search for: cisco7960s

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

I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the "Transfer" softkey does not show up - as a matter of fact only 2 softkeys show up (redial & something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are

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

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 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

I am currently testing a new Asterisk installation. The server has a T100P connected to a PRI, and about 50 Polycom IP600 phones connected via the local network. Every couple hours, Asterisk randomly stops responding to all calls, both incoming on the PRI and calls from the SIP phones. I'm not sure how or where to start debugging it. When Asterisk stops working I can still connect to

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -

Hi *, does someone has a directory that works with the Cisco 7960 and astdb or mysql/ldap? Regards, Andreas _________________________________________________________________ Gaming galore at http://xtramsn.co.nz/gaming !

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Shaun Ewing > Sent: Tuesday, November 04, 2003 7:15 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Anyone using * in a live > production environment? > [...] > The additional features are nice too.

...with numbered extensions for demonstration purposes. What is the syntax to associate a extension with SIP phone? Does the Dial application have a SIP specific entry for example: Dial,SIP/SIPphone/s|15 When I call from one extension to another I get "User is on the phone". We also have Cisco7960s to test. Currently Have X-Lite setup. Can log into the server and can execute the demo features. No CO line interface. Just experimenting with IP. SIP.conf configured voicemail.conf configured. extensions.conf configured. sip.conf [user] callerid="User Name"...

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

This might have been talked about before, but I'm posting anyhow. I've got down to testing Asterisk yesterday, and I couldn't help but compare it with Skype (a Windoze only product, yet, but extremely efficient for some reason). Skype has almost unperceptible delay (LAN), while there is almost half a second of delay regardless of the codec on Asterisk. An even if we were to