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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

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. > > > > The hardware...

...] sruffell <sruffell at localhost>: * kernel/wct4xxp/base.c: Work around for host bridges that generate fast back to back transactions which the current version of the quad span cards do not advertise support for. 2008-03-14 16:39 +0000 [r3983-3990] Matthew Fredrickson <creslin at digium.com> * firmware/Makefile, kernel/wctdm24xxp/base.c, kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update wctdm24xxp's VPMADT032 firmware to version 1.16 * kernel/wct4xxp/base.c: When doing the ISR rewrite, forgot to include the vpmdtmfcheck w...

More testing. Test test test. :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

On Tue, Apr 3, 2018 at 4:57 PM, Matt Fredrickson <creslin at digium.com> wrote: > On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield <tony at softins.co.uk> > wrote: > > In article <CAHZ_z=w5DMg93gShtC93kuC+fnmraPgV46BS956U5BQXVgyhxg@ > mail.gmail.com>, > > Matt Fredrickson <creslin at digium.com> wrote: > >&...

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

...risk-users > --001636c5ad1b56719904615f48aa Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable <br><br><div class=3D"gmail_quote">2009/1/24 Matthew Fredrickson <span dir= =3D"ltr">&lt;<a href=3D"mailto:creslin at digium.com">creslin at digium.com</a>&g= t;</span><br><blockquote class=3D"gmail_quote" style=3D"border-left: 1px so= lid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> <div><div></div><div class=3D...

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

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

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>

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,

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

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

Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

Sorry for disturbing you, but we have some problems with an installation with multiple (84) T1s from Quest. Now, our Problem is disconnected numbers are reported by sending in- band channel alert message and the B-Channel will have the tri tone and respective message but the line is never "picked up" and stays in ringing when dialling. So disconnected numbers are never detected as

...> 3. Re: ZyXEL 2000W (Jason Williams) > 4. Channel banks, voicemail, and immediate=no (Chris A. Icide) > 5. RE: STILL NO AUDIO (Eric Wieling) > 6. Re: STILL NO AUDIO (Holger Schurig) > 7. RE: Mac OS X installer for Asterisk (Wallingford, Ted) > 8. Re: PhoneGaim? (creslin@digium.com) > 9. Re: BroadVoice problems? (Chris Shaw) > 10. RE: STILL NO AUDIO (Sebastian Nocetti) > 11. Re: TDM400P Internal Extenion Config (Jason Williams) > 12. IP Phone recommendation (Yiannis Costopoulos) > 13. Re: Cheap PoE switches/injectors? (asteriskstuff@ziplip...

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

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. The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one of several systems that have been running without issue since 2010/2011. They have all been running CentOS 4 32-bit with Zaptel 1.4.12.1 (with patch for gen 5 card), libpri 1.2.8 and

Hello, I've been planning to replace my aging CENTREX switch with a new PBX and am seriously considering Asterisk as my solution. I work at a college, and we currently support just under 300 regular analog lines to the offices and whatnot. I was wondering.. Is asterisk ready for such a job? Would I be making a mistake to deploy this across the campus at this time? It seems that