Archives : November-2014
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Is there an equivalent to ${SIPPEER(${peer},status)} for PJSIP?The closest Ive been able to get is to use AST_SOURCERY to see if they have a contact${AST_SORCERY(res_pjsip,aor,${peer},contact) but Im not certain if Ill still have a contact entry af..
HiI have noticed that the code send by asterisk to some other party when a peer, represented by a soft phone, hangs up is 486 (Busy here). The problem I see is that when the user does not answer at all the same code is send by asterisk to the other par..
If Im not mistaken, it is not possible to get T.38 on a SPA3102 FXOport (it is possible with the FXS port). Do you know, by experience preferably, if this is possible with an SPA8800 FXO por..
Hellohttp://attivazionehosting.misterdomain.eu/jack.php?teeth=3qvnrhqvf56y7grschroe@gmail.comSent from..
We upgraded from asterisk 11 to asterisk 13.Recordings were working fine in 11 but nothing is being written on 13.Here is the dialplan segmentsame => n,ExecIF($[${TL_PHONE_CALL_RECORD}=TRUE]?SET(CONFBRIDGE(bridge,record_conference)=yes))same => n,ExecIF($[${TL_PHONE_CALL_RECORD}=TRUE]?SET(CONFBRIDGE(bridge,record_file)=/var/spool/asterisk/confbridge/${TL_PHONE_CALL_ID}.wav))s..
do you know if it is possible to define the SLA configuration in the database for realtime usage with asteri..
Howdy,Is there a way to use realtime with phoneprov.com and pjsip?Ive got a working pjsip realtime config currently but I have to add a phoneprov section to my pjsip.conf for each phone I want to provision.I was hoping the Sorcery page in the wiki wo..
Morning,We recently pushed our Asterisk video bridge into a DMZ and since then, local calls have been unreliable to say the least. While offsite calls work nicely, calls on our internal server usually fail to ring the far end. Two test calls that w..
2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes : I solved my issue by changing type=friend with type=peer in [555555555] section, afterwards, googling Ive found this article that explain me why: http://forums.digium.com/viewtopic.php?t=79338#p161214 ..
tengo la siguiente pagina pero no se como seguir despues del punto 22http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.ht..