Hi,
sorry for cross-posting on both mailing lists, but I think a setup of Asterisk with OpenSIPS as frontend isn't unusual. So maybe both parties would be interested in this.
I'm using Asterisk (v1.4.21) to connect my OpenSIPS (v1.5.1) server to the PSTN (Asterisk connects to a local SIP provider doing the PSTN termination) so the Asterisk just acts as an PSTN gateway here. For doing some call control stuff (channel limitation) I'm using the dialog module on OpenSIPS.
Generally everything works well with about 250 users at the moment. But sometimes there are stuck dialogs on the OpenSIPS server (seen by #opensipsctl fifo dlg_list). At the same time in Asterisk messages there is this: [Apr 19 07:21:50] WARNING[13498] chan_sip.c: Maximum retries exceeded on transmission [email protected] for seqno 7912 (Critical Response)
As I'm doing channel limitation to a single channel by using the dialog module a stuck dialog leads to the user not being able to do any further calls until the dialog is destroyed by dialog timeout.
Any ideas how to solve this issue?
Regards, Robert.
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