Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: <sip:[EMAIL PROTECTED]> Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 290
v=0 o=phone2 5972727 56415 IN IP4 192.168.1.1 s=SIP Call c=IN IP4 192.168.1.1 t=0 0 m=audio 10116 RTP/AVP 18 0 8 4 2 101 a=rtpmap:18 G729/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:4 g723/8000 a=rtpmap:2 g726/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
12 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.1 : 5060 (non-NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 101
Peer RTP is at port 192.168.1.1:0
Found description format G729
Found description format pcmu
Found description format pcma
Found description format g723
Found description format g726
Found description format telephone-event
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x11d(G723|ULAW|ALAW|G726|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found user 'phone1'
Looking for 90800500005 in sip
list_route: hop: <sip:[EMAIL PROTECTED]>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746
From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736
To: <sip:[EMAIL PROTECTED]>;tag=as71701551
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 192.168.1.1:5060 We're at 192.168.0.250 port 13586 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 To: <sip:[EMAIL PROTECTED]>;tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 265
v=0 o=root 24864 24864 IN IP4 192.168.0.250 s=session c=IN IP4 192.168.0.250 t=0 0 m=audio 13586 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -
to 192.168.1.1:5060 mars*CLI>
Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-YQM-30118 From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 To: <sip:[EMAIL PROTECTED]>;tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Content-Length: 0
9 headers, 0 lines mars*CLI>
Sip read: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 To: <sip:[EMAIL PROTECTED]>;tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Content-Length: 0
9 headers, 0 lines Sending to 192.168.1.1 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 To: <sip:[EMAIL PROTECTED]>;tag=as71701551 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0
to 192.168.1.1:5060 Destroying call '[EMAIL PROTECTED]' mars*CLI>
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