similar to: Per extension/user CDR?

Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it won't ring. All the other phones ring fine, and my phone wasn't the only one I rebooted with the changed sip.conf and impd.conf. I have reverted back to a standard sip.conf and impd.conf and I still can not get my phone to ring for any incoming

Greetings, Recently there has been talk of the presence/buddy feature with asterisk and Polycom phones. I have it setup, and working as expected, however I can only get 7 buddies to appear on the screen at any given time. Has anyone gotten more than 7 buddies to appear? I'm just trying to find out if this is some polycom limitation, bug, or my error. Thanks, Matt -- Matt Gibson VOIP

I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>

Greetings, I just received some netweb-301 phones frm Seshu down in NJ. I cannot for the life of me get it to register with the asterisk server, nor upgrade the firmware to the latest (1.41) i'm still using 1.37. The packets are traversing the router, going into the other subnet, hitting the asterisk box, but not actually making it to asterisk. Nothing in the asterisk logs, but tcpdump

Received two new Polycom 500 phones. Dhcp and ftp configured properly to load the various files including v2.5.0 bootrom.ld, etc. One of the phones loaded all firmware and config files properly, registers with *, and is usable. The second phone loads bootrom.ld (from the same ftp server on the same wire as the phone), but towards the end of the bootrom.ld load process (about 430 pkts as seen by

I reloaded my asterisk and found some red lines flushing by. When I stopped it I see: WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring signalling WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echocancelwhenbridge WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echotraining Reconfigure channel 1, FXO Kewlstart signalling Reconfigure channel 2, FXO Kewlstart signalling

I want to block following types of numbers in my extensions.conf like the premium number in Taiwan: exten => _90204X.,1,Congestion Since I have a DID in USA, I need to block these numbers in USA, as well all emergency numbers, but still let open free (???) service numbers. Can you help me to compile such a list? bye Ronald

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the *

I had a Grandstream 286 at my home hitting my Asterisk box at the office, all worked well and I received phone calls fine until the device just up and died. I replaced this unit with an SPA-2000 because I have been impressed with the Sipura devices and decided to use them for most of my needs in the future. Problem is that my phone attached to the device rings shortly after power up of the

I saw some of the heat about the $20 bounty earlier. So I don't want to put a low bounty out. Quote me a bounty, and I'll see if I can get it approved by management. :-) I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a

What does "t" mean in a CDR entry? This is in place of where the number that was dialed normally goes. For one IAX termination provider it always has a t instead of the number dialed. Also, we always see the word "hunguup" in the same record entry. This is the provider we have set to our secondary not primary. Is it transfer of some sort? I don't think there was a

Here is the situation: A T1 router going into an office which then plugs into the firewall box then into the switch. None of these devices support QOS.. Is there some sort of box/device that I can place between the T1 router and the firewall box which will allow me to prioritize voice traffic on this link? I can't change the T1 router to something that supports QOS because it has

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

I've seen the embedded posts. Is anyone running Asterisk on the MINI ITX? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Hi, again. I've spent a week trying to get asterisk to work on FreeBSD unix, with some success. Everything works until I plug the box into the TELCO line and then the line goes off-hook and stays that way. So I bit the bullet and decided to install the application on a fresh linux install. Not to start an OS war, here, but linux is ... difficult ... for an old unix hand to get his mind

Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. with help of agi scripts i am able to insert caller id number in database of mysql now i want to pop it up via

We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in

I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like

Hi, I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything is under control, I have one issue with the way CDRs are kept for queues. And I don`t mean ?I don`t like it?. I mean it crashes the server. I realize there are multiple CDRs per queue call ? one per ring/per phone, basically. The issue is that whenever the number of CDRs ?to be recorded? for a call exceeds 5000,

Hi Hekuran I have installed asterisk-1.0-RC2.tar.gz, asterisk-prepaid-0.3.1 and postgresql. When I tried to call from any IAX client to another IAX client and also sip client to sip client it worked fine. And also the cdr table filled properly. Now I tried to configure asterisk-prepaid-0.3.1 with asterisk. I have compiled asterisk-prepaid-0.3.1 and also copy the configure file. I