Hi Carter I got a 3102 too.... Would you mind share your sitting with me? Currently I have a problem about calling from my sip to my bell line. I have two sip account acanac and voipstunt. When I called from acanac, I can get 3102 forward the call to voip. But when I using the voipstunt, the phone rings....
Here is the config [acanac] callerid=905xxxxxxx context=from-trunk host=voip5.acanac.com insecure=very nat=yes qualify=yes secret=xxxxx type=peer username=905xxxxxxx [voipstunt] context=from-trunk host=sip.VoipStunt.com qualify=yes secret=xxx type=peer username=xxx debug info 1.20 is my 3102 1.5 is trixbox trixbox1*CLI> <--- SIP read from 192.168.1.20:5061 ---> INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5061;branch=z9hG4bK-4b2d3376 From: <sip:[EMAIL PROTECTED]>;tag=ce30f578bf0771ao1 To: <sip:[EMAIL PROTECTED]:5060> Remote-Party-ID: <sip:[EMAIL PROTECTED]>;screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="2003",realm="asterisk",nonce="41b07bbf",uri="sip:[EMAIL PROTECTED]:5060",algorithm=MD5,response="ca38010c9e69b86984d80c2bea743ab5" Contact: <sip:[EMAIL PROTECTED]:5061> Expires: 240 User-Agent: Linksys/SPA3102-5.1.7(GW) Content-Length: 440 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 31667 31667 IN IP4 192.168.1.20 s=- c=IN IP4 192.168.1.20 t=0 0 m=audio 16462 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (16 headers 20 lines) --- Sending to 192.168.1.20 : 5061 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found user '905xxxxxxx' <--- Reliably Transmitting (no NAT) to 192.168.1.20:5061 ---> SIP/2.0 403 Forbidden $$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$ why here is failed? Via: SIP/2.0/UDP 192.168.1.20:5061;branch=z9hG4bK-4b2d3376;received=192.168.1.20 From: <sip:[EMAIL PROTECTED]>;tag=ce30f578bf0771ao1 To: <sip:[EMAIL PROTECTED]:5060>;tag=as76686589 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> ****************************That's fromvoipstunt************************ trixbox1*CLI> <--- SIP read from 192.168.1.20:5061 ---> INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5061;branch=z9hG4bK-7e318af6 From: <sip:[EMAIL PROTECTED]>;tag=d81b186d4875988eo1 To: <sip:[EMAIL PROTECTED]:5060> Remote-Party-ID: <sip:[EMAIL PROTECTED]>;screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="2003",realm="asterisk",nonce="0db3b99c",uri="sip:[EMAIL PROTECTED]:5060",algorithm=MD5,response="e5a591c240568d9a6a421bf6cff34bd1" Contact: <sip:[EMAIL PROTECTED]:5061> Expires: 240 User-Agent: Linksys/SPA3102-5.1.7(GW) Content-Length: 440 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 60312 60312 IN IP4 192.168.1.20 s=- c=IN IP4 192.168.1.20 t=0 0 m=audio 16464 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (16 headers 20 lines) --- Sending to 192.168.1.20 : 5061 (NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found user '2003' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.20:16464 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G723 for ID 4 Found audio description format PCMA for ID 8 Found audio description format G729a for ID 18 Found unknown media description format G726-40 for ID 96 Found unknown media description format G726-24 for ID 97 Found unknown media description format G726-16 for ID 98 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.20:16464 Looking for 905xxxxxxx in from-internal (domain 192.168.1.5) list_route: hop: <sip:[EMAIL PROTECTED]:5061> Thanks Alex On Sun, 13 Jan 2008 16:20:43 -0500 Syd Carter <[EMAIL PROTECTED]> wrote: > Alex Wang wrote: > > Hi All > > > > I am using one of the FXO card from x100p.com and I have 40% chance not > > get Caller ID on my trixbox system. Any one got same experience with Bell > > line and x100p card? > > > > > > > > Thanks > > > > > > Alex > > > > > > > > > > > > --------------------------------------------------------------------- > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > For additional commands, e-mail: [EMAIL PROTECTED] > > > > > > > > > > > Yup.. caller id is somewhat the problem with the X100P for me too. I've > tried so many different settings. Seems that calls using a single ring > work fine. Those with a double ring, ie: long distance, don't. I've > given up and use a SPA 3000 fxo to xmit the caller id info for me. X100 > will detect distinctive ring fine, but that doesn't change the fact that > it cannot interpret caller id (atleast with all the zapata.conf settings > I've tried). If you find something that works then let me know. thanks.. > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
