SIP call hangup

General Help
1

I am using Asterisk 1.6, FreePBX 2.5.1.1 and Centos 4.7 Final.

I have noticed that when I place a call to my mobile or my analogue line if I do not aswer and hang up the PBX handset the phone keeps ringing until I answer, it then goes dead like a ghost call. Does anybody have any ideas about this one? I have recently taken over from an experienced asterisk engineer and am quite new to this system. I would be happy to supply whatever information is needed to diagnose this problem.

Here is the end of the output from “asterisk cli” when I try to make a call to my mobile from the PBX. I don’t see what is wrong with this as I can see the hangup macro being used. But I do not really understand asterisk yet.

TEXT
– SIP/voip-unlimited-0997c410 is making progress passing it to SIP/200-b7a08778
– SIP/voip-unlimited-0997c410 is making progress passing it to SIP/200-b7a08778
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/200-b7a08778’ in macro ‘dialout-trunk’
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/200-b7a08778’
– Executing Macro(“SIP/200-b7a08778”, “hangupcall|”) in new stack
– Executing ResetCDR(“SIP/200-b7a08778”, “w”) in new stack
– Executing NoCDR(“SIP/200-b7a08778”, “”) in new stack
– Executing GotoIf(“SIP/200-b7a08778”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing GotoIf(“SIP/200-b7a08778”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing GotoIf(“SIP/200-b7a08778”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing Hangup(“SIP/200-b7a08778”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/200-b7a08778’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/200-b7a08778’
– Saved useragent “Grandstream GXP2010 1.1.5.15” for peer 201
TEXT

Thank you in advance.
Phil Garner

2

Phil,

I have recently setup and now testing a new asterisk/freepbx install and having the same hangup problem.

Just wanted to know how you went about troubleshooting the problem further and if you resolved the problem ?

Cheers

3

It transpired that I had changed the amount of rtp ports (/etc/asterisk/rtp.conf) but not adjusted the firewall accordingly. After making the changes I should have run a ‘reboot’ After I had done this I also changed the SIP setup to ‘nat=no’ in sip_nat.conf and gave the system a public IP address in order to register the SIP trunk. This appeared to clear this issue. I was then able to configure the nat setup with a different SIP trunk.

Make sure that you have the correct RTP ports open as 5060 only initiates the session these are usually set to 10000-20000 and also if you are using webmin by default it will use 10000 so rtp.conf will need to be changed to 10001-20000.

I hope this will point you in the right direction.

4

Phil Garner -

Correcting your statement:

Webmin uses HTTPS which is a TCP protocol RTP is UDP there is no conflict

5

Thats correct sorry. I was thinking of my Trixbox installation in that instance as it uses HTTP but you are right it is a TCP protocol. Thankyou for the correction.