similar to: Checking voicemail from outside

Here is what I have in my extensions.conf file now. Trustrcid and sendrcid are turned to "yes" in the conf file. I'm not fully sure how the SIPCalledRPID works though. The example I found seems to try and provide the stuff automatically (id and name), but does the SIPPEER stuff even exist? I think this is probably the right track though. Any insight would be much appreciated.

I've gotten this Polycom 501 pretty much licked, but I need to know if there's a way in a dialplan to say if someone dials their own extension it goes straight to voicemail and asks them for their password. I thought I saw an example of this on the web but I can't seem to find it. Any advice appreciated! Phil -------------- next part -------------- An HTML attachment was

Hello, I am using macros to give the ability to a call-receiver to 'accept' a call. However, any keypress connects the caller. Anyone have any suggestions about how to re-engineer this so that the receiver can deny the call, or press other keys to do other actions, without connecting to the user? Thanks, Jason Wolfe

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

Hi Someone may give me an example of followme() application using in a dialplan (including what to configure in followme.conf) ? I use asterisk 1.6.1 so if your example can match to that release it's will be wonderfull. Thank in advance. Harry. -------------- next part -------------- An HTML attachment was scrubbed... URL:

How do you enable call waiting on sip phones? Ive looked and googled and can only find call waiting pstn phones butnot for sip. Is their a way of setting this up within the dailplan?

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

I have let's say a reception that is comprised of 2 zap extensions and a mobile phone to dial using ISDN through Capi. I want to have a delay before starting dialing the mobile phone so that it rings only when the call has been unanswered for say 25 seconds. I tried to use Capi/2106994444:ww6935555555 but without any success. There is any way to do it or the code has to be modified ? Thanks

Hi all, I've gotten asterisk installed on Debian only to realize that the packaged version is 1.0.7. Is there a reason why they're not up to a 1.2.x release? I'm building a system for production and I'm wondering if I should remain at this old version or if there are any serious issues with 1.2.13 on Debian? Should I be able to do an apt-get from unstable and get 1.2.13 and

Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Best regards,

Hey all, I've been doing a lot of playing, and a lot of reading, and it seems people are split as to whereas if they're running their favorite Linux distro and asterisk or Trixbox. I'm getting closer to really looking at a production environment and I'm just looking for any opinions. I'm really enjoying learning linux and asterisk, so initial "ease of use"

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

I've been banging my head on a brick wall for about an hour now trying to understand why the following doesn't work (which is even provided as an example in the distribution!). The goal is to create a voicemail-only extension not associated with a phone. I'd rather not have an extension dedicated to VoicemailMain(), so I would like the user to be able to hit '*' during

Dear All I want to integrate sugarcrm and asterisk , so when customer call the call center the agent or extension which answers the call , before pickup the phone and talk to customer , view his/her information if it is available. I do this as described below : 1-Setup login username for sugarcrm for each extension 2-Extension Users will login to the sugarcrm. 3-Develop php script to handle new

Hi, is there a way to retrieve a VM message pressing some key during the greeting playback? Our scenario is a PBX with analog trunks and no DID. There's a general mailbox and no way to assign a number to voicemail. I've seen these question before in this list, but seen no answer to it... Thanks, Renato

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

I'm trying to set up my extensions.conf file using some of the existing macros like stdexten, etc. while at the same time having two logically separate virtual PBX's (with no "default" context) and two trunks coming into separate contexts, i.e. one for residence and one for my at-home business. I noticed, however, that macro-stdexten depends on the "default" context:

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute