similar to: Busy Extensions.

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I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetwork at gmail.com if you can help. -------------- next part --------------

Is there anyway to expire password for windows clients with a Samba (in my case recent alpha version)? I found a post the TNG parameter "password expire time = 10" but a "strings smbd | grep expire" didn't turn in up in the binary. Anyone have any ideas if this is possble (I see it posted frequently to the archives, but no answers). Systems and Network Administrator

I'm trying to find the right Digium card for the Dell 2950 Dell 2850 HP DL380 G3 HP DL360 G3 Are these 3.3v or 5.0v machines ? I am out of the office, and need to buy a card today. I am looking at either the TE407 or TE412, and would appreciate any help. :) Julian

Hi all!, i search for some information about to setup my asterisk box with e-mail notification when a I call the voicemail application. Voicemail application works fine in the Dial Plan but nothing happens with email notification ...so what i need to know about this?...wiki pages did not help me ....thanks! G. ----- Original Message ----- From: <asterisk-users-request@lists.digium.com>

Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. I'm sure someone has done this already. Anyone want

Well guys here comes the fun part. I have a Microsoft access (VBA) application that interfaces with my SQL database. This app pulls of info from the SQL record and than picks up the phone and dials that locations number. I have purchased a few hundred NpaNxx's for my own use. I want get into too much detail there but no worries this is legal. I need to change my CID info on the fly. So I am

I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

hi list, I'm running a newly installed asterisk@home an i registered two soft phone. both soft phone are registered 8901/8901 x.x.x.x D 255.255.255.255 50710 Unmonitored 8900/8900 y.y.y.y D 255.255.255.255 6281 Unmonitored but when I call one another, they are always busy and directed to its voicemail Sorry, if this was posted before TIA

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

How to change status of peers "Unmonitored" to monitored? Home users showing "Unmonitored" some display timing. Name/Username Host Mask Port Status zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored clinic_server (null) (D) 255.255.255.255 0 Unmonitored voip

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

Hello, I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know how can i monitor the extension status? when i wrote sip show peers on asterisk Extension Domain port Status 111/111 (Unspecified) D 0 Unmonitored 1300/1300 192.168.50.111 D 5060 Unmonitored 222/222

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices. What are people using? STUN? SER? Thanks in advance! -blake -------------- next part -------------- An HTML attachment was