similar to: call initiation

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

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 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"/>

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

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

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 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

Is it at all possible to set a Global Variable freely whenever a context gets used without having to enter an extension priority to use SetGlobalVar? This is really limiting the dialplan for me. Heres an example of what I would like to be able to do. [globals] AREACODE= [local] exten=_NXXXXXX,1,Dial(SIP/${AREACODE}${EXTEN}/blah) [anyoldcontext1] AREACODE=313 include=local [anyoldcontext2]

Hello, everyone After having spent several time to look for any solution for my problem, I decided to write here. Here is the problem got a Digium X101 FXO card and Digium TDM400P alias freshmaker (1 fxs module on it on first port ) in my asterisk box. The X101 works perfectly. The problem comes from the TDM400, when i plug a tested working phone on it , i get no dialtone; the goal is to

Does anyone have a SEP<MAC>.cnf.xml file that works with asterisk? I have the SIP firmware loaded on my Cisco 7970 but the status log shows errors parsing the config. I copied a config that was posted to the list but it didn't seem to work. Any help would be appreciated. Jeremiah -- ______________________________________________________________ Rock River Internet

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

Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty

Thanks for your reply, [globals] AREACODE=381 [outbound] exten => _NXXXXXX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN},80) did not work for me, any ideas? Thanks, On 04/27/2015 01:59 PM, Phil Reynolds wrote: > > > On 27 April 2015 21:32:42 BST, Motty Cruz <motty.cruz at gmail.com> wrote: > >Hello, > > > >I would like to add area code if clients dial 7

Will this work to match any number from the 770,404, or 678 area codes? _[404|770|678]NXXXXXX If this won't work, is there a pattern that will do this? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> .

Hi, I was wondering how it was possible to access the actual cluster exemplars from the APResult class. Currently it only spits it out onto the terminal if you type the object but there is no other way to see which one is the examplar. Would appreciate any help. Thanks, Sachin [[alternative HTML version deleted]]

Hi all ! We have 3 NTBAs which are all going to our existing PBX. Our areacode is 06003 and our DDI enabled number 9141. I want to exchange that PBX with Asterisk, but still struggle to get it working. My CAPI.CONF is currently like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.800000 txgain=0.800000 [interfaces] context=in-capi msn=60039141 incomingmsn=* controller=1

Greetings, Currently my asterisk box is using Voicepulse. It works fine with the exception that people need to enter the 1+area code for local calls. I'd like to get around this if possible. The following is what I have in my extensions.conf.. exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=6162997590) exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) exten

Hi all, Can someone point me in the right direction here. What I'd like to do with Asterisk is a) dial a 3 digit extention (i.e. 100) on my polycom phones and after the 3rd digit is entered, it dials that extension and b) dial 9 to get out like older PBX systems. Since my internal extensions start with a 1 I think what happens is I enter extension 100 for example, and the phone sits

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission