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Unread 09-28-2007, 04:57 PM   #1
wishfull
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Default PSTN disconnect after call is answered

It was all working for last 2 years until about a week ago. Now when one call me using les.net pstn gateway call disconnects as soon as the call is answered. SIP to SIP works fine. Calls from some of other gateways I've tested work fine.
I don't know what to do anymore I have checked and double checked all the settings, hooked up the ATA (Sipura 3000) in a DMZ, even tried hooking it directly to the internet with any firewalls still no luck. Sent an email to les.net and sipphone.com no response either one. I have tried 2 other ATAs and same results. Please help I'm at a loss.
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Unread 09-28-2007, 05:58 PM   #2
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Which PSTN number?

It could very well be a problem with that PSTN, if it works from everywhere else...
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Unread 09-28-2007, 06:38 PM   #3
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Quote:
Originally Posted by emoci View Post
Which PSTN number?

It could very well be a problem with that PSTN, if it works from everywhere else...
NY 646-291-2168 (Les.net) doesn't work but 347-427-9019 1+# does work. Phoenix, AZ 602-427-5727 (Callcentric) is iffy sometimes it works not others.
I got a reply from les.net and they said it's SIP Broker PSTN gateway issue, I should be asking for help here. Is there someone who could look into this? I have failed log file, if that will help.

Last edited by wishfull; 09-28-2007 at 06:59 PM.
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Unread 09-28-2007, 07:01 PM   #4
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Well, I don't know if this helps, but I tried both numbers (646 numbers), and they do work for me....

Some things to look into:

-Before trying any of these other things, if you can, try some other Access Numbers and see if the problem persists with them too.

-The Symmetric NAT setting on your VoXalot account is it set to YES or NO
(if it's YES, leave it alone, if it is NO, try setting to yes)

-One problem that has been coming lately, is the support for ReInvites (the things is most Linksys Devices I've seen have it enabled by default)

-Also if you don't already, have STUN set up

-Also check your router, if it has UPnP enable it (it maybe that it's closing the port after the connection is established dropping the line) (If you have UPnP, I would turn off DMZ)
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Unread 09-28-2007, 07:59 PM   #5
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ReInvites is set to 30. I have no idea where to turn it off. Also remember I did try hooking up the ATA outside the firewall and same results. I also have tired other SIPPHONE.COM numbers again same results. I'm starting to wonder if the problem is with SIP Broker and sipphone.com (1747) numbers. UPnP is enabled and DMZ is off.

Last edited by wishfull; 09-28-2007 at 08:07 PM.
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Unread 09-28-2007, 08:10 PM   #6
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Ok..., so you have an ATA registered with Gizmo then, I assume!!!!

Try getting a free VoXBasic account and registering that in your ATA, and see if you have the same issue ? (You may be able to isolate if it's a Gizmo/SipPhone issue or not)

Your ReInvite setting is the same as mine...(so that should be fine)

BTW, you tried just powercycling the ATA right (turn off, wait a bit, plug back in)

Last edited by emoci; 09-28-2007 at 08:17 PM.
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Unread 09-29-2007, 03:26 AM   #7
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When I got home my wife said it was working again. So I don't know what was the cause of the problem. I'm happy for now. Thanks for your help.
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Unread 09-30-2007, 11:10 PM   #8
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Quote:
Originally Posted by emoci View Post
Ok..., so you have an ATA registered with Gizmo then, I assume!!!!

Try getting a free VoXBasic account and registering that in your ATA, and see if you have the same issue ? (You may be able to isolate if it's a Gizmo/SipPhone issue or not)

Your ReInvite setting is the same as mine...(so that should be fine)

BTW, you tried just powercycling the ATA right (turn off, wait a bit, plug back in)
This freakin thing stopped working again. Yes it's a Gizmo number. emoci I took your advice and signed up with VoXbasic account well with this account gateway does not hangup but there is no audio. So there is some sort of problem because both accounts work SIP to SIP without any issues.
Cycled power on my ATA many times still no go.
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Unread 09-30-2007, 11:59 PM   #9
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Quote:
Originally Posted by wishfull View Post
This freakin thing stopped working again. Yes it's a Gizmo number. emoci I took your advice and signed up with VoXbasic account well with this account gateway does not hangup but there is no audio. So there is some sort of problem because both accounts work SIP to SIP without any issues.
Cycled power on my ATA many times still no go.
You have STUN set up right..... this sounds a whole lot like a NAT problem....

Also how are you testing: You are now calling Access Number then *010123456 (where 123456 is your VoXalot Number) right?

I understand you are having a problem, and it's annoying when it seems to be working then not working from one moment to the next but short of trying to guide you through steps to troubleshoot, there isn't much more I can do. Re-Tested both those SipBroker Numbers with my account (using a PAP2 behind a WRT54G router) and calls seem fine....

Please make sure that you are calling the SipBroker Access number from a regular phone, not another VoIP line (Vonage included), some odd occurrences have been reported when SipBroker numbers are dialed from VoIP lines....

Quote:
Here are a few things that will help:

-Follow this Guide in terms of setting up your ATA http://forum.voxalot.com/voxalot-gen....html#post5813

-Preferably make sure you have STUN set up (stun.xten.org has been working for me)
-If you are behing a router that has UPnP, make sure it's active
-This is optional, but it may help to open these port ranges and forward them to your ATA's IP:

5050-5064
5000-5005
16300-16500 (this maybe slightly different for your ATA, there should be an RTP port range setting on your ATA, if it is different, note the range and open that range instead in your router...)

For some background on this see: http://forum.voxalot.com/12685-post20.html

-If your router has DMZ, try at first disabling it completely. However if you get through this suggestions and still aren't having any luck this worked for someone:
Quote:
Originally Posted by nacho View Post
Thanks a lot!

The problem was that the router was blocking.
I configured the sipura with static ip and specified that ip on the DMZ of the router.

-If you can check the router log, see if it is blocking any traffic


Once you are done setting up your ATA, STUN, DIAL PLAN (copy the one you have right now if it has worked in the past), UPnP and/or DMZ and/or Port Range forwarding in your router try the following:

-Try Calling *600, do you reach the Echo Test?
-Call one of the SIPBroker Access Numbers SIPBroker - PSTN Numbers (do this from a regular phone, not your ATA) and then enter *010123456 (replace 123456 with your VoXalot number). Do you receive the call?
-Enter your VoXalot SIP URI (123456@voxalot.com) here and see if the ECHO Test call comes in: SIPBroker - EziDial
-Make sure there are no Internet Connection problems. Run a SpeedTest, or maybe VoIP test at TestYourVoIP.com

Last edited by emoci; 10-01-2007 at 12:03 AM.
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Unread 10-01-2007, 04:40 PM   #10
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I find it hard to believe that it would be a configuration problem. Like I said before it have even tried hooking it up outside the firewall with same results. New Yorks IM number 347-427-9019 1+sip number +# never quit working. I think it's some sort of SIP Broker issue. Here is the part of the log when the call gets answered.

<sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 102 INVITE<013><010>Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK742e.66d48c77.0<013>< 010>Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bK742e.0354e857.0<013><0 10>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK53de55e7<013><010 >Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>S erver: Sipura/SPA3000-2.0.13(GWg)<013><010>Content-Length: 0
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local3.Debug 192.168.0.100 RSE_DEBUG: reference domainroxy01.sipphone.com
Local3.Debug 192.168.0.100 RSE_DEBUG: unref domain, proxy01.sipphone.com
Local3.Debug 192.168.0.100 RSE_DEBUG: last unref for domain proxy01.sipphone.com
Local3.Debug 192.168.0.100 [0]Off Hook
Local0.Info 192.168.0.100 [0:5060]->198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]->198.65.166.131:5060
Local7.Debug 192.168.0.100 SIP/2.0 200 OK<013><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 102 INVITE<013><010>Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK742e.66d48c77.0<013>< 010>Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bK742e.0354e857.0<013><0 10>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK53de55e7<013><010 >Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>C ontact: Paul Kainth <sip:1747668xxxx@192.168.0.100:5060><013><010>Serv er: Sipura/SPA3000-2.0.13(GWg)<013><010>Content-Length: 237<013><010>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<013><010>Supported: x-sipura<013><010>Content-Type: application/sdp<013><010><013><010>v=0<013><010>o=- 1764021 1764021 IN IP4 192.168.0.100<013><010>s=-<013><010>c=IN IP4 192.168.0.100<013><010>t=0 0<013><010>m=audio 16388 RTP/AVP 0 100 101<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:100 NSE/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-15<013><010>a=ptime:30<013><010>a=sendrecv
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 [0:5060]->198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]->198.65.166.131:5060
Local7.Debug 192.168.0.100 SIP/2.0 200 OK<013><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 102 INVITE<013><010>Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK742e.66d48c77.0<013>< 010>Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bK742e.0354e857.0<013><0 10>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK53de55e7<013><010 >Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>C ontact: Paul Kainth <sip:1747668xxxx@192.168.0.100:5060><013><010>Serv er: Sipura/SPA3000-2.0.13(GWg)<013><010>Content-Length: 237<013><010>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<013><010>Supported: x-sipura<013><010>Content-Type: application/sdp<013><010><013><010>v=0<013><010>o=- 1764021 1764021 IN IP4 192.168.0.100<013><010>s=-<013><010>c=IN IP4 192.168.0.100<013><010>t=0 0<013><010>m=audio 16388 RTP/AVP 0 100 101<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:100 NSE/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-15<013><010>a=ptime:30<013><010>a=sendrecv
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local7.Debug 192.168.0.100 INVITE sip:1747668xxxx@207.81.207.187:5060 SIP/2.0<013><010>Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>V ia: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK842e.b8098a01.0<013>< 010>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK3446360f<013><010 >From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>Contact: <sip:1250992xxxx@204.11.194.10:5070;nat=yes><013>< 010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 103 INVITE<013><010>User-Agent: Asterisk PBX<013><010>Max-Forwards: 16<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<013><010>Content-Type: application/sdp<013><010>Content-Length: 237<013><010>RemoteIP: 204.11.194.10<013><010>P-hint: rr-enforced<013><010>P-NATed-URI: YES (1)<013><010>P-RTP-Proxy: YES (1)<013><010><013><010>v=0<013><010>o=root 27143 27144 IN IP4 64.34.176.212<013><010>s=session<013><010>c=IN IP4 198.65.166.131<013><010>t=0 0<013><010>m=audio 46998 RTP/AVP 0 101<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=silenceSuppff - - - -<013><010>a=nortpproxy:yes
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local2.Debug 192.168.0.100 SIPDLG:Bad ReInvite
Local0.Info 192.168.0.100 [0:5060]->198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]->198.65.166.131:5060
Local7.Debug 192.168.0.100 SIP/2.0 400 Bad Request<013><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 103 INVITE<013><010>Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK842e.b8098a01.0<013>< 010>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK3446360f<013><010 >Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>S erver: Sipura/SPA3000-2.0.13(GWg)<013><010>Content-Length: 0
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local7.Debug 192.168.0.100 ACK sip:1747668xxxx@207.81.207.187:5060 SIP/2.0<013><010>Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>V ia: SIP/2.0/UDP 198.65.166.131;branch=0<013><010>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK563f006b<013><010 >From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>Contact: <sip:1250992xxxx@204.11.194.10:5070;nat=yes><013>< 010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 102 ACK<013><010>User-Agent: Asterisk PBX<013><010>Max-Forwards: 16<013><010>Content-Length: 0<013><010>P-hint: rr-enforced<013><010>P-NATed-URI: YES (1)<013><010>P-RTP-Proxy: YES (1)
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local2.Debug 192.168.0.100 CC:Connected
Local2.Debug 192.168.0.100 [0:0]ENC INIT 0
Local2.Debug 192.168.0.100 [0:0]RTP Tx Up (pt=0->c641a683:46998)
Local2.Debug 192.168.0.100 [0:0]RTCP Tx Up
Local2.Debug 192.168.0.100 [0:0]RTP Rx 1st PKT @16388(2)
Local2.Debug 192.168.0.100 [0:0]DEC INIT 0
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local7.Debug 192.168.0.100 ACK sip:1747668xxxx@207.81.207.187:5060 SIP/2.0<013><010>Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>V ia: SIP/2.0/UDP 198.65.166.131;branch=0<013><010>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK1aa2e014<013><010 >From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>Contact: <sip:1250992xxxx@204.11.194.10:5070;nat=yes><013>< 010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 102 ACK<013><010>User-Agent: Asterisk PBX<013><010>Max-Forwards: 16<013><010>Content-Length: 0<013><010>P-hint: rr-enforced<013><010>P-NATed-URI: YES (1)<013><010>P-RTP-Proxy: YES (1)
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 [0:5060]->198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]->198.65.166.131:5060
Local7.Debug 192.168.0.100 SIP/2.0 400 Bad Request<013><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 103 INVITE<013><010>Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK842e.b8098a01.0<013>< 010>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK3446360f<013><010 >Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>S erver: Sipura/SPA3000-2.0.13(GWg)<013><010>Content-Length: 0
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local7.Debug 192.168.0.100 ACK sip:1747668xxxx@207.81.207.187:5060 SIP/2.0<013><010>Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK842e.b8098a01.0<013>< 010>From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>CSeq: 103 ACK<013><010>Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>C ontent-Length: 0
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local7.Debug 192.168.0.100 ACK sip:1747668xxxx@207.81.207.187:5060 SIP/2.0<013><010>Record-Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>V ia: SIP/2.0/UDP 198.65.166.131;branch=0<013><010>Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK1aa2e014<013><010 >From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>Contact: <sip:1250992xxxx@204.11.194.10:5070;nat=yes><013>< 010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>CSeq: 103 ACK<013><010>User-Agent: Asterisk PBX<013><010>Max-Forwards: 16<013><010>Content-Length: 0<013><010>P-hint: rr-enforced<013><010>P-NATed-URI: YES (1)<013><010>P-RTP-Proxy: YES (1)
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local0.Info 192.168.0.100 [0:5060]<<198.65.166.131:5060
Local7.Debug 192.168.0.100 ACK sip:1747668xxxx@207.81.207.187:5060 SIP/2.0<013><010>Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK842e.b8098a01.0<013>< 010>From: "1250992xxxx" <sip:1250992xxxx@204.11.194.10:5070>;tag=as655273e 9<013><010>Call-ID: 5bef248f7b5ae4c83acb7b204fe2f870@204.11.194.10<013 ><010>To: <sip:*0131747668xxxx@sipbroker.com>;tag=8926ed722a c7be6ai0<013><010>CSeq: 103 ACK<013><010>Route: <sip:198.65.166.131;ftag=as655273e9;lr><013><010>C ontent-Length: 0
Local0.Info 192.168.0.100 <010>
Local0.Info 192.168.0.100 <010>
Local3.Debug 192.168.0.100 RSE_DEBUG: reference domainroxy01.sipphone.com
Local2.Debug 192.168.0.100 Discard RSP w/o trnsac
Local3.Debug 192.168.0.100 RSE_DEBUG: unref domain, proxy01.sipphone.com
Local3.Debug 192.168.0.100 RSE_DEBUG: last unref for domain proxy01.sipphone.com
Local3.Debug 192.168.0.100 [0]On Hook
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