similar to: How "real time" is realtime?

Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3

Hello, I'm new to this list. I'm trying to configure my Asterisk to have user access their email. SO far users can leave voicemail but they can't access voicemail. As you can see I had sip.conf and extensions.conf below. Please advice how to access configure extensions.conf to have users access their voicemail. Thanks in advance. -motty SIP.CONF [general] context=default

G'Day All; I am a little unsure on how to get Music On Hold to work. Please critique my extensions.conf. ????? Thanks ; SIP 5001 exten => 5001,1,Dial(SIP/5001) exten => 5001,2,Voicemail(u${EXTEN}) exten => 5001,3,Hangup exten => 5001,102,Voicemail(b${EXTEN}) exten => 5001,103,Hangup Thanks -------------- next part -------------- An HTML attachment was

On the wiki it say's that if you use the Goto commands you need to replace ',' with '|' in the app data field. But in the examples it uses '|' in place of ',' in the Dial command also in a couple of places. Is it safe to replace ',' with '|' everywhere in the app data field when using realtime? Or should I still to substituting ','

X101P with Ambient md3200 chip on it, with the zaptel wcfxo driver.... Just an indication of how many people have got this to work would be useful. Cheers Mark.

Mark, Got it. Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mark Phillips Sent: Thursday, December 16, 2004 6:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MusicOnHold. not getting it. This is well documented in the WIKI. And, it's not configured

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Adams, Gavin Is there any additional information I could provide to start tracking this down? I was thinking about looking into the various applications source to see how they access the data elements for callerid. I know where the values are pulled

Hey all, We've upgraded our PRI trunk to support what BellSouth calls "enhanced caller id name delivery". The weird part is, I'm only capable of seeing the name portion on incoming calls within voicemail2's email delivery. For example, on an incoming call, asterisk is reporting this: Context from extensions.conf (BS delivers 7-digit DIDs): exten => 9133727,1,Answer

Interestingly enough I had this same problem today.... 1. I created the directory and permissions for the directory " /var/lib/asterisk/festivalcache/ " (per the comment in the festival.conf file) 2. I had to comment out some things in the festival.conf file: the "host" line, the "port" line, and the "festivalcommand" line. I have also noticed the

Do you think this should be "Bug Reported"? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Samaritoni Sent: Wednesday, October 06, 2004 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Festival, getting gethostbynamefailed error

I am totally new to this project, I have been trying to set it up now for almost a whole day, and can not understand what is wrong. The following error appears when my phone tries to register: Oct 6 19:08:30 NOTICE[98310]: chan_sip.c:7519 handle_request: Registration from '<sip:2201@192.168.0.5;user=phone>' failed for '192.168.0.253' What exactly does this error mean,

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Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

I have been having problems getting PrivacyManager to work correctly. Right now I am running the 1/21/05 CVS but I have been unable to get this to work on asterisk-stable either. You can see from the debug below that the inbound call is arriving via IAX2 and both the CALLING NUMBER and CALLING NAME are both set as "Unavailable". However, PrivacyManager executes and determines that

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

I think I understand what''s going on, thanks to a small mistake ;-) I needed to add a 5th nic to the gateway box - this new nic was an identical mate for another isa in there, so I modified the module options accordingly. So, after 10 minutes of trying to figure out why the DSL device was unwilling to talk to, well, anything - it occurred to me that the new card was the next address up,

I have a couple of questions on the zapbarge: 1) zapbarge asks for a channel - how would a manager know what channel to enter ? Is there any way of being able to enter an extension number instead ? I know that you can get the information from the manager interface, but I wouldn't want to give my users access to this, or have to install / write a system just to get an extension number from a

Anyone know of any GUI's that can be used to manage/setup asterisk?

Hi, THIS IS IN DUBAI. I am having PRI line with 100 DID's (00-99) and when we call to any landline or mobile number then it shows us our board number or pilot number (i.e 4663000 means 00).. As i give all the extensions a particular DID, so people from outside world can call them. The problem is the CALLERID ... When we call from any of other extension PSTN line carries out our pilot number

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional