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Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,

I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: "I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The

Hello, I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones. Every once and a while I have problems with either dropped calls between Asterisk and my provider, or invalid RTP audio streams with phones behind NAT. I have had a few Asterisk developers look into my installation and even my provider check my setup but still am having problems. They tell me that I need to

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a

List, Okay, here's one that has me stumped, and it might just be something simple. Currently, we are setup so that when someone calls in and tries to reach the operator / front desk, it rings several different phones in sequence. (i.e. it rings the front desk for 15 seconds, then a guy down the hall from it for 15 seconds, then my desk for 15 seconds, and as a last resort, my cordless

I have been getting that message also. I have been using various versions of CVS head since Feb. 2005. -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jeremy Gault Sent: Friday, October 14, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am running into some problems here with PrivacyManager. We used to use it without any issue, but now there seems to be several problems. We are currently running Asterisk 1.2.4. First, it seems that if the user does not press the pound (#) key after entering their number, PrivacyManager will fail. I have the minlength set to 10, and

Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)

Hello! Do somebody knows how to compile meetme2 with 1.0.6. I readed wiki, applied patches, but no luck ;-( Me be someone can give me working meetme2.c ? :-)

>The combination of applications CBMysql and MeetMe2 seem to >address our goals. I have MeetMe2 working. CBMysql is >another story, the code looks simple enough and has been >modified to leverage MeetMe2, but * restarts everytime it >tries to launch CBMysql. I cannot find any examples of how >to launch it from the dial plan, nor have I been able to >get any meaningful

Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having

current setup SIP phone 192.168.1.30 --> linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)--> (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address.... or #2 asterisk trying to get back to me as 192.168 on public internet.. got

We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2

I've had 50+ people download the web components, and other than reports of compile issues, I have not heard if this collection has worked for anyone. I do plan to keep updating the * applications and the web pages, but I have almost meet all of our internal requirements and wonder if anyone else is finding it usefull. My focus has been and will likely stay on the user interface, since I have

From: Sylvain Gault <sylvain.gault at gmail.com> I found something suspicious while hunting another bug a while ago. The conditions for that bug to occur seems quite hard to meet, but it's still code quality improvement. See the commit message for details. Sylvain Gault (1): efi: Suspicious size reduction in emalloc efi/main.c | 4 +--- 1 file changed, 1 insertion(+), 3

I have Meetme2 (as well as the web ui) installed and am having some difficulty with the admin features. I've set up two extensions pointing to the same conference, one with the admin flag (1234|Maps) and another with (1234|Mmps). My issues: If the admin presses the * key, it goes to an endless loop of "enter the conf no. followed by pound key" If the user presses the * key, the