Hello Guys.
I have been reading through the forums but haven't seen this sort of a behaviour being reported elsewhere.
I have two OXO's (R910/042.001 and R700/015.001) that have a SIP trunk each to an Asterisk box. The trunks establish without any issues i.e. inbound and outbound calls all operate without any problems e.g. delays or call quality. SIP trunks for external calls sit on the Asterisk, so all external calls to the OXO's are handled by the Asterisk. Due to some limitation on the Asterisk side, the SIP trunk's username has to be in the format xxxx*xxx where x is a numeric digit. As the OXO's do not allow the usernames in this format, I have had to turn registration off - instead the systems just point at each other's IP addresses to route calls i.e. OXO to Asterisk. Both OXO's are independently connected to the Asterisk.
I have users complaining (on both OXO's) that calls drop mid-conversation (at random intervals into the call, however majority seem to be around the 16 min mark) - to be specific, the call doesn't drop as such however the RTP stream does and hence neither parties in the call can hear each other. They assume that the call has dropped, so they hang up and redial. This seems to happen very frequently however not on every call.
I am pretty confident that the Asterisk and its external SIP trunks are fine as we also have an Avaya IPOffice 500 connected via SIP trunks to the Asterisk which has no problems whatsoever (although the Avaya is much more forgiving when it comes to usernames, so we have the Avaya registering with the Asterisk).
I have looked at various traces, however I cannot seem to dig any useful information out of the logs - I have also looked at the router (Draytek 2830) to ensure that this is some to do with session timers.
I do have a feeling that this is related to keep alive messages which may be going missing due to the OXO's not registering with the Asterisk.
Is there anyone else out there who has seen this sort of an issue who could possible shed some light? Appreciate all the help as this has been going on for nearly 6 months and the users seem to be getting pretty edgy about it now!
OXO to Asterisk SIP Trunk - Call drop midconversation
- cavagnaro
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Re: OXO to Asterisk SIP Trunk - Call drop midconversation
Post SIP traces for a sample call
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