similar to: extension mobility with cisco phones

Hi. I'm trying to develop a module that emulates the Cisco Extension Mobility feature from CallManager (the ability to log in to a phone and temporarily acquire the extension, soft key programming, and all other settings for that user profile) with Asterisk 1.4 and Cisco 79xx phones (some with SIP and some with SCCP, as the 7914 extension module does not support SIP). I've almost

I've got a question regarding Cisco IP Phones and licencing. When using a third party PBX like asterisk is a licence required for the Cisco phones ? Has anyone got anything in writing from Cisco to clarify this ? Eg can I just use CP-7961G or do I need CP-7961G-CH1 even though I'm not using Cisco Callmanager ? HYPERLINK

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

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

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

For those of you running Cisco phones, did you start out with a Cisco CallManager and move to Asterisk? And if you did switch do you find that you or your users are missing features they once had? How have you handle the issue? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at

Does anyone have the 7970 work with sccp? I have used the example configurations from the wiki. I can see that the phone is failing to retrieve the CTLSEP<MAC>.tlv file and then is able to retrieve the SEP<MAC>.cnf.xml file. The phone is hanging with the message "updating ctl". Thanks, John SEP<MAC>.cnf.xml: <device> <devicePool>

I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/4c2821af/attachment.htm

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

I intend to add to my asterisk system a feature similar to cisco call manager's extension mobility so that agents can log in to any phone in the office and keep their profile (ex. the agent's specific directory number). But before doing that, I need to confirm that asterisk doesn't have a native solution for that (ex. application/addon), and that nobody has come up with their own

Still looking for cheaper (under $250,-) alternative to cisco 7940 with features needed for corporate use, mainly: - shared phone book (e.g. via LDAP or XML browser in phone) - in-line power - missed/dialed/received numbers - integrated switch (voice VLAN support) I found only aastara/sayson phone (and Intracom/Netphone in the past), that has xml services anounced, but still not available, so

Hi, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho,

Hi, according to discussion on asterisk IRC, where people said, that macros will be depracated, I tried to migrate from macros to contexts and Gosub but if I try to use gosub in extensions.ael, ael compiler complains, that I shouln't use Gosub app, but I can't find ael keyword, that will be Gosub equivalent, or can I ignore this ael warnings? thanks PJ LOG: lev:3 file:pval.c

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

Hi, have you some idea, how to make "roaming line" with Asterisk? i.e. is possible to have phone line assigned to user if migrating from one office to another? thanks PJ

Off topic, but involves an Asterisk deployment in a roundabout way. Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that can tell me where a directory of the standard system voice prompts for Callmanager might be obtained? I am looking for the text and filenames of the standard prompt set that ships with Callmanager, have been all over the Cisco site and I can't find it.

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

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

Hi Siggi, > > 7960 and then "Call Ended" on the Display (curious about that !!!). > > That seems to be normal for the 7920. I've sniffed the registration > procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's > doing the same thing. Maybe that's some odd way of testing if the > CallManager ("CCM") really works... >

Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/ On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > > Next step is packet capture to see if there is a clue as to the cause of > the > > failure in the SIP signalling. > > Right, I