2016-04-25 22:06 GMT+02:00 George Joseph :Yes, I could successfully use th..
2016-04-22 12:41 GMT+02:00 Olivier :I dont exactly understand what happened but I could positively observe successfull T.38 negociation today in the above setup.For reference, the dialplan in Sender machine should include a Set(FAXOPT(gateway)=yes) statem..
Ive just discovered PJSIP s support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ?On a fresh 13.8.2, it doesnt seem but I may have missed somthi..
At the moment I plan to migrate from asterisk 13.7 to 13.8. Because of relatively frequent updates I am building asterisk from a directory that is updated via git switch to the desired branch.1. Will be updated pjproject patches with git pull?2. W..
Hello!I encounter the following problem (asterisk 11 and 13) with Teconisy as trunk provider with enabled qualify and default t1min (100ms):Teconisy most often doesnt answer the first invite before asterisk default t1min ended. Therefore asterisk se..
Hello! You have a new message, please readkenc@vipmar..
I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk). VoIPtalk calls are unauthenticated and reach me fine, but Andrews &Arnold calls are authenticated. The last call I successfully received was on Tuesday afternoon. Initially, ..
Id like to record the barged call… but whichever leg of the call I try to barge, my speaking is never recorded using MixMonitor. Any idea about the reas..
Im using the following Dial command syntax:Dial*(SIP/peer/exten!sip:firstname.lastname@example.org *), the SIP URIafter the ! mark should be set as To-URI in outgoing INVITEfrom Asterisk. It works, but problem is that To-URI formatting is a bit messed up, It looks as follows:*sip:sip:xyz@xyz…
Im giving PJSIP a try on an Asterisk 13 box. More specifically Im studying in a lab, how to configure T.38.My setup is:SendFAX –>– asterisk 11asterisk 13 –>– ReceiveFAXI can observe fax are successfully sent end received but Im failling to see ..