PBXes » English » Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups » Very important port number considerations
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mem


Registration Date: 01.01.1970
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All subsequent calls being sent to voicemail. Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Ok, so I have a serious problem going on now.
It seems that whenever a call is picked up by a certain extension (5003) all subsequent calls are sent to voicemail.
Also, some type of "Freeze up" in the phone causes the all calls on that DID to go to voicemail and no outgoing calls allowed. Any clue what could cause that much damage?

21.06.2008 00:00 memo is offline Search for Posts by memo Add memo to your Buddy List
mem


Registration Date: 01.01.1970
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Here is a part of the log. I think this may be related to the
"Loop Detected" and the Everyone is busy/congested at this time (1:0/0/1)

Anyone have a clue? Thanks!

Jun 20 18:12:18 VERBOSE[32699] logger.c: -- Called memo-3
Jun 20 18:12:18 VERBOSE[16358] logger.c: -- Got SIP response 482 "Loop Detected"
Jun 20 18:12:18 VERBOSE[32699] logger.c: -- Now forwarding SIP/memo-5003-9290 to 'Local/@from-pstn' (thanks to SIP/memo-3-8937)
Jun 20 18:12:18 NOTICE[32699] chan_local.c: No such extension/context @from-pstn creating local channel
Jun 20 18:12:18 NOTICE[32699] app_dial.c: Unable to create local channel for call forward to 'Local/@from-pstn' (cause = 0)
Jun 20 18:12:18 VERBOSE[32699] chan_sip.c: Hangup call SIP/memo-3-8937, SIP callid 3aab3e6a45e8f28a7feba119357d551e@64.118.93.76
Jun 20 18:12:18 VERBOSE[32699] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
Jun 20 18:12:18 VERBOSE[32699] chan_sip.c: Hangup call SIP/memo-5003-9290, SIP callid cd826485-a9de68d3-d0ec7a00@192.168.0.101

21.06.2008 00:15 memo is offline Search for Posts by memo Add memo to your Buddy List
Dia
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Registration Date: 03.03.2006
Posts: 1443

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This is indeed a serious issue, which might take some time to investigate. What kind of SIP User Agents are you using for your Extensions?

21.06.2008 04:42 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
mem


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Hi. I am using Polycom Soundpoint IP500's .
So, I was going through settings and think I cleared up the loop issue. It seems to have left when I changed the "dial" settings on a few extensions. They were some changes I had made in an attempt to get the shared line working. (Never Worked)

But, I am still having the problem of the freeze up. So, I am getting the Phone brought in to test it here at home and possibly switch it out. I can't seem to find a difference in the setup at that location from here. I even went as far as putting the phone in a DMZ. Nothing....

Would love any ideas.

21.06.2008 05:37 memo is offline Search for Posts by memo Add memo to your Buddy List
mem


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Ok, so looking at the logs, it looks like I did clear up the loop issue. Thank you.

So, I still need to figure out the freezing.
Does anyone know if there would be issues concerning the ports seeing as they also have a "Vonage Box" on their network there?
Thanks.

21.06.2008 16:20 memo is offline Search for Posts by memo Add memo to your Buddy List
Dia
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Registration Date: 03.03.2006
Posts: 1443

Very important port number considerations Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Mixing and matching SIP User Agents on the same LAN requires very careful provisioning of the ports used for signaling (SIP) and media (RTP).

The signaling port needs to be unique for every SIP UA on the same LAN, otherwise all kinds of weird behavior is to be expected. The media port range should also be carefully selected, since some devices use a single port for RTP while others use a port range.

27.06.2008 13:37 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
 
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