Project

General

Profile

Actions

Defect #1837

closed

OK for Past Year - Incoming Calls Now Failing

Added by Joe Leighton almost 12 years ago. Updated over 11 years ago.

Status:
Closed
Priority:
High
Assignee:
-
Category:
Quality of Service
Target version:
Start date:
03/13/2013
Due date:
% Done:

0%

Estimated time:
Keywords:

Description

System has been working well for over a year. Now unable to receive phone calls for past 2 days. Below is a call log for 1 such failure.

My SIP account credentials are:

SIP address:
Password: a5c8rr6673

- Username: 2233455417
- Domain/Realm: sip2sip.info
- Outbound proxy: proxy.sipthor.net
- XCAP root: https://xcap.sipthor.net/xcap-root

Please advise resolution.

Thanks, Joe

CDRTool SIP trace SIP session

Click on each packet to expand its body content

Click here for RTP media information

Packet     Size     Time     66.54.140.46    node11    node05
1/10 884 bytes 21:16:26
UDP port 5060 INVITE sip:
audio->66.54.140.46:18080
UDP port 5060
2/10 349 bytes 21:16:26
UDP port 5060 100 Trying for INVITE
UDP port 5060
3/10 1151 bytes 21:16:26 INVITE sip:
Diversion: counter=1
audio->81.23.228.129:59242
UDP port 5060
UDP port 5060
4/10 644 bytes 21:16:26 100 Trying for INVITE
Contact: 85.17.186.5:5060
UDP port 5060
UDP port 5060
5/10 660 bytes 21:16:26 180 Ringing for INVITE
Contact: 85.17.186.5:5060
UDP port 5060
UDP port 5060
6/10 572 bytes 21:16:26
UDP port 5060 180 Ringing for INVITE
Contact: 85.17.186.5:5060
UDP port 5060
+2s 7/10 1005 bytes 21:16:28 200 OK for INVITE
Contact: 85.17.186.5:5060
audio->85.17.186.5:6376
UDP port 5060
UDP port 5060
8/10 920 bytes 21:16:28
UDP port 5060 200 OK for INVITE
Contact: 85.17.186.5:5060
audio->81.23.228.129:59240
UDP port 5060
9/10 460 bytes 21:16:28
UDP port 5060 ACK sip::5060
UDP port 5060
10/10 488 bytes 21:16:28 ACK sip::5060
UDP port 5060
UDP port 5060

Files

sip2sip call log_failed incoming call.png (238 KB) sip2sip call log_failed incoming call.png call log of failed incoming call Joe Leighton, 03/13/2013 04:29 AM
sip2sip support_image of Xlite setup.png (46.4 KB) sip2sip support_image of Xlite setup.png Xlite setup Joe Leighton, 03/13/2013 06:07 PM
Actions #1

Updated by Adrian Georgescu almost 12 years ago

  • Description updated (diff)
  • Status changed from New to In progress

Your SIP device is not registered and the call went to voicemail.

There is nothing wrong about this server-wise. You should make your end-point register if you want to receive calls.

Actions #2

Updated by Joe Leighton almost 12 years ago

My SIP device is a X-Lite softphone downloaded on my computer.

Please advise how to "make my end-point register".

Please also note that I have been receiving calls successfully for 12 months up until 3 days ago and I have done nothing to change the setup.

Thanks, Joe

Actions #3

Updated by Adrian Georgescu almost 12 years ago

Three days ago we enabled TLS transport which is now advertised in DNS as default. Perhaps your client does not work well with TLS, try change the outbound proxy settings to use UDP or TCP instead.

Actions #4

Updated by Joe Leighton almost 12 years ago

Attached is a screenshot of my Xlite setup.

As you can see,I can change to send outbound via proxy.

If that makes sense, please advise proxy address.

Thanks.

Actions #5

Updated by Adrian Georgescu almost 12 years ago

  • Status changed from In progress to To be closed

Our service end-points are described here:

http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDeviceConfiguration

How you configure X-lite with them is a question for their manufacturer forum.

Actions #6

Updated by Tijmen de Mes over 11 years ago

  • Status changed from To be closed to Closed
Actions

Also available in: Atom PDF