similar to: defaultip= in sip.conf doesnt work?

We would like to setup in house SIP phones 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.

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 did some searching in the archive, but found only one message with this same question and no answer. Hopefully it's a simple config problem. When the Caller-ID is delivered, it is surrounded by double-quotes, like this: "ATA-57 1" On long caller-id strings, the last character is cut off to make room for the leading double-quote: "BudgeTone 1234 instead of BudgeTone

Hi there, can anyone shed some light about the use of "host=dynamic" and "defaultip=xxx.xxx.xxx.xxx" in view of iax.conf and sip.conf? In bug 558 I learned that at lest for iax.conf these two settings should NOT be used together. Does the same apply for sip.conf? That would mean that both the Wiki as well as the draft handbook need to be adjusted. Cheers, Philipp

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

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

I have a cisco 7940 with the following sip.conf config: [Desk1.1] type=friend secret=****** defaultip=192.168.1.14 insecure=no mailbox=102 callerid="Desk1.1" qualify=500 canreinvite=no context=extensions host=dynamic group=2 I do not get message waiting indicator (mwi) on this phone. Is the another .conf file invilved in configuring this function other than the mailbox=xxx in the

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

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

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

I have a few Cisco 7905G phones and I having a little trouble configuring them. They are working with Asterisk. I'm able to get the sip image loaded, but I can't get the phones to blind transfer. Does the Cisco 7905G Phone use XML Services? If you are using the 7905G phone, would you post any of your configuration files so I can try and figure out where I'm going wrong? Thanks for

Hi...I've got two Grandstream phones attached to my Asterisk on the same subnet. The phones have fixed IP addresses. Asterisk is generated an error for one of them only, even though both appear to be registered correctly. The current state of the sip.conf is included below. Anyone know what is going on here? Both appear to be working fine between each other and between themselves in and

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 everybody Could someone give a tip on how can I configure asterisk to use 2 ATA's 186 to communicate each other using SIP with asterisk. I know this most be a very simple task, however this is the very first aproach I have to asterisk. I set the following config but I don't get dial-tone when I off-hook the phone from any of the two ATAs. Can some one tell what I'm missing in

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

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

Whe using a particular web page, the default values of the form are set by a java function, then I can modify the web page and submit it. But when I use Mechanize, I am having trouble figuring out how to get those values so I can put them into the form I am going to submit. Below is the function that has the data...and I have changed pertinent info. What other choices might I have to

I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco