similar to: Creating 79xx Configs

Anyone try the new Cisco firmware for the 79xx sip phones? In my test it seems to work fine for a little and than soon the phone looses its time. At first the status shows clear, and then it appears to get confused about the ntp time source and the time goes away on it. No features, just bug fixes. Perhaps I have something wrong with my ntp settings, but the other firmwares work just fine. --

What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very

Does anyone know if the 3rd patch listed on this bug fixes the seg fault problem related to voicemail? http://bugs.digium.com/view.php?id=4800 -- respectfully, Joseph --------------------

Is there a way to get the OnHold music to restart without restarting asterisk? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Does anyone have any experience with an Asterisk compatible softphone application which meets the following criteria: 1) Is able to use touch screen rather than mouse for on-screen functions. 2) Has an API which can be used to export Caller ID info to another App on the same compuer. Thanks Bill

I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another

Ok I've got SCCP running I have my 7970 firmware files. Can anyone send an XMLdefault config and an SEP config file? There are a bunch of sbn files in the package...not sure what needs to be loaded. -Darren

Was in the process of upgrading a 7960 to SIP and in advertently applied a skinny image (P003G302.bin), now no matter what i put in OS79XX.TXT and OS7960.TXT it simply wont upgrade. The phone is pulling down OS79XX.TXT from my TFTP server but then goes on to repetedly ask for SEP<MACaddr>.cnf. Help!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

The chan_sccp page at http://www.voip-info.org/tiki-index.php?page=chan_sccp2 has been updated. See the bottom of the page. Thanks. Comments welcome. -- respectfully, Joseph =============== ---------------------= ********** =

Hi, No one seems to have any issue with the following posting. Can any one suggest how to install/configure channel drivers to work. Basically I am trying to send the SIP calls to GNUGK but Asterisk reports the error "No channel driver found". >>> I was trying to compile the oh323 channel driver but unable to compile the openh323_1_13_5 (which is the only required version as

Does anyone have a more clear beep tone for the voicemail? The default one seems to cut off and give the feeling that there is a problem with the vm :) If no one has this available, I may try to create a new one. Thanks. -- respectfully, Joseph - ------=============

Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Reading the documentation I can see that MCI AVI dll is not implemented except as stubs. Is there any way to play AVI files without installing Windows, say by installing Media Player? Regards, Simon B. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.winehq.org/pipermail/wine-users/attachments/20010604/38066af3/attachment.htm

I am hoping that someone has had better luck with this than I have. I would like to connect Cisco 7940s and 7960s to the Netgear FS116P, and take advantage of its POE. I know that the Cisco phones use Cisco's "pre-standard" POE implementation, but as I understand it the difference is really only two pins. I bought a bunch of PowerDsine inline power adapters which supposedly change

What third party programs are available for parsing the queue_log file and CDR file? I know about XC-AST, but management would prefer a php based solution. What have other admins done to retrieve detailed call information about the queue system? Anyone develop their own that they don't mind sharing? --johann

Has anyone tried to get dialogic ss7 trunking to work with Asterisk? I did some googling but nothing helpful turned up... Could a person get an intel dialogic and get * to see it like a zap channel? And use it for incoming and outgoing trunk access? -- respectfully, Joseph ------=============

Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Cheers, Ben Merrills -------------- next part -------------- An HTML attachment was scrubbed... URL:

The thing I dislike the most about the 79xx phones is that in DHCP mode, they expect the DHCP server to tell them their TFTP server address. They won't let you set it manually. So if I don't have DHCP server that gives TFTP server info, which is most of the DHCP servers at out there, then the phone won't be able to download any updates made to the SIP000*.cnf file. Using dhcpd on

Is it possible to select the audio stream that's played as a user enters a meetme conference? If you could, it would be very simple to record a users name, and then play that as the greeting to other attendees as they join the conference. If not, could someone tell me how hard it would be to modify the source? I presume at the moment the file to be played it stored in a var somewhere,