similar to: changing cisco 7940/7960 standard menus ?

Hi, as noone out there seems to be able to maintain chan_sccp, i'm trying to switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly wrong/non functional. I see Redial NewCall CFwdAll more (more) CFwdBu... GPickUp Confrn more NewCall works, CFwdAll seems to toggle DnD, the rest of the buttons do notting. Any ideas how to fix this? Regards, Andreas

PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to

Hello all, I'm in final testing stages and preparing training for a new Asterisk rollout. I'm replacing a Cisco Call Manager system, and re-flashing the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call, I use the "Confrn" softkey to invite other participants. I can add one other participant endpoint into the conference, but no more. I know I can (and

Hi all I'm pretty sure someone must have done this before but I couldnt find any trace of it on the web so I thought I would drop a note about how I ended up doing it. I have also posted this info on voip-info. Warning : This is not very elegant and I'm currently trying to write a patch in order to make it better but so far, this the only way I've gotten this to work. Scenario : I

Hi, I?ve been following some information about the latest chan_sccp drivers for a few weeks. I installed and tried about any variant of driver that exists in the chan_sccp system. Despite a few little changes in the CLI options I am still encountering a lot of problems with the phone. When not using the phone for a while, it "disconnects" from asterisk (1.0.7) after a few hours

Hi, I am new to Asterisk and have some general questions _before_ I start buying equipent to install and get everything up-and-running. (this means I have no running Asterisk (yet)). I have read already a lot of doc, but some things are not clear to me, since I'am inexperienced in Asterisk and PBX. The goal: Make PBX using Asterisk and Cisco 79xx equipment for 25 phones (1 x 7970, 4 x 7960,

Hi, People here often work on 2-3 places (office 1, office 2 and home). I would like to give them 1 extension (XXX) and to ask them to 'register' the phone they use at a certain moment. The idea is that, when you need someone, just dial XXX and the phone near him (in Office 1, Office 2 or at Home), will ring. This will keep my queue system and other tricks intact, where I always use the

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'

Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, Asterisk is running strictly VoIP over the network and using CCM as the trunk. Calls from the SIP phones connected to Asterisk work fine. They can call both external numbers and any Cisco extensions attached to CCM. Calls from CCM to Asterisk fail without any notification in Asterisk (and I DID have this working at one

When I call to VoicemailMain it just sits. ; Retrieve Voice Mail exten => 2500,1,Wait(2) exten => 2500,2,VoicemailMain(s100) exten => 2500,3,Macro(endcall) 1.4.3 latest SVN. voicemail(100) works and the mwi systems works. I am not using ODBC or SQL. Voice mail to email works ok. I just cannot retrieve it by the application. I'm not sure when this quite we get little voice mail

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

Hi, I have just started to setup and configure my asterisk box and am having trouble with it. I have a dlink nic in the box as well as the digium card. When the nic is in there by itself it works, but when I put the tdm400p in, there seems to be some sort of conflict with the network card. The tdm appears as another type of network card, which I don't think it should, should it? It shows

does anyone know howto set the softkeys of an Cisco 7960 running on an asterisk server via chan_sccp ? Mit besten Gr??en Dirk Rieger

hi All I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI. 1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper. 2. I Want To seeting E1 in ASterisk/PC Back To Back To Cisco E1 AS5300 Use ISDN Signaling, my configutration :

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi, I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. Sometimes I receive a normal page, without problems. Sometimes half of a page and the rest is scrambled, but most often, I receive nothing and the other site reports a Fax error... The Fax machine

Does anyone know how to customize the order of the soft keys on a 7960 running SIP? All the documentation I could find is CallManager related. Specifically, I want to move the transfer function to the first set of buttons during a call.

Hi... I've been fighting this for a while now, trying clean builds of Asterisk 1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today. No workee. :-( Here's the results for various calls made off-hook (push the blue Speakerphone button on the Polycom 430): 988852700 - Phone waits for me to either hit the soft-key "Send" or "EndCall". If I hit "Send",

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