For the mailing list archive and for anyone else interested.A few years ago we needed to automatically run a second AGI if the first AGI failed i.e. a failsafe setup.Mainly because Im not a very good programmer. 8-|The code below is very similar to w..
We are seeing something weird we havent come across before. It seems they are sending us a different IP in the SIP from URI, than the IP they are actually sending us the traffic from.Basically, the traffic is coming from 22.214.171.124 but the hea..
I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 126.96.36.199 is an Asterisk 11 box.[Oct8 11:59:27] NOTICE: rtp.c:849 process_rfc3389: Comfort no..
We are having a weird problem where calls get cut off in the middle.Im not a SIP expert but could the INVITE with an empty SDP be the problem?|Time | 188.8.131.52|| | | 184.108.40.206 | |9687.369 | INVITE SDP (g729 telephone-eventRTPType-101)|SIP Fr..
I would like to configure Asterisk send back only a Trying or Progress message to the SIP client and not any early audio for ringback. Ive confirmed Asterisk is sending RTP when the call is ringing by using rtp debug on Asterisk.Does anyone have ..
From voip-info.org:If SRVlookup is turned on, Asterisk supports DNS SRV lookups partially. Currently, Asterisk only reads the first SRV entry without bothering with priorities and weights.Is this still the case with Ast..
We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below.Offer expires Dec 31.We are a direct customer of Level 3, there is no other carrier involved.W..
Im trying to get T.38 passthrough working on Asterisk 220.127.116.11.It isnt working. Calls go through and are answered, but the fax machines are unable to communicate. I checked the value of CHANNEL(t38passthrough) and it seems to always be 0. One side..
We are getting this message on an Asterisk 1.4.44 box.[2012-11-19 08:49:27] WARNING app_voicemail.c: List of extensions is too long (>1323).Truncating.I know Asterisk removed many of limitations in string lengths in in 1.6+.Does anyone know..