similar to: Spam? Re: Cisco 7970 running SIP question

Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question Aaron Yes it

Group I have a Cisco 7970 Running the newest SIP image. I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC When I get a call the callerid number show something like 5555555@192.168.1.2 I thought I seen somewhere what that was but I'm unable to find the correct wording when searching Google to find that post again. Can anyone help me out here. How can I remove the asterisk

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Anyone have the 7970 xml config for sip yet? Aaron

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone successfully gotten an SCCP cisco phone to register to two different asterisk servers at the same time for redundancy? I can't seem to get the phone to recognize the second server in the callmanagergroup... Aaron -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I'm getting Number but when I look at the CDR database. I do see the name -----Original Message----- From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you

Hello; I am facing a problem to let Cisco IP Phone 7942G register on Asterisk. The firmware has been downloaded from the TFTP successfully and currently I am running this load SIP42.9-3-1SR2-1S* I feel that there is a problem in the SEPMAC.cnf.xml but really I do not know which one to be used exactly. Basically, there is some effect that appears on the Phone (for example, it is appearing the

I don't see that anywhere. Here is my zapata.conf This is only happing on my 7970 all other phone are working without trouble. [channels] context=pri signalling=pri_cpe switchtype=dms100 group=1 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=incoming musiconhold=default echocancel=yes

Hi. I'm seeing some issues where Entourage clients can see new mail, download it (or partially), but won't display the mail in the mailbox list. Mail shows up as having arrived, but never displays. These clients have worked in the past with versions of 0.99, but not in 1.0-test-56. I have dumped the connection traffic and it looks like Entourage aborts the IMAP session before it

Hey all, Has anyone had any issues with phones having multiple lines that are in different contexts? We've got a couple phones that we're testing intercom functionality for, and I'm noticing that for some strange reason, no matter what line we use, the phones tend to be completely in one context or another, not segregated like I would expect. Our contexts look like this: context

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

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 -

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

I'm currently running a setup that's worked great with MySQL in the past, but we're migrating to ODBC for when we actually integrate the system into our current legacy/voip network. The database I'm trying to use is PostgreSQL, and I've got it working great for Realtime, and if I use cdr_pgsql for the cdr records. However, if I try to use cdr_odbc against the same

Would anyone know whether a jabber module would be in development for asterisk? What I'm looking for is something like the SER module that's out there, with capabilities to send SMS messages from jabber to a phone connected to the system. Aaron Daniel SHSU Computer Services amdtech@shsu.edu

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....." Timeout="0"/> >

In article <Pine.LNX.4.64.0603270059460.8782@systems.shsu.edu>, amdtech@shsu.edu says... > Absolutely right :) > > "\" escapes the next character, so if you wants *69 to go through > immediately, you'd put "\*69" so that the * gets recognized as a digit. > > "," returns the dialtone sound. When my users hit "9", they like to

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk