search for: cunningham

...the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111121/476e9edf/attachment.htm&gt...

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > Does anyone know a way with chan_sip to tell Asterisk to use a specific IP > address for its end of the communication for a specific device? Something > like: > > [device] > type = friend > host = 11.22.11.22 >...

----- Original Message ----- From: "Gene Cumm" <gene.cumm at gmail.com> To: "Bill Cunningham" <billcun at suddenlink.net> Cc: "For discussion of Syslinux and tftp-hpa" <syslinux at zytor.com> Sent: Saturday, September 06, 2014 9:27 AM Subject: Re: [syslinux] bootable dvd > On Sat, Sep 6, 2014 at 9:10 AM, Bill Cunningham <billcun at suddenlink.net> &gt...

OK, thank you George. On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote: > > > On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hi George, >> >> Thank you for the response. I'm a little unclear on what you mean by a >> transport. We're using chan_sip, not pjsip. >> >> Do you mean a device in sip.conf, using bindaddr to set...

Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? Will Cepstral work with 1.6.x? Thanks, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

----- Original Message ----- From: "Bill Cunningham" <billcun at suddenlink.net> To: "Gene Cumm" <gene.cumm at gmail.com> Cc: "For discussion of Syslinux and tftp-hpa" <syslinux at zytor.com> Sent: Saturday, September 06, 2014 9:31 AM Subject: Re: [syslinux] bootable dvd > > ----- Original Message -...

...one B answers the call, phone C dials something to "steal" the call from B, and finally A and C are talking. Searching on voip-info.org shows a "BristuffSteal" command but it's very out of date (Asterisk 1.2). Thanks in advance for any suggestions. Kind regards, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180709/495bd0c1/attachment.html>

I have just jined this list. I can't seem to boot from a USB and my BIOS recognizes my sandisk USB by name. I can't figure out what modules to use. mbr.bin and parted is one method. Then syslinux is run and a syslinux.cfg is needed. Is this sufficent for syslinux? LINUX bzImage //kernel image INITRD initrd // initial ramdisk to boot a small liinux system on a USB. Bill Thanks for any

Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130103/d33f5eef/attachment.htm>

...Thank you. On Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote: > Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass > it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio > > On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> > wrote: > >> Hello, >> >> Does anyone know a way with chan_sip to tell Asterisk to use a specific >> IP address for its end of the communication for a specific device? >> Something like: >> >> [device] >> ty...

...send them to our resellers. Would it be possible to have a fixed address for a particular asterisk release that will never (or at least not for a long time) change? Perhaps put all (except very old) versions in the same directory, with a 'latest' link to the latest one? -- Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 sip:acunningham@integrics.com http://integrics.com/

I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like > STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: > > if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) > && STRICT_RTP_LEARN_TIMEOUT <...

...o set the address to bind for that device? We've only used bindaddr in the [general] section before, but if it will work in a device that could be the answer. On Fri, 23 Oct 2020 at 00:13, George Joseph <gjoseph at digium.com> wrote: > > > On Wed, Oct 21, 2020 at 9:16 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> We have an Asterisk server with two public IP addresses, let's say >> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with >> a call dialled from Asterisk to an external d...

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording. Currently we're using the AGI command "record > > file" which does allow us to specify which DTMF keys can end the > > recording. > > > &g...

After a few hours of running, I get tons of the following errors in my logs: dovecot: Oct 08 07:41:50 Error: auth(default): ldap(user at domain,x.x.x.x): Request queue is full I removed the username and IP, obviously. Any idea how to stop this? I have about 5 Thousand users using horde that login ever 1-5 minutes to refresh their page. I assume it is a setting, but I am confused as to

On 8/27/2014 18:05, Bill Cunningham wrote: > I have just jined this list. I can't seem to boot from a USB and my > BIOS > recognizes my sandisk USB by name. I can't figure out what modules to use. > mbr.bin and parted is one method. Then syslinux is run and a > syslinux.cfg is > needed. Is this sufficent...

....11.22 ouraddress = 33.44.33.44 This is for use on a server with multiple IP addresses. There is the "extenip" setting, but it's really designed for NAT, and can only appear in the [general] section. Any suggestions would be greatly appreciated. On Sat, 24 Oct 2020 at 09:43, David Cunningham <dcunningham at voisonics.com> wrote: > OK, thank you George. > > > On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < >> dcunningham at voisonics.com> wro...

...file. [2003/01/02 15:46:36, 1] smbd/reply.c:(909) Rejecting user 'cunning': authentication failed Nothing has changed on the server where samba runs. Could something like growth in the number of accounts on the domain controller cause this ? Any other suggestions ? Dave Dave Cunningham Computing and Network Services (408) 756-1382 Voice, (408) 539-4912 Pager dave.cunningham@lmco.com

...p <jcolp at sangoma.com> wrote: > It's probably best if you read the logic[1]. There's an entire comment > that talks about how it works. > > [1] > https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 > > On Mon, Apr 17, 2023 at 7:10 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hi Joshua, >> >> Could you confirm if the 5 second period for learning a new audio stream >> is a minimum or a maximum? The unusual call flow in question results in >> Asterisk learning a new audio stream when w...

...re is no gluster related process left, then start gluster on br first and check the `gluster peer status` then start gluster on sg. ( If you can take the downtime ? ) Thanks, Anant ________________________________ From: Gluster-users <gluster-users-bounces at gluster.org> on behalf of David Cunningham <dcunningham at voisonics.com> Sent: 23 February 2023 9:56 PM To: gluster-users <gluster-users at gluster.org> Subject: Re: [Gluster-users] Big problems after update to 9.6 EXTERNAL: Do not click links or open attachments if you do not recognize the sender. Is it possible that versio...