rendered paste body<--- SIP read from 64.158.177.99:5060 --->
INVITE sip:2482467007@200.xx.36.42 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;Refresher=uac
Supported: timer
To: <sip:2482467007@200.xx.36.42:5060>
From: <sip:5868735246@64.158.177.99>;tag=3416263034-861359
Remote-Party-Id: <sip:5868735246@64.158.177.99:5060>;privacy=off
Call-ID: 1747302-3416263034-861223@is02.broadvox.net
CSeq: 1 INVITE
Via: SIP/2.0/UDP 64.158.177.99:5060;branch=e9f4d0bea6c5385e4e4d8a41c3f5b20a
Contact: sip:5868735246@64.158.177.99:5060
Content-Type: application/sdp
Content-Length: 324
v=0
o=Version 1234 17934 IN IP4 64.158.162.74
s=sip call
c=IN IP4 64.158.162.71
t=0 0
m=audio 26350 RTP/AVP 0 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:30
<------------->
--- (13 headers 15 lines) ---
Sending to 64.158.177.99 : 5060 (no NAT)
Using INVITE request as basis request - 1747302-3416263034-861223@is02.broadvox.net
Found peer 'broadvox'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 64.158.162.71:26350
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Got unsupported a:fmtp in SDP offer
Found audio description format G723 for ID 4
Got unsupported a:fmtp in SDP offer
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 64.158.162.71:26350
Looking for 2482467007 in hosted_in (domain 200.xx.36.42)
list_route: hop: <sip:5868735246@64.158.177.99:5060>
sip1*CLI>
<--- Transmitting (no NAT) to 64.158.177.99:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.158.177.99:5060;branch=e9f4d0bea6c5385e4e4d8a41c3f5b20a;received=64.158.177.99
From: <sip:5868735246@64.158.177.99>;tag=3416263034-861359
To: <sip:2482467007@200.xx.36.42:5060>
Call-ID: 1747302-3416263034-861223@is02.broadvox.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2482467007@200.xx.36.42>
Content-Length: 0
<------------>
-- Executing [2482467007@hosted_in:1] Dial("SIP/broadvox-0086ce20", "SIP/02482957753") in new stack
[2008-04-03 22:00:43] WARNING[6472]: chan_sip.c:3008 sip_call: No audio format found to offer. Cancelling call to 02482957753
-- Couldn't call 02482957753