I'm glad to know that VOIP.ms is capable of working well. They have some nice features and great pricing. I've created a new sip trace it is below.
Jun 3 07:03:38 VERBOSE[103301] logger.c:
<-- SIP read
INVITE sip:19122890417@pbxes.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.212:5062;branch=z9hG4bK-daa24472;rport
From: John Littlefield <sip:jslittlefield-12@pbxes.com>;tag=32faecfab3884b32o0
P-src-ip: 68.51.129.191
To: <sip:19122890417@pbxes.com>
Remote-Party-ID: John Littlefield <sip:jslittlefield-12@pbxes.com>;screen=yes;party=calling
Call-ID: 628d469a-ec30be12@192.168.0.212
CSeq: 101 INVITE
Max-Forwards: 70
Contact: John Littlefield <sip:jslittlefield-12@192.168.0.212:5062>
Expires: 240
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 259
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 17740133 17740133 IN IP4 192.168.0.212
s=-
c=IN IP4 192.168.0.212
t=0 0
m=audio 16478 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/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
Jun 3 07:03:38 VERBOSE[103301] logger.c: --- (16 headers 13 lines)Jun 3 07:03:38 VERBOSE[103301] logger.c: --- (16 headers 13 lines)---
Jun 3 07:03:38 VERBOSE[103301] logger.c: Using INVITE request as basis request - 628d469a-ec30be12@192.168.0.212
Jun 3 07:03:38 VERBOSE[103301] logger.c: Reliably Transmitting (NAT)
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.212:5062;branch=z9hG4bK-daa24472;received=67.231.245.210;rport=63474
From: John Littlefield <sip:jslittlefield-12@pbxes.com>;tag=32faecfab3884b32o0
To: <sip:19122890417@pbxes.com>;tag=as431f72d1
Call-ID: 628d469a-ec30be12@192.168.0.212
CSeq: 101 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:19122890417@67.231.245.210:30557>
Proxy-Authenticate: Digest realm="pbxes.org", nonce="0ef70cd25302d41449b1296311428c4218c69363"
Content-Length: 0
---
Jun 3 07:03:38 VERBOSE[103301] logger.c: Scheduling destruction of call '628d469a-ec30be12@192.168.0.212' in 15000 ms
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found user 'jslittlefield-12'
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found peer 'jslittlefield-12'
Jun 3 07:03:38 VERBOSE[103301] logger.c:
<-- SIP read
ACK sip:19122890417@pbxes.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.212:5062;branch=z9hG4bK-daa24472;rport
From: John Littlefield <sip:jslittlefield-12@pbxes.com>;tag=32faecfab3884b32o0
P-src-ip: 68.51.129.191
To: <sip:19122890417@pbxes.com>;tag=as431f72d1
Call-ID: 628d469a-ec30be12@192.168.0.212
CSeq: 101 ACK
Max-Forwards: 70
Contact: John Littlefield <sip:jslittlefield-12@192.168.0.212:5062>
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 0
Jun 3 07:03:38 VERBOSE[103301] logger.c: --- (11 headers 0 lines)Jun 3 07:03:38 VERBOSE[103301] logger.c: --- (11 headers 0 lines)---
Jun 3 07:03:38 VERBOSE[103301] logger.c:
<-- SIP read
INVITE sip:19122890417@pbxes.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.212:5062;branch=z9hG4bK-c73aa77e;rport
From: John Littlefield <sip:jslittlefield-12@pbxes.com>;tag=32faecfab3884b32o0
P-src-ip: 68.51.129.191
To: <sip:19122890417@pbxes.com>
Remote-Party-ID: John Littlefield <sip:jslittlefield-12@pbxes.com>;screen=yes;party=calling
Call-ID: 628d469a-ec30be12@192.168.0.212
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="jslittlefield-12",realm="pbxes.org",nonce="0ef70cd25302d41449b1296311428c4218c69363",uri="sip:19122890417@pbxes.com",algorithm=MD5,response="ee3017a19d5e79fbb6ef44c293df185a"
Contact: John Littlefield <sip:jslittlefield-12@192.168.0.212:5062>
Expires: 240
User-Agent: Linksys/SPA2100-3.3.6
Content-Length: 259
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 17740133 17740133 IN IP4 192.168.0.212
s=-
c=IN IP4 192.168.0.212
t=0 0
m=audio 16478 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/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
Jun 3 07:03:38 VERBOSE[103301] logger.c: --- (17 headers 13 lines)Jun 3 07:03:38 VERBOSE[103301] logger.c: --- (17 headers 13 lines)---
Jun 3 07:03:38 VERBOSE[103301] logger.c: Using INVITE request as basis request - 628d469a-ec30be12@192.168.0.212
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found user 'jslittlefield-12'
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found peer 'jslittlefield-12'
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found RTP audio format 0
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found RTP audio format 100
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found RTP audio format 101
Jun 3 07:03:38 VERBOSE[103301] logger.c: Peer audio RTP is at port 192.168.0.212:16478
Jun 3 07:03:38 VERBOSE[103301] logger.c: Peer video RTP is at port 192.168.0.212:65535
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found description format PCMU
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found description format NSE
Jun 3 07:03:38 VERBOSE[103301] logger.c: Found description format telephone-event
Jun 3 07:03:38 VERBOSE[103301] logger.c: Capabilities: us - 0x18061e (gsm|ulaw|alaw|g726|speex|ilbc|h263|h263p), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Jun 3 07:03:38 VERBOSE[103301] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Jun 3 07:03:38 VERBOSE[103301] logger.c: Looking for 19122890417 in from-internal (domain pbxes.com)
Jun 3 07:03:38 VERBOSE[103301] logger.c: list_route: hop: <sip:jslittlefield-12@192.168.0.212:5062>
Jun 3 07:03:38 VERBOSE[103301] logger.c: Transmitting (NAT)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.212:5062;branch=z9hG4bK-c73aa77e;received=67.231.245.210;rport=63474
From: John Littlefield <sip:jslittlefield-12@pbxes.com>;tag=32faecfab3884b32o0
To: <sip:19122890417@pbxes.com>
Call-ID: 628d469a-ec30be12@192.168.0.212
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:19122890417@67.231.245.210:30557>
Content-Length: 0
Jun 3 07:03:39 VERBOSE[14249] logger.c: -- Called southernpicker@gmail.com/+19122890417@voice.google.com
Jun 3 07:03:40 VERBOSE[14249] logger.c: -- Gtalk/+19122890417@voice.google.com-ae63 is ringing
Jun 3 07:03:40 VERBOSE[14249] logger.c: Transmitting (NAT)
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.212:5062;branch=z9hG4bK-c73aa77e;received=67.231.245.210;rport=63474
From: John Littlefield <sip:jslittlefield-12@pbxes.com>;tag=32faecfab3884b32o0
To: <sip:19122890417@pbxes.com>;tag=as4f0e73c6
Call-ID: 628d469a-ec30be12@192.168.0.212
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:19122890417@67.231.245.210:30557>
Content-Length: 0
---
Jun 3 07:03:41 VERBOSE[103301] logger.c:
<-- SIP read
INVITE sip:9122890417@67.231.245.210:30557 SIP/2.0
Via: SIP/2.0/UDP 174.34.146.162:5060;branch=z9hG4bK40f4ecc3;rport
From: "+19122891015" <sip:9122891015@174.34.146.162>;tag=as613527b7
To: <sip:9122890417@67.231.245.210:30557>
Contact: <sip:9122891015@174.34.146.162>
Call-ID: 10ed043968edc9c162278fe76a0e852e@174.34.146.162
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+19122891015" <sip:9122891015@174.34.146.162>;privacy=off;screen=no
Date: Fri, 03 Jun 2011 12:03:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 9433 9433 IN IP4 174.34.146.162
s=session
c=IN IP4 174.34.146.162
t=0 0
m=audio 12556 RTP/AVP 0 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Jun 3 07:03:41 VERBOSE[103301] logger.c: --- (15 headers 15 lines)Jun 3 07:03:41 VERBOSE[103301] logger.c: --- (15 headers 15 lines)---
Jun 3 07:03:41 VERBOSE[103301] logger.c: Using INVITE request as basis request - 10ed043968edc9c162278fe76a0e852e@174.34.146.162
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found peer 'VOIPms'
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found RTP audio format 0
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found RTP audio format 18
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found RTP audio format 3
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found RTP audio format 101
Jun 3 07:03:41 VERBOSE[103301] logger.c: Peer audio RTP is at port 174.34.146.162:12556
Jun 3 07:03:41 VERBOSE[103301] logger.c: Peer video RTP is at port 174.34.146.162:65535
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found description format PCMU
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found description format G729
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found description format GSM
Jun 3 07:03:41 VERBOSE[103301] logger.c: Found description format telephone-event
Jun 3 07:03:41 VERBOSE[103301] logger.c: Capabilities: us - 0x61e (gsm|ulaw|alaw|g726|speex|ilbc), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Jun 3 07:03:41 VERBOSE[103301] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Jun 3 07:03:41 VERBOSE[103301] logger.c: Looking for 9122890417 in from-pstn (domain 67.231.245.210)
Jun 3 07:03:41 VERBOSE[103301] logger.c: list_route: hop: <sip:9122891015@174.34.146.162>
Jun 3 07:03:41 VERBOSE[103301] logger.c: Transmitting (NAT)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 174.34.146.162:5060;branch=z9hG4bK40f4ecc3;received=174.34.146.162;rport=5060
From: "+19122891015" <sip:9122891015@174.34.146.162>;tag=as613527b7
To: <sip:9122890417@67.231.245.210:30557>
Call-ID: 10ed043968edc9c162278fe76a0e852e@174.34.146.162
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9122890417@67.231.245.210:30557>
Content-Length: 0
---
Jun 3 07:03:41 VERBOSE[14294] logger.c: We're at 67.231.245.210 port 45308
Jun 3 07:03:41 VERBOSE[14294] logger.c: Video is at 67.231.245.210 port 41552
Jun 3 07:03:41 VERBOSE[14294] logger.c: Adding codec 0x4 (ulaw) to SDP
Jun 3 07:03:41 VERBOSE[14294] logger.c: Adding codec 0x2 (gsm) to SDP
Jun 3 07:03:41 VERBOSE[14294] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jun 3 07:03:41 VERBOSE[14294] logger.c: Reliably Transmitting (NAT)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.34.146.162:5060;branch=z9hG4bK40f4ecc3;received=174.34.146.162;rport=5060
From: "+19122891015" <sip:9122891015@174.34.146.162>;tag=as613527b7
To: <sip:9122890417@67.231.245.210:30557>;tag=as712505db
Call-ID: 10ed043968edc9c162278fe76a0e852e@174.34.146.162
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9122890417@67.231.245.210:30557>
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 103289 103289 IN IP4 67.231.245.210
s=session
c=IN IP4 67.231.245.210
t=0 0
m=audio 45308 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
---
Jun 3 07:03:41 VERBOSE[103301] logger.c:
<-- SIP read
ACK sip:9122890417@67.231.245.210:30557 SIP/2.0
Via: SIP/2.0/UDP 174.34.146.162:5060;branch=z9hG4bK0b6b310f;rport
From: "+19122891015" <sip:9122891015@174.34.146.162>;tag=as613527b7
To: <sip:9122890417@67.231.245.210:30557>;tag=as712505db
Contact: <sip:9122891015@174.34.146.162>
Call-ID: 10ed043968edc9c162278fe76a0e852e@174.34.146.162
CSeq: 102 ACK
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+19122891015" <sip:9122891015@174.34.146.162>;privacy=off;screen=no
Content-Length: 0
Jun 3 07:03:41 VERBOSE[103301] logger.c: --- (11 headers 0 lines)Jun 3 07:03:41 VERBOSE[103301] logger.c: --- (11 headers 0 lines)---
Jun 3 07:03:42 VERBOSE[14294] chan_sip.c: Hangup call SIP/121057-c3f5, SIP callid 10ed043968edc9c162278fe76a0e852e@174.34.146.162
Jun 3 07:03:42 VERBOSE[14294] logger.c: set_destination: Parsing <sip:9122891015@174.34.146.162> for address/port to send to
Jun 3 07:03:42 VERBOSE[14294] logger.c: set_destination: set destination to 174.34.146.162, port 5060
Jun 3 07:03:42 VERBOSE[14294] logger.c: Reliably Transmitting (NAT)
BYE sip:9122891015@174.34.146.162 SIP/2.0
Via: SIP/2.0/UDP 67.231.245.210:30557;branch=z9hG4bK3886559f;rport
From: <sip:9122890417@67.231.245.210:30557>;tag=as712505db
To: "+19122891015" <sip:9122891015@174.34.146.162>;tag=as613527b7
Contact: <sip:9122890417@67.231.245.210:30557>
Call-ID: 10ed043968edc9c162278fe76a0e852e@174.34.146.162
CSeq: 102 BYE
User-Agent: PBX
Max-Forwards: 70
Content-Length: 0
---
Jun 3 07:03:42 VERBOSE[103301] logger.c:
<-- SIP read
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.231.245.210:30557;branch=z9hG4bK3886559f;received=67.231.245.210;rport=30557
From: <sip:9122890417@67.231.245.210:30557>;tag=as712505db
To: "+19122891015" <sip:9122891015@174.34.146.162>;tag=as613527b7
Call-ID: 10ed043968edc9c162278fe76a0e852e@174.34.146.162
CSeq: 102 BYE
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
Jun 3 07:03:42 VERBOSE[103301] logger.c: --- (10 headers 0 lines)Jun 3 07:03:42 VERBOSE[103301] logger.c: --- (10 headers 0 lines)---
Jun 3 07:03:42 VERBOSE[103301] logger.c: Destroying call '10ed043968edc9c162278fe76a0e852e@174.34.146.162'
Jun 3 07:03:43 VERBOSE[14249] logger.c: -- Gtalk/+19122890417@voice.google.com-ae63 answered SIP/jslittlefield-12-34b0
Jun 3 07:03:43 VERBOSE[14249] logger.c: We're at 67.231.245.210 port 42222
Jun 3 07:03:43 VERBOSE[14249] logger.c: Video is at 67.231.245.210 port 38722
Jun 3 07:03:43 VERBOSE[14249] logger.c: Adding codec 0x4 (ulaw) to SDP
Jun 3 07:03:43 VERBOSE[14249] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jun 3 07:03:43 VERBOSE[14249] logger.c: Reliably Transmitting (NAT)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.212:5062;branch=z9hG4bK-c73aa77e;received=67.231.245.210;rport=63474
From: John Littlefield <sip:jslittlefield-12@pbxes.com>;tag=32faecfab3884b32o0
To: <sip:19122890417@pbxes.com>;tag=as4f0e73c6
Call-ID: 628d469a-ec30be12@192.168.0.212
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:19122890417@67.231.245.210:30557>
Content-Type: application/sdp
Content-Length: 195
v=0
o=root 103289 103289 IN IP4 67.231.245.210
s=session
c=IN IP4 67.231.245.210
t=0 0
m=audio 42222 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
---
Jun 3 07:03:43 VERBOSE[14249] chan_sip.c: Hangup call SIP/jslittlefield-12-34b0, SIP callid 628d469a-ec30be12@192.168.0.212
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