similar to: chanstats console errors

Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone call to a specific number and make an announcement? I imagine the first part is the big question. joe a.

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

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./

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 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

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

I have regularly (once a week, once per few hundred calls?) been having problems with Asterisk's SIP stack not responding to packets from any of my registered devices. In the past, I could not tolerate the outage, so i would restart asterisk to make things happy. My Asterisk server is currently in this broken state and I can leave it this way for a short while. Devices are registered to

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers I was getting this : [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf after fix global issue

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or,

Thanks but no Adtran here. I do think these stats are indicating an issue, I just don't know how to prove it outside Asterisk. From: EWieling at nyigc.com To: tjrlist at live.com; asterisk-users at lists.digium.com Date: Mon, 19 Jan 2015 13:55:33 -0500 Subject: RE: [asterisk-users] sip show channelstats reliable? I?ve seen something similar with Adtran SIP gateways. When a re-invite

On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog <sgriepentrog at digium.com > wrote: > I would recommend capturing traffic outside your Asterisk server with > Wireshark, then running the Telephony/Rtp/Analysize Streams option to > determine if you have packet loss at that point in the network. > > On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote:

Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable. Peer Call ID Duration Recv: Pack Lost ( %)

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

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've seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is

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'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 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

I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > Thanks but no Adtran here. > > I do think these stats are indicating an issue, I just don't know how to

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