similar to: More testing

I am unable to force a hangup on a channel that has been stuck for over two days: IAX2/from-CD-11006 oficina 2770 1 Up Dial IAX2/to-CD/2883 3467130007 46:24:59 Sotelo Sotelo IAX2/to-CD-20713 I have tried "hangup request IAX2/from-CD-11006" several times but no joy. I also see the following in the CLI: [Nov 3

Anyone is using 1.6 in production?? Is it ready? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081107/df5bb63a/attachment.htm

Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Thanks in advance! System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card OS: Centos 5 Kernel: 2.6.18-53.1.14.el5 (also tested under

In article <CAHZ_z=w5DMg93gShtC93kuC+fnmraPgV46BS956U5BQXVgyhxg at mail.gmail.com>, Matt Fredrickson <creslin at digium.com> wrote: > On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield <tony at softins.co.uk> wrote: > > I have some more investigation to do on this, but I wanted to see if anyone > > here had any insight into the issue I've run into. > > >

Hi Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expected Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/7e14a4e0/attachment.html>

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

All, I'm having a heck of a time getting hdlc to work on kernel 2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the kernel (note into, and not 'modules'). System comes up, I configured zaptel.conf span=1,0,0,esf,b8zs nethdlc=1-24 modprobe wct4xxp ztcfg sethdlc hdlc0 cisco ifconfig hdlc0 up All of this works fine, believe it or not. I have a T1 cross plugged

Sorry, I forgot to write that the SIP peer must keep ringing while the announcement is being played. Le 22/08/2016 ? 17:42, John Kiniston a ?crit : > This seems like the obvious answer but maybe I'm misunderstanding the > question. > > exten => s,1,Dial(SIP/alice,20) > same => n,Playback(myannouncement) > same => n,NoOP(Whatever else you want to do goes

Hi, I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Thanks a lot in advance, best

Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this

That's just what I'm using, John. But I'm getting (eg) [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:449 file2format: Cannot open '/home/logs/anonymous.txt': No such file or directory [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:949 file_write: File '/home/logs/anonymous.txt' not in line format Asterisk is running as root (yeah, I know!), and has

Hello I have contacted my line provider who is saying that in order to get my 0845 or 0870 number to id as the incoming number on a landline that i may call i need the following. User must provide - NPI set to E.163/E.164 User must provide - TON = "national or international I have had a good search around and can't seem to find a good answer to this. Does anyone have any idea where i

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. >

I'm implementing a SBC with my Asterisk PBX but the keeps disabling the trunk group I've configured and I think it may be because Asterisk is returning a 4r04 to the OPTIONS. I've created a test context and have put in a wildcard pattern match to try and catch those options but it doesn't seem to work. Is there a way to have asterisk respond with an 200 OK instead of a 404? --

Hi, 1. How do you then, synced then unread message presence with custom device status ? From an external program ? When a user leaves VoiceMailMan application ? Using externnotify ? 2. What is MWI:101 at default expression for (see [2] ? Cheers [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at

Hi, My setup is: IPPhone1 --- Asterisk1 with B410P ---- Patton 4638 --- Asterisk2 --- IPPhone2 I want to evaluate Asterisk1 in TE/PtmP mode. So, Patton box is configured in NT/PtmP (with 3 BRI links between both systems). Anyway, asterisk -rx "pri show spans" keeps replying : PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active PRI span 3/0: Provisioned,

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the

John, >From looking at the wiki won't STRFIME just give me what I need based on the unix time that I put in? What I am actually looking to do is convert over from 12 hour format to 24 (unless strftime does just that and I don't kow what am I am doing?). On Thu, Feb 13, 2020 at 12:03 PM John Kiniston <johnkiniston at gmail.com> wrote: > Try using the STRFIME function

My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John,

thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new