On 2003-03-12 at 09:44, you wrote: > > Who is generating this ringback? The ATA or asterisk?
> Find out by doing a trace. If you're using callprogress, then you should > see a 180 Ringing sent to the ATA when we detect ringing on the FXO. If > you're not using call progress, then we should not be sending 180 ringing. We're not using callprogress (at least it's not set in zapata.conf). I also tried explicitly setting it to 'no', reloading, and trying again. No change. I do get a 180 Ringing. I am dialing from my ATA-186 to 18189950699, a telco busy test. Yet all I hear is a ring, though on one call I heard a short blip of busy before the ring started. [See attachment for the trace output] > You can also use the new "debug channel Zap/1-1" to see if the FXO is > ringing. If I do a 'show channel' on it, I get: Zap/21-1 (intrunk 6197474525 1 ) Ringing AppDial (Outgoing Line) SIP/0054-b2bc (outtrunk 8189950699 2 ) Ring Dial Tor/g1/BYEXTENSION That 6197474525 is strange. That's probably the DNIS used on the last incoming call to that channel, but has nothing to do with my outgoing call. I was calling from 6193640054 (SIP 0054). My definition in sip.conf is: [0054] type=friend insecure=yes secret=myownsecret callerid="Jim Gottlieb <(619) 364-0054>" ; dynamic binding seems to time-out; try defaultip host=dynamic defaultip=192.168.40.90 ; need to set the following so we can use voicemail and other DTMF apps dtmfmode=rfc2833
[I dialed...] Sip read: > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone> Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> User-Agent: Cisco ATA v2.15 ata18x (020927a) Expires: 300 Content-Length: 247 Content-Type: application/sdp v=0 o=0054 5680 5680 IN IP4 192.168.40.90 s=ATA186 Call c=IN IP4 192.168.40.90 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 11 lines Interface is eth0 IP Address is 198.51.175.9 Using latest request as basis request Sending to 192.168.40.90 : 5060 (non-NAT) Capabilities: us - 14, them - 13, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone>;tag=06e4b177 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: <sip:[EMAIL PROTECTED];user=phone> Proxy-Authenticate: Digest realm="asterisk", nonce="028f4554" Content-Length: 0 to 192.168.40.90:5060 Sip read: > ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone>;tag=06e4b177 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 8 headers, 0 lines Sip read: > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone> Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="0054",realm="asterisk",nonce="028f4554",ur i="sip:[EMAIL PROTECTED]",response="a8dbe8f8d6faee139756514c82cad48f" Expires: 300 Content-Length: 247 Content-Type: application/sdp v=0 o=0054 5686 5686 IN IP4 192.168.40.90 s=ATA186 Call c=IN IP4 192.168.40.90 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.40.90 : 5060 (non-NAT) Capabilities: us - 14, them - 13, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 18189950699 in sip Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone>;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: <sip:[EMAIL PROTECTED];user=phone> Content-Length: 0 to 192.168.40.90:5060 We're at 198.51.175.9 port 53614 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone>;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: <sip:[EMAIL PROTECTED];user=phone> Content-Type: application/sdp Content-Length: 211 v=0 o=root 5860 5860 IN IP4 198.51.175.9 s=session c=IN IP4 198.51.175.9 t=0 0 m=audio 53614 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 to 192.168.40.90:5060 Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone>;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: <sip:[EMAIL PROTECTED];user=phone> Content-Length: 0 to 192.168.40.90:5060 [At this pont I was hearing ringback tone. Then I hung up...] Sip read: > CANCEL sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone>;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 CANCEL User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 8 headers, 0 lines Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.40.90:5060 From: <sip:[EMAIL PROTECTED];user=phone>;tag=850095511 To: <sip:[EMAIL PROTECTED];user=phone>;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 CANCEL User-Agent: Asterisk PBX Contact: <sip:[EMAIL PROTECTED];user=phone> Content-Length: 0 to 192.168.40.90:5060
