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)

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.

I'm alone with this problem ?