similar to: Cisco 79xx + asterisk + some functions Q

Hi Everyone.... Got a newbie type question regarding MOH & Cisco phones. I'm still new to Asterisk (very new in fact) & built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 & 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems.

I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me

Hi list, I have a "Wildcard TDM400P REV I Board 1" with 4 FXO modules and * 1.0.9 up-and-running. Only 2 FXO ports are used for 2 analog phones and are doing fine. I now wanted to use the 3rd and 4th port, but when I insert an analog phone, take it off hook, I do not get a dial tone. With my 1st and 2nd port, I get messages like: -- Starting simple switch on 'Zap/13-1'

The documentation of how to use the 79xx series' phones and features with Asterisk is really hard to find and put together. The higher end phones like 7970 are more like converged PC+phone, a thin client to telephony and network apps. But it's really hard to target it as a development and deployment platform because the docs and techniques are so obscure. There seems to be a fair amount

Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

I have a running * with a TDM40B board in it. I have 3 analog phones that works (rings) perfectly when connected to a Telco POTS line. When connected to the Digium TDM40B (with FXS port), I have problems with 'ringing': 1 phone 'ringes' normally 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it does 'ring-ri... ri.... ring... ri...) and the 3rd

Hi, I'am new to * and googled/read a lot, but did not find (yet) a lot of info to do the above. Some months ago, I did find a 'story' from somebody having put * between his PRI and current PBX as IVR, but I can not find it back :-( Any help/pointers are appriciated. Txs alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs

Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is

Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? What caveats are known about using

I have a serious problem that repeats very often, after 30 - 50 calls and I can only solve it by stopped and restarting * :-( After a while, * seems to loose track of something. When an ISDN call from PBX needs to go to the Telco, I get 'Ring requested on unconfigured channel 255/255 span 2' It's always channel 255/255 ??, the 'span' number is random Setup: * between Local

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! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

I am preparing to setup a system using Cisco 7940 and 7960's I have the 7.1 SIP firmware on them. One issue I have run into is how to silence the ringer if a call comes in and you don't want to take it. Many phones have a DND button. I know the 79XX has the DND in the menu but it is to cumbersome to go into the settings then phone preferences then the DND and select yes. Is there any other

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

Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEP<mac>.tlv file. Most of the howtos say something about an empty file but

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

Are the 7.x SIP images for the 7940/7960 compatible with the 7912 or no? I ask because there is a 'special' section for 7912 SIP images that seem to be on a totally different revision train. -- GPG/PGP --> 0xE736BD7E 5144 6A2D 977A 6651 DFBE 1462 E351 88B9 E736 BD7E

Hello all, I feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for "7960 sip firmware" and find that fast). Thanks for the help, Jeb Campbell jebc@c4solutions.net

Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, asterisk-users-request at lists.digium.com wrote: > Date:

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot