similar to: How many lines do you use.

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

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok, so multiple people have said that hinting is possible with chan_sccp on the 7940/7960's and such, has anyone got this working? How do you go about getting this to work? I'd use the wiki, but it's link to the mailing list topic on that doesn't work anymore :( -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

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, Does anyone have any experience with a Cisco 7961 + 7914 Operator Console setup and speeddials/BLF on Asterisk 1.4? Would appreciate feedback if this works reliably. I have a 7961 on skinny registering on an 1.4.19 box with chan_sccp and speeddials work fine so that part seems ok. I have no experience with the BLF part. >From googling around it seems that BLF on the 7914 only works with

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is on a call, other incoming calls will ring through to her phone and can be answered. So far I have only been able to get this working by using the call-waiting function, which is cumbersome and does not properly allow the first call to be retreived. Is there a better way to do this? Thanks in advance. John Mensel

Hi, I was wondering if it is possible to setup with Asterisk a Cisco 7960 to use extension mobility / roaming. Meaning that a user logs into a phone and his profile moves with him / her. I have a network of ~75 Cisco 7960 phones, running SIP 7.5 distributed across 2 asterisk servers in 2 cities. I would like to enable the users to have the capability to move from one phone to another one

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 encountered a semi-lockup situation with chan_sccp and a 7960 when > switching between calls. It's kinda reproduceable - When holding, > resuming, holding, resuming between two calls on a line, one of the > calls become kinda zombie. The phone enters a state where the call is > signalled active but no softkey is there. Already fixed in the CVS Sergio

What is involved in getting SIP firmware into a Cisco 7960 with sccp installed? Expensive image from Cisco? Plated in unobtanium? joe a.

I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday Morning Daylight Savings time occurred. The server shows Mon Mar 10 10:59:42 PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59 10/03/08. How do I update the time display on the telephones please? No virus found in this outgoing message. Checked by AVG. Version: 7.5.518 / Virus Database:

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 everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian

Hey all! Have configured a Cisco 7960 with no problems, put up an TFTP server and it downloaded new sip binaries ? all went well. However, now I am having trouble getting two 7040s to work. Basically, my problem is the above stated error message. If I had any entries in the TFTP logfile, I could probably solve this problem. But the thing is, it just doesn?t look as if the phones are

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

I?ve read various stories about the Asterisk SCCP implementation. Some say it works fine, others say it doesn?t ? stick with a SIP phone! My dilemma is that I?d like a WI-FI phone with multi-line capability, and so far Ciscos 7920 is the only one I?ve found. The disadvantage is that it cannot be upgraded to support the SIP-protocol, so if I purchase such a phone I am stuck with the SCCP.

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