07-26-2008 07:20 PM - edited 03-15-2019 12:13 PM
Hi All, Im having some problems trying to setup a CME site to use a SIP trunk. Outgoing calls work, but drop out after 30 seconds exactly. Incoming calls ring on the ip phone but when i go to answer it i comes up answered on the ip phone but on the mobile that initiated the call it still thinks the call is yet to be answered.
Any help would be much appreciated.
Config attached
07-26-2008 09:43 PM
Well I am not sure how, but I imagine this is dropping the calls.
notify telephone-event max-duration 3000 - seeing as its 30 seconds, just a guess :p.
To configure the maximum interval between two consecutive NOTIFY messages for a particular telephone event, use the notify telephone-event command in SIP UA configuration mode. To reset the interval to the default, use the no form of this command.
as far as the incoming calls, you need to enable this as a CUBE gateway, I think that will fix you up.
voice service voip
allow-connections h323 to h323
allow-connections sip to h323
allow-connections sip to sip
allow-connections h323 to sip
HTH, please rate useful posts!
Chad
07-27-2008 12:32 AM
Hi Chad,
Thanks for the reply, I have removed notify telephone-event max-duration 3000 and also added the allow connections however both problems still exist... Any other ideas?
Thanks,
Warwick
07-27-2008 04:28 AM
hi,
To troubleshoot dropped calls, use the command
debug ccsip messages..
and make a test call. after the call drops, post the out of the debug here
07-28-2008 12:45 AM
07-28-2008 01:25 AM
Hi,
I have looked the cause code...
Timestamp: 1217234131
CSeq: 309524721 BYE
Reason: Q.850;cause=16
Content-Length: 0
The value of 16 means the end user hung up. I know this is not the case, but its difficult to troubleshoot when you dont know the casue code..Infact its frustrating.
Sorry, I couldnt help as much as you want.
07-28-2008 02:47 AM
I agree.. I have asked the SIP provider for help. It doesnt look like an issue with the trunk, if i setup a SIP softphone directly registering with the provider calls work without a drama... Could this be a NAT related issue?
Any other ideas?
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