IVR issue in asterisk using with Dinstar GSM

english

#1

Hi!
Before i’m sorry for my english my first language is French
please i have a issue using asterisk with a GSM gateway.

I use asterisk with a GSM gateway DINSTAR DWG2000F to make IP calls to GSM as part of the establishment of a call center. I set up an IVR on asterisk that works great when IP calls to IP. without the IVR the IP to GSM and GSM to IP calls work perfectly well. but by integrating the IVR, during incoming calls (GSM to IP) if the caller chooses the option to talk with an operator, the caller is put on hold and the operator’s softphone starts ringing; when the operator picks up, the caller is still waiting and neither end hears the other: no conversation is possible.
please help me to see more clearly.

when i scan the communication between the GSM Gateway and asterisk with wireshark, i can see a reINVITE packet send by asterisk to the GSM Gateway. i think that why the caller is put on hold.

i try to enable this directive in sip.conf file but nothing i have the error.

i think it’s my asterisk the issue…

Please an idea, a track, …
Thank you


#2

Hi,

general sip.conf should contain canreinvite=no by default. So in case that you have no calldeflection on the used trunk enabled there should never be an reinvite initiated from the asterisk. In case that calls direktly distributed to an user works the problem seems to be lated to the ivr callflow, are there some custom scripts or drop to external numbers included?

Best regards,
Steve


#3

Hi Steve
thank you for your reply.
i don’t have any custom scripts, what do you means about external numbers ?
because i try to make a incoming call from GSM Number.

Best regards
Seb


#4

Sure, but the call is distributed to a local sip peer not to an external destination through the same GSM Gateway or another trunk, right?


#5

Yes Steve, the call is distributed to a local sip through the GSM Gateway.
if you need some files or architecture to help you to more understand, please let me known.


#6

only deeper digging into sip traffic and/or asterisk cli can clarify this. Sadly I can not support you on this here, maybe gateway manufacturer or provider can assist you.


#7

Hi Steve
i discussed with Dinstar GSM Gateway support, and he told me that:
in a moment my asterisk server send a packet reINVINTE to the GSM Gateway and this why the caller is put on hold.
i analysed the directive “canreinvite” in “sip.conf” file and put the option to cancel the sending of reINVITE packet. but i have the same behaviour.
So i will continued to search a solution
thank you Steve for your assistance.