search for: jeremykister

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.

...TML attachment was scrubbed... > URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/6fa782f1/attachment-0001.html> > > ------------------------------ > > Message: 5 > Date: Tue, 28 Apr 2015 16:01:38 -0400 > From: Jeremy Kister <asterisk-03 at jeremykister.com> > To: asterisk-users at lists.digium.com > Subject: [asterisk-users] Asterisk 13/PJSIP + registration > Message-ID: <553FE722.6000906 at jeremykister.com> > Content-Type: text/plain; charset=utf-8; format=flowed > > Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4...

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./

Running Asterisk 10.12.2 on Debian/sparc i'm doing all sip/rtp. directmedia=yes directrtpsetup=yes I frequently see on the console: WARNING[7832]: chan_sip.c:19134 show_chanstats_cb: Could not get RTP stats What is this error trying to tell me ? 'sip show channelstats' shows all 0s (save Peer/CallID/Duration) I looked for that string in the source but i didnt learn much.

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 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

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

..... >> URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/6fa782f1/attachment-0001.html> >> >> ------------------------------ >> >> Message: 5 >> Date: Tue, 28 Apr 2015 16:01:38 -0400 >> From: Jeremy Kister <asterisk-03 at jeremykister.com> >> To: asterisk-users at lists.digium.com >> Subject: [asterisk-users] Asterisk 13/PJSIP + registration >> Message-ID: <553FE722.6000906 at jeremykister.com> >> Content-Type: text/plain; charset=utf-8; format=flowed >> >> Using Asterisk 13.3.2 on C...

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

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

For anyone interested, Allison Smith's AMA (not sure she's still around): http://www.reddit.com/r/IAmA/comments/2rrb7m/iama_professional_telephone_voice_ama/ -- Jeremy Kister http://jeremy.kister.net./

Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4, I can't make asterisk try to send a register. I have configured my pjsip.conf similar to https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples#res_pjsipConfigurationExamples-ASIPtrunktoyourserviceprovider,includingoutboundregistration my pjsip.conf: http://pastebin.com/raw.php?i=EA0PEcrb using tcpdump, I never even

using app_page on asterisk 1.6.1.6, as documented, the 'q' option only determines if the caller is sent a 'beep' tone when conferencing. is there a way (existing or someone sending me a patch) to also make app_page beep all of the extensions being called? someone adding an 'a' (announce tone) parameter to app_page would be perfect. with auto-answer turned on with my

Using Asterisk 1.6.1.9, I'm looking for a way to "share an extension" between a SIP phone (Cisco 7940) and a SLT on a FXS port of a Cisco 1760 (via sip) -- at any given time I want to be able to pick up either phone and it should be "bridged" to the other - just like having two SLTs on the same copper pair. The goal is to have a cheap cordless telephone sit right next

I'm trying to figure out how calls are ending up in my default context (which should never happen). I've got a Cisco 1760V with a VIC-2FXO-M1/VIC-4FXS and 5 Cisco sip phones. When I make a call from one of the FXS ports on the 1760, the call goes into asterisk's default context instead of where i think i'm directing it. Can someone tell me what I have misconfigured? 1760

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

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've got the parking lot set up in asterisk 1.6.2.8. when a caller calls into the pbx and connects to an extension, the answering extension can place the person into the parking lot by dialing #72 during the call via the parkcall parameter in features.conf. this works just fine. however, when another extension picks up the call from the parking lot (e.g, by dialing x701), that extension

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

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./