similar to: 6.0 image for Cisco 7960's?

I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 => 3213,Bill Smith Thanks! Paul Mahler mail:pmahler@signate.com phone: 650.207.9855 fax: 877.408.0105 -------------- next part -------------- An HTML attachment was

Might be a stupid question, but is there a default gateway set on the 7960? -----Original Message----- From: Paul Mahler [mailto:pmahler@signate.com] Sent: Thursday, December 18, 2003 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT? I have a 7960 running behind a firewall running NAT. From a telnet session to the 7960, I can't ping

Hi, Does * support extension mobility ? If not, is there any way this is possible ? Thanks ! Chris Clifton

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

Hi all, Five minutes ago I heard a rumor that Lennart Poettering has rewritten rsync. My top urgency is to download the latest numbered version of rsync untouched by Lennart Poettering: I'll find out if the rumor is true or false later. I've downloaded rsync rsync-3.1.3.tar.gz and rsync-patches-3.1.3.tar.gz from https://download.samba.org/pub/rsync/ . Is the source code in both those

Does anyone know of any inexpensive alternatives to the four port analog module offered by Digium ($305) what work seamlessly with asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/1644638e/attachment.htm

A Client runs a RAS network. Users access mailboxes using Outlook Express. The server runs dovecot-1.0.3-13_60. Almost a week ago users began to report that they can't retrieve their mail. The session starts, they seem to retrieve messages, but then the client reports a problem and the session is terminated. Here's what I have in the log: May 28 08:56:46 server dovecot: pop3-login:

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm

> Tilghman Lesher wrote: >> On Sunday 18 January 2004 12:01, Adthrawn wrote: >> >>> Spell out RAID - Redundant Array of Inexpensive Disks. Bingo. >> >> >> That's "Independent Disks". It's the independence of each spindle >> that >> is valued, not the cost proposition. If one spindle goes, it's not >> all >>

Paul, I wholly agree with what you're saying - I too ensure that we have at the very minimum, a set of full spares. However, this thread really has the wrong name at this point... We're now looking at embedded solutions, in the same way Cisco has with it's ICS 7750 solution. I'm looking to build a robust embedded solution, that we can run in tandem - of course, we want to

Hi, I am a beginner to Asterisk. Can anybody clear my following doubts regarding the configuration needed? 1) What is the ideal system configuratin required?(like processer, RAM, h/d space etc) 2) How many connections it can handle at a time? 3) How many Virtual PBXs it can handle? 4) Whether Postgres or Mysql is best suited? 5) How many IVR's it can handle simultaneously? 6) How many

Digium X100P / new cards are is available on ebay for $43. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 <http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 > Hope this helps to who want to play with X100P! Are these being sold by Digium ? I don't know ?? - SamW -------------- next part -------------- An HTML

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08

Good morning all, I currently have a website that was written in ASP back in 1999. The system is currently running Windows 2003 Server with MsSQL. Before everyone flames me for being in the wrong place, I was wondering if there is a way to allow centos to run old ASP code? I know years ago this wasn't possible without a program like ChiliASP, but noow I heard rumor that apache might have

Rumor on the IRC channel has a bug being fixed in the g729.so codec that was causing some hangups under certain circumstances. I see that codec_g729b.so in the ftp://ftp.digium.com/pub/asterisk/g729/ directory has a timestamp of today. Can anyone comment on the advisability of upgrading all of my g729-licensed sites? JT

(new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone?

Hi all; I have been searching for an answer to a question that a customer asked me and I have only found a few older answers. So, wanting to find out if anyone has any experience with this issue and can help provide me with some advice. I have a customer which is strongly interested in using Asterisk as a PBX. One of the core requirements, however, is that the system MUST be able to

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen