similar to: asterisk 11 AGI

I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. All my custom modules (including swift <thanks darren!>) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip

With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while I am listening to the playback, i interrupt and dial: - "12345", SWIFT_DTMF is set to

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? -- Jeremy Kister http://jeremy.kister.net./

using asterisk 11.6.0-rc1 i just converted my "nat=yes" to "nat=auto_force_rport,auto_comedia" I have my asterisk box on the same subnet as a cisco 1760 (vgw1). a few times per day, Asterisk thinks vgw1 is dead (by qualify/options). A 'sip reload' always fixes the problem. i left 'sip set debug peer vgw1' on the console. but i dont see what's

running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the

I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' - No matching peer found my logger.conf

i'm trying to review issues that i'm monitoring and/or have reported at http://issues.asterisk.org when I click on 'My View' or 'View Issues' I get an error: APPLICATION ERROR #401 Database query failed. Error received from database was #1142: DELETE command denied to user 'mantisreadonly'@'localhost' for table 'mantis_tokens_table' for the

When I place an outbound call from asterisk 1.6.1.12 to a FXO port on my Cisco 1760V 12.4, the channel changes - seemingly incrementing: e.g., in the first call, below, the channel name is "SIP/vgw1-00000075" -- the second call (on the same FXO port after a soft hangup on the CLI) is "SIP/vgw1-00000077" How can I extract this information in the dialplan so that I can use

somewhere along the way, i noticed incoming calls from google voice are no longer working on my asterisk 1.8.3.2 system. When the call comes in, asterisk immediately prints on the console: == Spawn extension (google-in, s, 2) exited non-zero on 'Gtalk/+12153930924-f947' [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to

just thought this was cute enough to pass along, https://www.youtube.com/watch?feature=player_detailpage&v=GLwct15X_3g#t=135 -- Jeremy Kister http://jeremy.kister.net./

I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. The system itself is happy and phone calls (between two parties) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. To

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

Since digium is apparently blind to users of their Free Fax for Asterisk, does anyone have advice on how to report a crashing problem with res_fax_digium and Asterisk 1.8.2 ? I have detailed logs/reports and a backtrace ready, but I have no idea who can help. -- Jeremy Kister http://jeremy.kister.net./

Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3 when i start asterisk, i immediately see two mpg123 processes spawned which sit there forever. I can't imagine it's normal behavior, but if it is, please explain :) # /etc/init.d/asterisk stop stopping asterisk. #[...] # /etc/init.d/asterisk start starting asterisk. # psg aster root 14573 1 0 16:29 pts/2 00:00:00

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio

ok it works i had a problem with a syntax: i had to wrire: exten =>_!X.,n(external),Dial(SIP/011212664800450 at pstn2,,S(20)) thanks ________________________________ De : Adil Zaaraoui <adilzeaaraoui at yahoo.fr> ? : Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Envoy? le : Jeu 8 juillet 2010, 19h 41min 15s Objet : Re :