similar to: cisco spa phones and sal

Hi I am trying to set up a hold system so that a call is always parked in the same spot no matter how many times it is picked up. My problem is I cannot fins a variable the identifies the call all the way through until it is destroyed. ${UNIQUEID} and ${CHANNEL} both seam to get lost when the call is parked. I tried setting set($[?${UNIQUEID}-hold?=?701?]) and

On 8/7/07, raviprakash sunkara <sunkara.raviprakash.feb14 at gmail.com> wrote: > > Hello Russell, > Nice To meet U and Good Morning. I got u r mail-Id from > http://www.asterisk.org/node/48325 > Recently i started the SLA configuration. But i didn't understand the > Flow of its Functionality > One of the My Client Ask to have do deploySLA feature > He Using

Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4? Specifically with the Aastra 55i. Specifically, I am using the Aastra 55i with the expansion module. We want to see if other handsets are being used. (BLF) Getting BLF to work would be a great start. It sounds like setting up the hints properly will achieve this. right? Not totally sure how this should be configured.

Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

Zhang, Xiantao wrote: >>From 5c70c038c57190144390ae9d30c3d06afba103d4 Mon Sep 17 00:00:00 2001 > From: Xiantao Zhang <xiantao.zhang at intel.com> > Date: Tue, 1 Apr 2008 14:59:30 +0800 > Subject: [PATCH] KVM:IA64 : Add kvm sal/pal virtulization support. > > Some sal/pal calls would be traped to kvm for virtulization > from guest firmware. > Signed-off-by: Xiantao

SAL is the directory of Scientific Applications on Linux. It is rather well done, I think. There is an entry for R http://SAL.KachinaTech.COM/F/1/R.html that mentions the Debian package but does not mention the RedHat package. I submitted a user-feedback comment to tell them about the RedHat packages and give the address of the CRAN archives.

The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * ---

hi friends, i'm researching to see if i can run my own email server on centos. i'm looking at qmail & centos, but the problem that i have is it doesn't have the web interface that can similar with outlook. our users don't like squirrelmail, or NOCC, or horde interfaces) however, i found atmail.com and it seems to have what i'm looking for b/c it's kindda

Hi I have asterisk 1.8.18 with freepbx 2.10.1.9. I get an asterisk crash occasionally with the followingerror. It always seems to happen while paging. 16 spa508g phones 1 snom pa1 paging amp Kelly == Extension Changed 1101[ext-paging] new state Idle for Notify User 101 pbx*CLI> *** glibc detected *** /usr/sbin/asterisk: malloc(): smallbin

I'm a newcomer to Ogg Vorbis and I am trying to get it to work on my Mac OS X 10.4.11 and Quicktime 7.5.5 so that I can play the following CBC radio stations through iTunes. http://www.cbc.ca/listen/ogg.html I tried XiphQT and couldn't get that to work so I downloaded Ogg Drop. It seemed to install fine, but I still can't get either of those radio stations to play either from

Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I'm a little disappointed with the support openvox, for some reason , The call doesn?t get trough support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly. the problem is that

Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility,

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

On 10-09-16 00:50, Richard Mudgett wrote: > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list > of my queues and then the following : > > > failed to extend from 240 to 327

We recently switched to atmail, as well as dovecot. I noticed in atmail the size of the mailboxes was always reported as 0kb. So, I did some debugging, and it boiled down to the fact that the regular expression used by dovecot expected UID before SIZE, but Dovecot returned SIZE before UID. No biggy, I changed the regex, but I was curious if there was a standard. Here's the imap query

It's long over due, but now I finally plan on upgrading from v1.0.15 to v1.2.16 on our servers. Would appreciate any comments! We have 7 servers, all running of the same GPFS filesystem. Both maildirs and indexes are stored here. We use dovecot for POP/IMAP and deliver. Currently we're using mysql as authentication source, but want to move to using ldap instead. Can we upgrade one server

Hi. Little story :-) I'm playing with dovecot 2.2.25 and multiple SSL certificates. ~7000 certificates which are loaded twice, so my dovecot has ~14 000 certificate pairs (14k key + 14k cert) in config. 14 000 local_name entries. Like these: local_name imap.example.com { ssl_cert = </etc/certs/cert1.pem ssl_key = </etc/certs/cert1.pem } local_name pop3.example.com { ssl_cert =