202.26.1.06 on my VigorPHone. My IPPBX is an Asterisk 1.6.
It works fine but there are just a problem with the MOH. The caller hears nothing (or awful periodic sound) when I do a transfert or when I press the hold button. Nothing changes when I fix the option Music On Hold on disable or enable.
My IPPBX works perfectly with other phones.
Here you can see SIP trace (during between 1007 and 1005, 1007 press hold button)
Thanks,
Sppintec.
I use the firmware v12202.26.1.06 on my VigorPHone. My IPPBX is an Asterisk 1.6.
It works fine but there are just a problem with the MOH. The caller hears nothing (or awful periodic sound) when I do a transfert or when I press the hold button. Nothing changes when I fix the option Music On Hold on disable or enable.
My IPPBX works perfectly with other phones.
Here you can see SIP trace (during between 1007 and 1005, 1007 press hold button)
Code:
<--- SIP read from UDP:192.168.1.210:5060 --->
INVITE sip:1005@192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.210:5060;branch=z9hG4bK1931320272
From: ;tag=3765407264
To: "xxxxxx" ;tag=as2709fab5
Call-ID: 0dd632a97041aded4084a1cf4020b212@192.168.1.250
CSeq: 102 INVITE
Contact:
max-forwards: 70
user-agent: sipagent MAC-00-50-7F-xx-29-B4 V-12202.26.1.06-SIP
Supported: replaces
Content-Type: application/sdp
Content-Length: 151
v=0
o=- 1224224908 654321 IN IP4 192.168.1.210
s=A conversation
c=IN IP4 0.0.0.0
t=0 0
m=audio 10002 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendonly
<------------->
--- (12 headers 8 lines) ---
Sending to 192.168.1.210 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.210:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.210:5060;branch=z9hG4bK1931320272;received=192.168.1.210
From: ;tag=3765407264
To: "xxxxxx" ;tag=as2709fab5
Call-ID: 0dd632a97041aded4084a1cf4020b212@192.168.1.250
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact:
Content-Length: 0
<------------>
Audio is at 192.168.1.250 port 19480
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.1.210:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.210:5060;branch=z9hG4bK1931320272;received=192.168.1.210
From: ;tag=3765407264
To: "xxxxxxx" ;tag=as2709fab5
Call-ID: 0dd632a97041aded4084a1cf4020b212@192.168.1.250
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 1287143708 1287143708 IN IP4 192.168.1.250
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.250
t=0 0
m=audio 19480 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:192.168.1.210:5060 --->
ACK sip:192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.210:5060;branch=z9hG4bK1365896255
From: ;tag=3765407264
To: "xxxxxxx" ;tag=as2709fab5
Call-ID: 0dd632a97041aded4084a1cf4020b212@192.168.1.250
CSeq: 102 ACK
Contact:
max-forwards: 70
user-agent: sipagent MAC-00-50-7F-xx-xx-B4 V-12202.26.1.06-SIP
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Thanks,
Sppintec.
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- sppintec
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06 Dec 2010 22:49 #65175
by sppintec
Replied by sppintec on topic SIP Compatibility between Vigorphone350/Asterisk with MOH
I'm alone with this problem ?
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