[asterisk-users] Respond to an out of call SIP MESSAGE
Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf [sip-im] exten _X!, 1, NoOp(Got message) exten _X!, n, Answer() exten _X!, n, Agi(agi://localhost/messagehandler.agi?...) exten _X!, n, SendText(Message received) I can see in the log from Asterisk that the script in the sip-im context is running, but there is no message sent. I have followed the code in the call, and it seems like there is no channel registered with the SendText application. Is there some other approach that I could use to send a SIP MESSAGE back to the client? Does the client need to register for this to work? BR, Emil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to enable SIP text messaging with PJSIP ?
Hi sir , How to enable SIP text messaging with PJSIP ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Voicemail MWI
On Wed, Sep 16, 2015 at 04:44:45PM -0400, Nick Olsen wrote: > I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, > These are loaded into asterisk without the mailbox info. Leading to > "Received SIP subscribe for peer without mailbox" notices. And non-working > MWI. > > Occasionally, It just works. But only on a peer or two at a time. And > it'll stop working after a few minutes. Here it seems to be the other way round. Occasionally I see that peers have lost there mailbox setting and don't get notify messages with voicemail information. It is Asterisk 13.5.0 "sip prune realtime peer ..." "sip show peer ... load" After this the setting is restored, but until now I have no idea why this happens. The database field mailbox remains unchanged. Could you post the Realtime SIP Settings? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
