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...failed: (lock_type == F_RDLCK || ibox->mbox_lock_type != F_RDLCK) dovecot: Dec 07 09:21:44 Error: child 16735 (imap) killed with signal 6 **************************************************************************** tcpdump snippets: 0.99 server with Entourage client: 14:43:17.997889 IP noctis.shsu.edu.61359 > ab1-1-41.shsu.edu.imap: P 52:128(76) ack 156 win 65535 <nop,nop,timestamp 519041503 335641251> 0x0000: 4500 0080 adf9 4000 4006 384b 9e87 0bfc E..... at .@.8K.... 0x0010: 9e87 0b29 efaf 008f a87e ee7c fc30 de0e ...).....~.|.0.. 0x0020: 8018 ffff c71...

...e host=dynamic nat=no qualify=60000 secret=secret type=peer regexten=44198 [0004F2100526_2] canreinvite=no context=intercom host=dynamic nat=no qualify=60000 secret=secret type=peer regexten=44198 A sip debug from one of the intercoms: <-- SIP read from 10.20.136.130:5060: INVITE sip:4000@tcm1.shsu.edu:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.20.136.130;branch=z9hG4bK2a5fd1d91B78BACE From: "Aaron Daniel" <sip:0004F2100526_2@tcm1.shsu.edu>;tag=DDF0722-FFF8D457 To: <sip:4000@tcm1.shsu.edu;user=phone> CSeq: 1 INVITE Call-ID: dd12cb03-99065278-efdfa12d@10.20.136.130 Conta...

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

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 - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <Pine.LNX.4.64.0604131916030.26388@systems.shsu.edu> > Content-Type: TEXT/PLAIN; chars...

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

...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

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.

...ully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

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

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

In article <Pine.LNX.4.64.0603211635320.7043@ab1-1-246.shsu.edu>, amdtech@shsu.edu says... > You have to set up a dialplan.xml file in your tftpboot directory for the > phone to pull: > > <DIALTEMPLATE> > <TEMPLATE MATCH="9,59....." Timeout="0"/> > <TEMPLATE MATCH="9,29....." Time...

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

...-bridge "argdesc.l", line 21: unrecognized %option: bison-locations make[1]: *** [argdesc_lex.c] Error 1 make[1]: Leaving directory `/usr/local/src/asterisk-svn/asterisk/pbx' make: *** [subdirs] Error 1 -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

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>

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,

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

I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic mailbox=company1@vmusers

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