similar to: Anyone willing to share an Australian dialplan.xml file for Cisco phones?

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>

I have found a phone that I wish I had not! This is by far the worst phone to setup. I have finally upgraded it to Sip but once this got done it I am not able to get it unlocked so I can enter the rest of the settings. So if anyone out there can tell me how to setup my DNS server to tell it where the tftp is located (Windows 2000 DNS server) or please let me have the default password. I called

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Odd problem here--I just got a couple of Cisco 7960s from Ebay that are not functioning as expected...... These 7960s can't seem to be unlocked for manual configuration via any mechanism that I can find. If you go to settings, there is no option 9 (unlock). Available options stop at 4 (Status). **# has no effect. The Phones report that thier current firmware version is 3.1 MF.G2.

Friends, http://www.voip-info.org/wiki/view/Cisco+7905%252F7912+IP+Phones states under "SIP Software Limitations" that XML is not supported on this phone. However, my tech data summary of this product states: "In addition, XML applications deliver impressive applications and network data to the Cisco IP Phone 7912G display." Can someone please clear this up for me?

For anyone interested, our company released a PHP/MySQL based content manager for the Cisco 79XX series IP Phones compatible with the SIP load yesterday. It's available via: http://www.sourceforge.net/projects/open79xxdir Best wishes, -Corey ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

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 everybody, just tried to install my cisco 7960 - but without much success :-( I want to set it up as a sip phone - but I can not setup the phone's IP address... after plugging it in it says "Configuring IP" - I unlocked it and entered the "Network Configuration". I can see the edit-buttons but when I trie to press then it says "That key is not active here"

I have been beating my head against the wall trying to get my ATA186 to send through the *8 (call pickup) sequence back to Asterisk. The Administrator's Guide from Cisco would indicate that the first element in the default dialplan *St4- would mean that any sequence of digits following a * would be sent out after a 4-second timeout. But if I hit *8 it just sits there forever. If I hit

Hi all, I have a Cisco ATA186 connected to an Asterisk Server (SIP) The dialplan uses 1XX for local extensions and XXXXXXX for external numbers, where the first digit is always different than 1. In this moment, when I dial 123 for example, ATA waits till timeout before dialing that number. The same for the longer one. How can I do to make it dial imediately when 3 digits starting with 1 are

Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get

I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Mikel

I've noticed that a few people here are from Australia. I'm wondering where you all get your hardware from. I'll probably order some of the X100P cards from Digium soon (and possibly some FXO cards), but for other things such as ATAs, etc. However - these obviously aren't ACA compliant. Does anybody know of compatible hardware (for POTS lines) that is ACA compliant? The cheapest

- How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is <line button="4"> <featureID>9</featureID> ... For speeddial is <line button="5"> <featureID>2</featureID> <featureLabel>341</featureLabel> <speedDialNumber>341</speedDialNumber> </line>

Hi, An Cisco 7960 ipphone has been set to SCCP firmware by one of our students. I want to set it to 7.5 SIP firmware and I've been unsuccessful yet. Firmware versions are SCCP 3.0 (Source: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1045789) ie: Application Load P003F300 Boot Load ID PC030300 When I browse, phone settings, I see

I saw a previous post about this problem but no one seems to have a real solution for it... I have a few Cisco IP Phones 7960 that I want to install at remote locations. I can easily setup my router to forward the ports of Asterisk BUT I don't want to open my TFTP server over the Internet as all the SIP configs are in clear text... I was first of all considering making VPN connection

Anyone has a working CARD.XML for cisco MGCP phones? The one on the cisco site is old and it's not working with the new firmwares. Thanks Nico -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060505/f7b965bb/attachment.htm

Does anyone have a small, plain services.xml file for a cisco ip phone, preferably one that will work on a 7960? I can't seem to get my xml right, and no matter what I send to the phone I keep getting parse errors. Thanks Shawn

I posted a week ago to see if anyone could share a good, or great, dialplan for a small business who wants to make good use of some polycom 601s. I received a pointer to some the loligo dialplan, which is nice, but doesn't help me with the polycoms. And nothing else, except a number of "please share if you find one" requests. Is nobody willing to share? Thanks- Tom

Hi to all, I'm using asterisk 1.4.9 with chan_h323. When someone in the H323-VoIP cloud dial 1234 this number is assigned to my asterisk-machine, so the VoiceGW forward the flow to my machine, asterisk though the dialplan can delivery the call to a particular SIP phone...this is ok... I can also dial from my sip phone every phone in the H323-VoIP cloud like siemens....BUT...when I call to a