similar to: Cisco7960/SCCP Transfer Help?

Hi All, I have what IMHO is an interesting issue. I'm using Cisco 7940's with the 7.2 SIP load and Asterisk CVS-HEAD-09/10/04-10:11:46. Everything is working great so far, except one small issue. When a user presses the 'Trnsfer' soft-key, dials the other extension, and presses 'Trnsfer' again, before the other party picks up, hold music for the original caller

I am trying to register cisco 7920 to asterisk using sccp since to sip firmware upgrade to it ,but its ends with failed registration.Can you please send me a sample for sccp.conf configuring cisco 7902. Thanks -- SCCP: Accepted connection from 192.168.5.163 -- SCCP: Using ip 192.168.5.228 -- SCCP: Accepted connection from 192.168.5.163 -- SCCP: Using ip 192.168.5.228

Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request

Can someone provide example configuration files and/or instructions for configuring Asterisk to work with chan_sccp2 and a 7960. I have the searched the chan_sccp2 site, the voip_info wiki site, asterisk doc project, and mailing list archives for the past year for help, but failed to find anything helpful. Some of the questions I have are: 1. What settings need to be made on the phone? 2.

Hello All, We are looking for a SIP provider teminating calls in India, Pakistan and Bengladesh. Any one knows a good one? Regards, Cesar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040824/4d5dd4b4/attachment.htm

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with

Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing "55 <phone icon with x>" so it looks like the phone is not registered. The phone and the asterisk are in the same local

I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on

Dear Skinny/SCCP lovers :-) I've just completed & uploaded to the cvs the newest version with fixed redial key AND implementation of speed dials. please test extensively and report any bugs. i know that the display is not yet set correctly but the buttons are working as expected. Enjoy testing... --jan (*1) http://chan-sscp.sf.net (*2) yes, bugtracker is down at the moment, will fix

Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls

I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the

HI all: I have spent a large amount of time configuring/installing phones connected to Asterisk. Halfway through the process I discovered that my Cisco7960 with 2 7914 expansions was not supported in the SIP protocol. After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of configuring SCCP to properly work with Asterisk. So far I have gotten the phone to dial and receive calls

Hi, I have succeed using Asterisk on VMWare on an Athlon@1GB with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem

I have several (various brand) sip devices with static IP's. I understand that asterisk will not accept a registration from these devices if the host= parameter is not set to 'dynamic' in sip.conf. I want calls to these extensions to be routable even before the device registers. I understand that is what defaultip= is supposed to do, but it doesn't work. I get a busy tone when

I have the option of running either SIP or SCCP for my cisco VoIP rollout..can someone shed light on what the pros/cons are? Seems everything is SIP these days so that's the option im leaning. Thanks- Matt

Hi All I'm trying to get my 7970 working on the latest head version of Asterisk without success, perhaps someone could help. I'm using the head version of Asterisk with the chan_sccp from berlios.de dated 7th August. The compilation of chan_sccp seems to have worked without any problem, however I'm pretty sure the problem I have is with the config files in TFTPBOOT:- With the 7970

Dear all I have configure Asterisk 1.4 with sccp and configure Cisco phone 7975G model with Asterisk and it is working fine but i have one problem when i going to confance like when i call to someone and put it on hold and want to take in another New call to new person and want to take both live call in confrance how to do it in asterisk is there any configuration XML

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

Dear is any way to have a secure (encrypted) rtp line between cisco 79XX and asterisk with SCCP? best -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110206/d1a1c1a9/attachment.htm>

Hi all. I compiled the module chan_sccp, now its possible deploy trunk SCCP with Callmanager? Anyone? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120918/881c4e7b/attachment.htm>