similar to: [OT] Polycom provisioning

hi there I'm setting up asterisk@home and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some

From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x10000 error Any Ideas? 1005195711|so |4|00|---------- Initial log entry ---------- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry

I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns

It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. <http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html>

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

Hi, I'm trying to configure a Polycom IP Phone SoundPoint 500 to connect it to my Asterisk PBX but with no success. First of all, I downloaded the SoundPoint IP SIP Administration guide I found on internet and then I tried to make a boot server creating an FTP account on my Mandrake 9.1 Linux box but I needed the following files: 000000000000.cfg sip.cfg phone1.cfg ipmid.cfg sip.ld so I

Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no

Hello all, I first want to thank everyone for all your contributions. I've building an asterisk system for a month or so now and without everyone in the online asterisk community I wouldn't have made it this far yet. Thanks! ...ok, mushiness out of the way.. :) I am looking for a failover and ultimately a load balancing asterisk solution. I've done a good bit of research and I

Anyone know if a way to have voicemail files stored as mp3's? Thanks, Doug.

Has anyone successfully gotten two separate servers to handle voicemail on an NFS shared mount? The main thing I'm concerned about at this point is keeping both systems from writing the voicemail file to the same filename... any thoughts? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I too had a server room fry and need to replace h/w. So what specific Dell servers did/do you deploy? Where is the link w/Digium/s Dell caveats? I'm using the Digium TDM400 card w/* > Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT) > From: Aaron Daniel <amdtech@shsu.edu> > Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! > To: Asterisk Users Mailing List -

Does anyone know if asterisk can detect and handle if a phone is forwarded in the dialplan? Aaron

Arrgh. I just made a call with Asterisk to extension 2944093. That extension exists in astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database query... SELECT * FROM ast_sip_users WHERE name = '2944093' Uhm... Why? Doug

Ugh. We thought we'd fixed some problems by using regexten and DUNDi. Guess not. We have a configuration with three Asterisk boxes. Phones register with a single, primary asterisk box under normal conditions. For voicemail deposit, retrieval, we trunk the calls over to our asterisk voicemail server. However, the voicemail server now has no knowledge of the location details of the phones,

Figured this was worth passing on... This was reported due to the proxy IP address showing up in CallerID on the phone. -----Original Message----- Sent: Thursday, March 23, 2006 12:01 PM Tim, I have tracked down the source of the change in the SIP firmware. The behavior was changed as a fix to bug id CSCsc22406 (host part of the callerid not preserved in ReceivedCall entry). This was a

Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr

I know I compiled AEL2 into 1.2 before, considering I just copied my source from one server to another, yet I can't seem to figure out why I'm getting this error. Anyone have any ideas? make[1]: Entering directory `/usr/local/src/asterisk-svn/asterisk/pbx' flex argdesc.l "argdesc.l", line 19: unrecognized %option: reentrant "argdesc.l", line 20: unrecognized

Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

Hello Guys, I just got my new Cisco 7940* IP Phone Unfortunately I can't find out how to setup this phone. Am I really that stupid ;-) ? I've read the Cisco Manuel, but everything I tried did not help anything. 1. When I turn on the phone it will display "Configuring VLAN....Configuring IP".. This message will not disappear. 2. I can see that the phone has a local IP. I can

Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA.