similar to: Cisco 79xx Ringtones

Hi All, I recently received my Cisco 7970 and have it up and running with 8.0.4 firmware, with asterisk 1.4. Seems to function pretty great so far, aside from a few issues. Here is what I have noticed so far, anybody have any fixes for these issues? 1. Contrary to the forums and lists I've been reading - the phone indeed does register. The little registration animation stays on the screen

I have a done google seaches on convertion and so far they all failed. Rich adamson and wheely-bin.co.uk Here is what I have Laptop running solarwinds tftp with the following files OS79XX.txt <- POS30201 SIP<phonemac>.cnf.xml RINGLIST.DAT <-Ringer1.pcm Ringer1.pcm The phone (7940) is hardcoded to the TFTP ip of the laptop and has a hardcoded ip is was on firmware version:

Hi All, I am running into a problem setting the ringtones via _ALERT_INFO on the Cisco 7960 phone. I am using * 1.2.1 and have tried setting the variable to several values. I have also tried setting the phone's software to both 7.5 and 8.2 thinking that it might be a version issue, but with no success. I have examined the packets and do see the ALERT_INFO header being sent, but the

I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten => 3010,1,SetVar(ALERT_INFO=<Bellcore-dr1>) ; selects Ringer exten => 3010,2,Dial(SIP/3010,15) I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone

Hello: Does anyone know how to configure any of the Cisco 79XX phones to get custom ringers when using chan_sccp with Asterisk? I've currently got Asterisk's 05-24-04 CVS-HEAD and Zozo's 0.2 release of chan_sccp. I've tried using ringlist.dat, but that appears to only be for the SIP phones... Thanks for any input, Andrew

The about-to-be-released Grandstream firmware now supports multiple ringtones, but (so far) I haven't been able to unearth any documentation as to how one uses them. Anyone out there know anything about this? I've googled, read the firmware READMEs and combed the GS site without any luck. Thx. B.

Hello, We have a DID partner sending traffic to Asterisk via SIP, but we are not hearing ringtones. When we call the same extension via SIP, we can hear that's it"s ringing (virtually).. Is is something related with call-progress not recognized by DID provider ? Thanks, ________________________ a b d o u l aba at gcomnetworks.com SIP: (131) 229-1002 at sip.freeipcall.com

Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file.

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From:

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel).

Hi, I know it's not really the place, but if anybody in the UK (or US) is interested, I'm clearing out lots of new Cisco stock... 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), 7935's (conference phone) and 3550-24-PWR switches. I also have boxes of 7914's, the single-7914 foot stand and double-7914 foot stand (these are required to

> Tilghman Lesher wrote: >> On Sunday 18 January 2004 12:01, Adthrawn wrote: >> >>> Spell out RAID - Redundant Array of Inexpensive Disks. Bingo. >> >> >> That's "Independent Disks". It's the independence of each spindle >> that >> is valued, not the cost proposition. If one spindle goes, it's not >> all >>

Hello Everyone. I usually find my own solutions for problems but this time, after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call, ringtones indicating that the call is being connected play normally for the first 5 seconds to the

If you wan't to create a ringtone with makering.pl for firmware 1.0.50, be sure to create it as ring.bin and then rename it to ring1.bin / ring2.bin or ring3.bin. This seems to be the only change between the format from 1.0.4.68. Regards, Maron

Anyone know what the file format for the cisco 7912 RINGLIST.DAT file is? I have a RINGLIST.DAT file for a 7960 that is working fine. It's just a list of the ring name and the file name but it doesn't work for the 7912. I've read somewhere that the 7912 needs the ringlist.dat file to be in xml but I can't figure out how exactly it is constructed. Thanks. -- -------------- next

Hi, I'm just considering buying two Telecoms grade Sun Netra's to run a lab-based VoIP solution. Not my immediate thoughts as a VoIP platform, but from what I've heard, they can run Linux, and run it well. Only thing is: The Wiki and the Whitepaper just state that Asterisk is for the x86 architecture, but has been compiled to run on PPC architectures. No mention of UltraSparc.

Hi, How can I configure R to allow http connection (through company firewall)? Your help are greatly appreciated! Wei Ding, Ph.D. Schering-Plough Research Institute 2015 Galloping Hill Road Kenilworth, NJ 07033 Tel: (908) 740-2592 Fax: (908) 740-7664 ********************************************************************* This message and any attachments are solely for the intended rec...

Hi, I read in the Asterisk Whitepaper, that you can run two cloned servers, one as a primary, one as a backup, and have them automatically failover to the other unit when it crashes, or when you need to restart it. The primary application of course, would be ensuring calls can be made when frequent updates are being handled, or when an update must be restarted on a busy network. The term

Hi all, I've been modifying the ringtone on a group of Snom phones like this, depending on certain dial-plan conditions: Exten => s,1,SIPAddHeader("Alert-Info: <http://192.168.0.200/tel_ring01.wav>") exten => s,n,Dial(SIP/mjc_office&SIP/mjc_home&SIP/mjc_lab&SIP/mjc_server,20,trj) Now, I'm migrating slowly to Digium D70 phones, which have a different

I have just successfully got the TFTP file remapping to work such that I can have unique ringtone files for each and every extension. I added the following to my server_args line in the xinetd configuration for TFTP: -m /home/asterisk/grandstream/ringmap.cfg Now the entire line reads: server_args = -v -s /home/asterisk/grandstream -u asterisk -m /home/asterisk/grandstream/ringmap.cfg (There