search for: rmudgett

On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:55:28 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > > > > > On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota > > <murthy64 at hotmail.com<mailto:murthy64 at...

...by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release: * Prevent a CONNECT message from sending a CONNECT ACKNOWLEDGE in the wrong state. (issue #17360. Reported by: shawkris. Patched by rmudgett) * Made Q.921 delay events to Q.931 if the event could immediately generate response frames. (closes issue #17360. Reported by: shawkris. Patched by rmudgett) * BRI PTMP: Active channels not cleared when the interface goes down. (closes issue #17865. Reported by: wimpy. Patch...

...by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release: * Prevent a CONNECT message from sending a CONNECT ACKNOWLEDGE in the wrong state. (issue #17360. Reported by: shawkris. Patched by rmudgett) * Made Q.921 delay events to Q.931 if the event could immediately generate response frames. (closes issue #17360. Reported by: shawkris. Patched by rmudgett) * BRI PTMP: Active channels not cleared when the interface goes down. (closes issue #17865. Reported by: wimpy. Patch...

On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > <snip> > >> Here is the CLI command used: > >> > >> ubuntu*CLI> originate SIP/732-xxx-xxxx at vonage-out a...

Hi, I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel => 1-23

...getting loaded at first. Is it expected that if bridge_softmix handled a normal two party call then MOH would no longer function? ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Richard Mudgett <rmudgett at digium.com> Sent: 09 December 2014 20:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%] On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont <p.beaumont at hatsoffsoftware.co.uk<mailto:p.beaumo...

...have not been possible without your participation. Thank you! The following are some of the issues resolved in this release: * Fix issue where calling name is not successfully processed on inbound QSIG PRI calls from Mitel PBX. (Closes issue #17619. Reported by: jims8650. Patched by rmudgett) * Added missing code specified by Q.921 (Figure B.8 Page 85) when receive RNR in "Timer Recovery" state. (Closes issue #16791. Reported by: alecdavis. Patched by alecdavis) * Fixed issue where incoming calls specifying the channel using a slot map could not negotia...

...isk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) * Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read. (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett) * Resolve issue where wait for leader with Music On Hold al...

...have not been possible without your participation. Thank you! The following are some of the issues resolved in this release: * Fix issue where calling name is not successfully processed on inbound QSIG PRI calls from Mitel PBX. (Closes issue #17619. Reported by: jims8650. Patched by rmudgett) * Added missing code specified by Q.921 (Figure B.8 Page 85) when receive RNR in "Timer Recovery" state. (Closes issue #16791. Reported by: alecdavis. Patched by alecdavis) * Fixed issue where incoming calls specifying the channel using a slot map could not negotia...

...isk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) * Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read. (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett) * Resolve issue where wait for leader with Music On Hold al...

...c for a non QSIG card, and change to switchtype=euroisdn in chan_dahdi.conf. Would DAHDISendCallreroutingFacility then do the equivalent ETSI methods to reroute the call? I may be able to test this over the weekend, in the mean time, I thought I'd ask, if this was the correct way, or if mattf, rmudgett or others had 'team' branch that is a work in progress that we can perhaps have a look at. Alec Davis -----Original Message----- From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Sa?l Ibarra Sent: Thursday, 19 November 2009 12:28...

Hi Richard, Thank you for your response. But after I remove the parameters of dial command (tTkK). The call was still not native bridge. Let me know if you have any suggestion. Best regards, Charles 2015-01-30 0:34 GMT+08:00 Richard Mudgett <rmudgett at digium.com>: > > > On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang <lazy.charles at gmail.com> > wrote: > >> Hi all, >> >> I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk >> 11.14.2 and DAHDI 2.8.0. >> >> I try t...

...ialplan application - all this "dialling a local channel" is just to get via music on hold so there's not silence while a long operation happening, as per http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html :) ) On 27 November 2016 at 16:16, Richard Mudgett <rmudgett at digium.com> wrote: > [svtest1] > exten = s,1,NoOp() > same = n,Answer() > same = n,Set(__MY_CALLER=${CHANNEL(name)}) > same = n,Dial(Local/s at svtest2,,g) > same = n,NoOp(Returned SHARED(sharedVar) = '${SHARED(sharedVar)'} > same = n,Hangup() > > [svtest2]...

...in this release: * Only offer codecs both sides support for directmedia. (Closes issue #17403. Reported, patched by one47) * Resolution of several DTMF based attended transfer issues. (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett) NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) * Fix channel redirect out of MeetMe() and other issues with channel softhangup...

...in this release: * Only offer codecs both sides support for directmedia. (Closes issue #17403. Reported, patched by one47) * Resolution of several DTMF based attended transfer issues. (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett) NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) * Fix channel redirect out of MeetMe() and other issues with channel softhangup...

On Tue, Jun 26, 2018 at 7:59 PM, Richard Mudgett <rmudgett at digium.com> wrote: > > > On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender <dovid at telecurve.com> wrote: > >> I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so >> often asterisk crashes and then restarts. I am not seeing any core dumps on >&...

I'd be grateful if anyone here could comment knowledgeably on an idea that I have had, as to whether it should be possible or not. Consider two Asterisk boxes, each with one or more E1s on EuroISDN. Each box has a different telephone number that hunts across all its E1 channels. In addition there is another number that hunts across all the channels on all the boxes. A call comes in to one of

On Tue, Apr 26, 2016 at 11:07 AM, Administrator TOOTAI <admin at tootai.net> wrote: > Le 26/04/2016 17:23, Mamadou NGOM a ?crit : > >> Hello, >> >> >> Having installed DAHDI to be able to use the meetme() application , when >> I start the dahdi service it generates me the following error: >> >> -bash: /etc/init.d/dahdi: No such file or

...lled). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with "core show applications" Do I need to install mISDN or other modules for using that ? Regards M.Shirazi --- On Mon, 2/27/12, Richard Mudgett <rmudgett at digium.com> wrote: From: Richard Mudgett <rmudgett at digium.com> Subject: Re: Libpri To: "Mehdi Shirazi" <mahdi_shirazi at yahoo.com> Date: Monday, February 27, 2012, 11:55 AM > Is it possible to do this with Libpri: > > exten => _X.,1,Set(_SS7_LSPI_IDEN...

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at lists.digium.com >