similar to: extension not found

Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for "dialog-info +xml" notifications with no success. This is what i'm doing: - Phone A has a key configured as type "extension" pointing to Phone B. - In sip.conf I added

Not able to compile asterisk,zaptel,libpri in /usr/src -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100208/fa5ff126/attachment.htm

We're using FreePBX 2.8 and there is a Reports tab but it doesn't seem to actually do anything. Is there some secret/trick to getting a report out of it that will tell us which extensions are placing calls? I've tried every query on the form that I can think of. Is the reporting disabled by default or ??? Any tips/pointers appreciated. Ben M. Schorr Chief Executive Officer

How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? Can packages be used to build the whole system, like using debs and rpms alone? /vfclists

We deployed a single phone handset (Polycom 331) at a remote site. We have a IPSEC VPN running between the firewall at the remote site and the firewall at the site where our Asterisk/FreePBX box lives. We have used a similar configuration for this site before and it worked fine. We gave the phone a static IP address and pointed it to the configuration server on the remote end that has the

Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows "(ULAW)" (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be

Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have

O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium TE205P connected to a single span if ISDN PRI. The Telco has assigned us two local numbers to test incoming calls. I created an inbound route for one of those DID's and assigned it to one of our extensions. Sounds simple enough. Too simple, apparently, when I dial the number the caller gets a recording that it's a

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid:

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

Mandi! Rowland Penny via samba In chel di` si favelave... > > No one have tried to add 'shadow' context in winbind? I'm simply > > curious... ;-) > If you mean adding 'winbind' to the shadow line in /etc/nsswitch.conf, Ahem, no: i meant adding to winbind nss library the support for the 'shadow' context, so i suppose this means adding some sort of

Hi R-users, How to get a bug contex? My R code ran there for several hours, but a bug crashed it, printing such message like "Error: subscript out of bounds". I want to use browser() to catch the bug, but I don't know which loop caused the bug (there are many loops in the code). I even don't know which line of code caused the bug. Is there any utility or something in R which

Hi Guys, I am not so familiar with asterisk and hope to get help here. I am having now some stupid errors. My goal for the first, is to create a simple pbx with different context. As long as I use only the contex 'default' everything seems to work perfect. Now I tried to add another context i.e 'internal' and the asterisk is complaining for not finding the required extension in

Hi, On Tue, Nov 12, 2013 at 01:41:29PM -0800, H. Peter Anvin wrote: > On 11/12/2013 01:22 PM, Andrew J. Schorr wrote: > > I am using 4.05 because that's what's packaged in the latest Fedora > > release (Fedora 19 in my case). I wonder why they are using such an > > old version. I will take a look at how difficult it would be to build > > the latest version on

SIP SECURITY WARNING Version: v1-0 (cvs today) Problem: sip context in general section ignored - goes to default - allowing unauthorized sip devices to place calls in default context Fix [workaround]: Remove or rename "default" context in extensions.conf Notes: I am not sure what other asterisk functionality may be affected by this - review your other config

You have no dial-peer telling the router what to do with the outbound call. http://www.tape.net/~gerry/asterisk/cisco26x0.html At 12:50 PM 9/26/2003, you wrote: >Like Gerry wrote for callerid you need VIC-2FXO-M1 card. > >Right now I am stuck on making outgoing call. > >Could soembody help me with the configuration. > >On cisco I have soemthing like that: > >dial-peer

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: --