similar to: Cisco 7970 SIP Config

We wrote and submitted a patch to do this. Just modify app_directory.c and recompile. It adds a new flag "b" to the directory( ) app where you can have it use both first and last name. -= Info about application 'Directory' =- [Synopsis] Provide directory of voicemail extensions [Description] Directory(vm-context[|dial-context[|options]]): This application will present

I have a question - if a user calls up w/ blocked caller id I get the following on my phone Incoming call from asterisk This is the same on my Cisco 7940s and Polycom phones. For average users this is not intuitive at all.. I'd like to configure this so if I deploy this at a customer site it says "caller id unavialable". With the spelling done right.... Any ideas on how this

Users withing the office can dial a 3 digit extension and that will ring a phone. The problem I'm running into is you have to press xxx then press 'send or 'dial'. The pbx doesn't recognize a 3 digit number as an internal extension and automatically dial it the user has to initiate that call. Asterisk automatically initiates calls w/ 9+7 digits and LD calls,

I have some Cisco 7940's running SIP image 6.3 and a newphone account. Reguarding my dialplan I'm having a small issue. I'd like to dial 9,2,xxx-xxx-xxxx for a LD Nufone calls - however I also need to dial local phone numbers ie 9,2xx-xxxx Currently my dialplan looks like so <TEMPLATE MATCH="9,1.........." Timeout="0" User="Phone"/>

I recently had a bear of a time getting a Polycom Soundpoint 500IP up and registered.. Now that its registered I ran into a problem w/ the dialplan. Needing to dial x101 I'd dial 10 - then get a fast buzy.. Also making a local call - dialing 95551212- would give me a fast busy after the 7th digit - so 9555121.. Same w/ LD calls... This dialplan really got me down as I didn't find

I'd like to propose a change to the voicemail remove script found in the contrib directory of the asterisk source Currently the find command looks like so system('find '.$dir.'/'.$context.' -name msg????.??? -mtime +'.$age.' -exec rm {} \; -exec echo Deleted {} \;'); I'd suggest it be changed to system('find

I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked.... Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten => 107,1,Dial(SIP/102&SIP/107,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup exten => 107,102,Voicemail(b102@pstn)

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3

I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for

I currently have two Cisco phones, a 7960 and 7970. The 7960 has a SIP OS on it and the 7970 has a SCCP. When the 7960 powers up it loads OS79XX.TXT, SIPDefault.cnf, SIP000E3875266C.cnf, RINGLIST.DAT, and dialplan.xml. I have a Cisco SmartNet agreement with the phone so I have access to download the firmware. I recently purchased a Cisco 7970 phone and was in the process of configuring

Trying to use: Asterisk 1.4.2 Zaptel 1.4.0 chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0. The changelog has this entry: * channels/chan_zap.c, configure, configure.ac: If we receive ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, tzafrir) Also, update the configure script to make sure that we don't try to build chan_zap

I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 phones in a paging group. I have all the phones set up with an extra line that auto answers the dial from my paging extension when the primary line is not in use. All of these are operating correctly however the 7961/7970s all ring once and then auto answer so the person paging all the phones has the first part of his

Does anyone know what application I should place this function in? For example with the DB function I currently do something like this to add an entry to the asterisk database: exten => s,n,Set(DB(AGENT/${MACRO_EXTEN:1})=${CALLERID(num)}) To delete the entries I do something like this: exten => s,n,DBDel(AGENT/${MACRO_EXTEN:1}) DBDel is marked as deprecated in favor of the DB_DELETE

I've downloaded: asterisk-1.4.1 zaptel-1.4.0 I've compiled and installed zaptel. When I go to install asterisk I do: ./configure make menuselect I then take a look under the codec selection menu and I see that codec_zap can not be compiled. *************************************

I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten => 3010,1,SetVar(ALERT_INFO=<Bellcore-dr1>) ; selects Ringer exten => 3010,2,Dial(SIP/3010,15) I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

On Sat, 2016-12-03 at 21:25 +0200, Aki Tuomi wrote: > > On December 3, 2016 at 9:11 PM "Jeremiah C. Foster" <jeremiah at jerem > > iahfoster.com> wrote: > > > > On Sat, 2016-12-03 at 12:23 +1000, Noel Butler wrote: > > > On 03/12/2016 12:08, Jeremiah C. Foster wrote: > > > > > > > On Fri, 2016-12-02 at 10:48 +0200, Aki Tuomi

> On December 3, 2016 at 9:11 PM "Jeremiah C. Foster" <jeremiah at jeremiahfoster.com> wrote: > > > On Sat, 2016-12-03 at 12:23 +1000, Noel Butler wrote: > > On 03/12/2016 12:08, Jeremiah C. Foster wrote: > > > > > On Fri, 2016-12-02 at 10:48 +0200, Aki Tuomi wrote: > > > On 02.12.2016 10:45, Jonas Wielicki wrote: On Freitag, 2.

Thank's Philipp and Dmitrijs. That got it fixed. I just pointed the web and admin settings where they belong and all's working great. Thanks for the sighup help, Dmitrijs. Jeremiah Rogers Cell: 704-996-5334 Email: jeremiahzrogers at gmail.com Social Networking: /jzrogers > On Nov 2, 2015, at 08:35, Philipp Schafft <lion at lion.leolix.org> wrote: > > Good afternoon, >

I still can't get my Shoutcast DSP to connect to my Icecast 2.4.2. Is there someone who can help me offlist, or is there someone who has this working and can provide a config file? Thanks! Jeremiah Rogers Cell: 704-996-5334 Email: jeremiahzrogers at gmail.com Social Networking: /jzrogers > On Sep 8, 2015, at 08:48, Jeremiah Rogers <jeremiahzrogers at gmail.com> wrote: > >