similar to: Cisco 7960 sccp

Hello @List, i'm using a Cisco 7970 / 7914 phone with Asterisk 1.4 & sccp part of my sccp.conf type = 7914 (Cisco 7970 with 7914 phone extension) autologin = 117 description = Test speeddial = 10,Test (10),10@mycompany my speeddial line is for internal call howto call a external number eg. +xx467584933 ?? i want to press a button and call directly out thank you greetings, Daniel

Hi all, Does someone know if it's possible to use hint function with skinny ? Can anyone send me an example ? Thanks in advance, Alex.

Any one knows where to install chan_sccp for asterisk 1.4 ???????. Please guide me from where can I download the asterisk 1.4 sccp channel driver and how to install it because I tried to get chan_sccp-mayday05.tar.gz When I trying to install it ,error happened like this. Please help me how to solve this issue. [root@XPL t]# cd chan_sccp [root@XPL chan_sccp]# make clean rm -rf

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

I'm setting up a 7921 and now want to add a second line to the phone. In my SCCP.conf file I have: autologin = 235,299 However, on reloading SCCP the phone fails to login to the second line with this error: [Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register: SEP001BD457F8B1: Failed to autolog into 299: Max available lines phone limit reached 299 Is there a

Getting "no service" display on aastra 480i. Sip debug shows an "unathorized" blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out.

Hello, Here is a breakdown of the issue I am experiencing. I have three remote employees, in various states, who have Polycom 501 phones. They are unable to receive incoming calls after a few minutes of the phones being plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not

Hello guys, The so expected 2.0 release of Idefisk 2.0 softphone is a fact. Idefisk and Zoiper became one - Zoiper 2.06. Here are some of the features: SIP and IAX, TCP, TLS support, Multi-language support, Automatic provisioning (XML), URL handling, Outlook Integration, Native conferencing, API, Changeable number of lines.... You could read the complete Press Release here:

After upgrading from 1.2.25 to 1.2.26 i noticed that IAX -> IAX calls always result in Asterisk just exiting without any message. Asterisk also seems to die when using a TDM400 with 2 FXO modules, placing 2 outgoing calls on both lines as Zap/g2 and then trying to make a 3rd call. Went back to 1.2.25 for the moment

What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: Final-Recipient: rfc822; asterisk-users@lists.digium.com Action: failed Status: 5.0.0 Diagnostic-Code: X-Postfix; mail forwarding loop for asterisk-users@lists.digium.com

Bristuff has been updated; http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz -- Vidar

Hello List, Question: Has anyone been able to create multiple queue_log files in /var/log/asterisk for multiple queues? We are designing a multi-tenant system and separating the log files would be useful, instead of dropping all queue actions into one file. Is it possible this is a user configurable option I am missing? Cheers, -- ------ Christopher T Aloi ------

Hi, Reading from www.voip-info.org, i can see that "Junghann's chan_capi is now part of bristuff <http://www.voip-info.org/wiki/view/Bristuff> , as of version 0.3.0-pre". What does that really mean ? Shall I understand I can share a Junghanns QuadBRI board between 2 CAPI-enabled software (like a 0.3.0-pre bristuffed Asterisk for instance) ? If positive, what are the pros and

Any potential testers eager to build imap storage support using proper Debian packages: Resonably up-to-date packages of c-client (uw-imap) 2004/2006 are by now only availble from experimental: http://packages.debian.org/experimental/mail/uw-imapd On my test Etch system I simply downloaded the sources of those packages nd rebuilt them (debuild). IIRC with that installed the configure script had

Hi all, Since i cant get latet beta of zaptel installed on the latest test version of Debian with kernel 2.6.17-2-686 can someone who is using debian give me some tips on how to get it working and installed? Many thanks, Christian

Other than rebooting the server or restarting Asterisk from cron does anyone know how to kill a meetme room at midnight. Or perhaps other creative ways people deal with callers who don't hang up. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061123/7429572e/attachment.htm

Greetings folks. I seem to be having a problem where calls made from an IAX device (three single-line phones attached to IAXys) do not play the ring tone when calling out. There's a dial tone when I pick up the phone, and the call goes through just fine, it just doesn't ring. All my SIP phones ring normally, however. Is there an option I need to enable that I'm missing?