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I don't know if the is possible on not. I would like to know the easiest way to ring all extensions in the sip.conf file for intercoms. I have phone to phone intercom working.

I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong.

I had asterisk 1.0.5 running fine. I upgraded to 1.0.6 and now the music on hold does not work. More Detail: While I was running asterisk 1.0.5, when someone called into an Polycom IP500 and was put on hold via the Polycom "Hold" button, the hold music would play. After upgrading to 1.0.6 that does not work. But if I set up an extension to play the hold music, it plays.

I have spandsp installed and working, but when it emails using Scotts mailfax, the attachment is a dat file. I tried to rename the file to .tiff or .pdf, but it will not open. In the /var/spool/asterisk/fax folder, that faxes are there as tiffs, and I can open those without any trouble. The problem is in the conversion from tiff to pdf. Is there another package that needs to be installed for

I don't know if this is possible, so I will let the collective decide. Here is what I would like to do. BossA calls BossB, BossB's admin assistant sees the call from BossA on her phone. CallerID would look something like: BossA to BossB : on her phone. And she would be able to pick if BossB was not in his office. I am sure this is possible, but I do not know where to start, or even

Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '703XXXXXXX@147.135.8.129' timed out, trying again -- Got SIP response 404 "Not found"

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the

List: I already have asterisks up and running on a PRI, but where we are moving we cannot get a PRI so we are going to get T1. My question is: We are going to us E&M Wink for signaling with DTMF and caller id. The channels are going to be setup like this, 12 channels for 2-way and 12 channels for incoming only with DIDs. How would I configure the zaptel.conf? I realize that I will have two

Has anyone been able to find a way to disable call-waiting on Polycom phones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/b7ea7b5d/attachment.htm

Has anyone on the list tried one of these new cards with built-in echo cancellation? This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are

List, I am having trouble with one of our IP600. Every five days or so, the phone locks up. This is the third 600 I have put in place. I am running asterisk 1.0.9. Has anyone had this problem with the IP600? This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively

Hello Everyone, We've connected OCS to Asterisk via OpenSips, and the voice functionality is working fine. I was wondering if anyone out there who has implemented a similar system would be willing to share any information on how they have implemented this, specifically with regards to URI numbering on the OCS side. So far, we've done this: 1. Extensions in * are 4-digit, and the

List, I just tried to swap out our 410 for a 411 and we started have problems with on of our T1's. Setup: Span 1 - Dedicated PRI for long distance. Span 2 - 12 channels fxs_gs outgoing local. 12 Channels em_w incoming DID's. I didn't have any problems with the PRI. The trouble was with the T1. We were unable to place any local calls, and all incoming DID's where garbled.

I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB of memory. This is serving about 75 sip clients, Polycom500's and 600's. We are running into problems with the memory. Asterisk, right now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7, Zaptel 1.0.7 with Digiums TE410 1xT1 RBS and 1xT1 PRI, Libpri 1.0.7 on Fedora Core 3. My question; is this

We are in the process of testing * for company wide deployment. We are using Polycom 300 phones, the only problem that I am running into is when I call an 800 number that has an IVR I get disconnected after about 60 seconds. Here are the logs from asterisk. I am not sure if this is a problem with asterisk timing out or if it is the phone. To me this looks like asterisk is timing out.

I have googled this and come up empty. Has anyone had any problems compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am getting when I run make. app_nv_faxdetect.c: In function `nv_detectfax_exec': app_nv_faxdetect.c:210: error: structure has no member named `cid' app_nv_faxdetect.c:227: error: structure has no member named `cid' app_nv_faxdetect.c:265: error:

Does anyone know of a AGI script that takes advantage of the weather sound files that's included with the extra sound files available from www.loligo.com/asterisk/sounds/ <http://www.loligo.com/asterisk/sounds/> ? Thank, Jeramie -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this.