similar to: Cisco 7960 SIP Craziness...

Cisco should really be ashamed of this product... While it is physically well constructed, and has excellent sound quality along with a very pleasant user interface, the device has SERIOUS stability issues, unless you run your network with an iron fist... Quite by accident, while configuring my Asterisk system to connect to a Cisco 7960 via SIP in a standard office PBX type arrangement, I

Hi All! Can anyone direct me to any websites / manufacturers out there who are making small, put-it-in-the-closet-and-forget-it type systems for building routers, home gateway servers, that sort of thing? My fantasy machine for this purpose would be along the lines of a mini-itx system with external power supply, dual Ethernet interfaces on board, and one PCI slot available. If it had one

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

Does G.729 provide better voice quality than GSM? (a question for people who have tried both) --J. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 188 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030715/52f18205/attachment.pgp

Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a "basic" Linux/Asterisk system? Just re-boot and config. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

Hi, I'm looking for a good codec to use on a personal VoIP setup. It is strictly for my personal use, I'll never resell it, make money or it, or whatever. It seems a free personal-use G729 codec is available as a WIN32 library. I find it puzzling that at the same time one has to pay license fees to use it under Linux, even non-commercially. I was wondering -- am I missing something?

Arrrrgh I hate trying to sift through all these messages and keep track of the various threads going on ......... Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers....... only problem being that this list requires list membership before

Does anyone have any insignt on this? Any client programs that could be used? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/0e9e9920/attachment.htm

Hi, I am still toying with the idea of going ahead with using the G.729.. Can those using it tell me about some of your experiences using G.729.. Things like and problems you had running it, the voice quality and anything else you can think of... I have read in the archives that asterisk has to be run with -c.. Is this still the case? and if so does this mean that * can't be run using the

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

Hi I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've searched the mailing list quite extensively, but didn't come up with anything promising (some things wer helpful, though). Does anyone know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310) phones can be made to work with the TDM400P card or if they are ADSI compatible at all? I kind of doubt they will work if they are

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or

According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is this? This should not be a problem for Asterisk? Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN? BR Daniel Nystr?m

G'Day All. So I got the TFTP server all set up -thanks to much help from this list- the 7960 found it and updated to SIP the first firmware P0S30200. What I am now trying to do is upgrate through all the versions, as recommended, to the latest version, P003-07-3-00. I thought this would be accomplished by simply changing the sole line in the OS79XX.TXT file to P0S30203 and reboot the phone.

Hi, I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ? From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2, I got the following: 1. Copy the desired binary image from Cisco.com to the root directory of the TFTP server. 2. Specify the image in the configuration file image parameter for the protocol

Hello all, I recently purchased a Cisco 7940 IP phone to do some testing with (to validate a migration to asterisk for our internal PBX needs). I understand that I need to update the phone for it to support SIP, so I configured the phone with an IP address and pointed it at my tftp server. When I reboot the phone I am currently getting "TFTP File Not Found SEPDefault.cnf" in the

Hi! Can anybody help me with convert Cisco 7940 CallManager Phone to a SIP Phone? I have continious error in tftp log: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests OS79XX.TXT, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080 from=192.168.1.111 Mar

Hi, Can anyone tell me what should be included in h323.conf to get asterisk to talk to a Cisco 2600 gateway? Any statement examples for extensions.conf would also be appreciated. Thanks. Will chan_h323 use the Cisco as a gateway anyway? Regards, Steven Thomas

I have read different posts over the months wondering who Vonage uses for their VoIP technologies. I stumbled across this article (although it's from 2002, I think) that suggests strongly that they use Cisco. There is no telling what they might use in conjunction with this but this should clear some of the conjecture.