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jsl


Registration Date: 01.01.1970
Posts:

VOIP.ms Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

I signed up for a new DID with Voip.ms I've set it up the same way as my existing DID (callcentric.com) but my voip.ms calls are not being completed. I've attached a sip trace. I can not tell if there are any clues in the trace. If anyone can point me in the right direction I'd appreciate it.

Jun 2 07:53:16 VERBOSE[95902] logger.c:
<-- SIP read
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.231.245.210:30557;branch=z9hG4bK07c8f8e5;received=67.231.245.210;rport=30557
From: <sip:9122890417@67.231.245.210:30557>;tag=as55e35d83
To: "9122306743" <sip:9122306743@174.34.146.162>;tag=as0f3c16f0
Call-ID: 009a5f970567626c70ca2bc97079b8e4@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 2 07:53:16 VERBOSE[95902] logger.c: --- (10 headers 0 lines)Jun 2 07:53:16 VERBOSE[95902] logger.c: --- (10 headers 0 lines)---
Jun 2 07:53:16 VERBOSE[95902] logger.c: Destroying call '009a5f970567626c70ca2bc97079b8e4@174.34.146.162'
02.06.2011 14:59 jslittlefield is offline Search for Posts by jslittlefield Add jslittlefield to your Buddy List
Dia
Premium Account


Registration Date: 03.03.2006
Posts: 1443
RE: VOIP.ms Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

I know for a fact that VoIP.ms trunks work fine with PBXes, so in order to identify the issue, please provide us with more of the trace, especially the message before the one displayed above.
02.06.2011 22:20 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
arn


Registration Date: 01.01.1970
Posts:

RE: VOIP.ms Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Yes, I can confirm that VoIP.ms works excellent with PBXes.

I have never had a problem with VoIP.ms on PBXes, it just works.
02.06.2011 23:42 arnebolen is offline Search for Posts by arnebolen Add arnebolen to your Buddy List
jsl


Registration Date: 01.01.1970
Posts:

RE: VOIP.ms Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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
03.06.2011 14:22 jslittlefield is offline Search for Posts by jslittlefield Add jslittlefield to your Buddy List
 
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