similar to: FollowMe dials numbers but can't confirm the call or hear anything

Hi all, sorry if the subject is a bit confusing, but I just couldn?t think of a good way of better describing the situation? Basically, I travel a lot and have several SIM cards for my phone from local carriers. What I?d like to do now is to setup Asterisk, so that people who want to reach me just have to dial one number which forwards the call to all my cellphone numbers in turn. I?m still

In followme , is it be possible to have a third option.... Whereas, takecall=>1 declinecall=>2 proposed option transfercall=>3 or, transferring the call directly from followme isn't really neccessary, if the callee could answer the call, and transfer it someplace, that would work as well.... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody, I am trying to set up followme. As far as I understood from followme.conf multiple digits are allowed to be used for taking call. Currently call is accepted only if one digit is configured for takecall, e,g takecall=>5 Don't you know are there any real limitations for using multi-digit combinations for accepring calls? Thanks, Kseniia

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm

Guys. Ive seen on my asterisk messages log that asterisk has shutdown itself about 12 times in 5 days... The logs show nothing but: [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call [Jun 28 09:40:02] VERBOSE[3172]: [Jun 28 09:40:02] -- Unicall/4 released [Jun 28 09:40:02] VERBOSE[3084]: [Jun 28 09:40:02] Asterisk cleanly

Hi ! We have Cisco 7912 phones, and the doc says that I can "create up to four speed dial buttons on my phone using the Cisco CallManager". Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D

Hi All I was wondering if anyone knew whether there is a built in set of functions for handling "speed dial", basically a set of numbers that can be entered from that handsets and stored in locations such as 3000, 3001 etc. I can do this by adding the extensions to the extentions.conf file and hard coding the numbers but ideally I wouldn't want to have to add them in this way. Any

Hey All, Having a bit of a problem with a Wildcard X100P card. When I try to make an outbound call using the card, it picks the line up and then only dials a single digit. I've confirmed it's only dialling a single digit by listening on a phone plugged into a parallel socket. Incoming calls work fine, it's only outbound calls I'm having a problem with. I've tried

Hi, I have some mySQL-sipfriends and connectivity to PSTN. When a call from PSTN comes, it shows a callerid, and that callerid is displayed at the called sip phone. When the call comes from another sip user (defined as mySQL-sipfriend), no callerid is displayed at the called sip phone. I turned on sip debug and discovered, that in the last case in the SIP-header to the called phone: From:

Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. To ensure that CLI or called number is attached to a database which automatically updates records of the

Hello, This seems to be a weird one. I'm at work now and will get some more-verbose logs later when I get home if nobody has any ideas about what's happening here. I've got a tdm card with 1 FXO and 1 FXS. Asterisk is in the 1.2.x line, so is zaptel. astlinux to be specific. I can get the versions at home later if it might help. It's running on a silent epia 5000 board

Hi gang, I know this is off-topic for Asterisk, but I don't know where else to ask: I've setup a central directory.xml file for my Polycom IP501 phones with a list of all the internal extensions. None of them have <sd>1</sd> as I don't want to enable any speed dials, just have a list in each phone. However, when a phone boots, it seems to pick a random entry and put it

Hi, I've got an important question: I use an E100P directly connected to PSTN, but it does not *really* work as it should be: exten => 1000,1,Dial(Zap/1/1234) BUT: It does NOT dial "1234" but it says in debug mode: -- Called 1/72976451 Apr 26 00:53:00 WARNING[10251]: chan_zap.c:5979 zt_pri_error: PRI: !! Facility message shorter than 14 bytes -- Channel 1, span 1 got

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout

How can I let asterisk immediately dials a trunk when off hook?

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm having a problem with * being very finicky about the length of DTMF key-presses during menus, voicemail, etc. Basically, short (<100 ms) tones are ignored, anything between 100ms (or so) and about 300ms is correctly detected, and anything >300ms is interpreted as multiple presses of the same key. This is terrible for callers who are trying to get to the correct

Hi, Has anybody been successful in running the 7914 expansion unit for the Cisco 7960G IP phone? For anybody unaware of what the expansion unit does, it provides 14 additional buttons, with an LCD display. The idea, is that with an expansion unit (a 7960 can take upto 2 of these units), a user can either assign more speed-dial's, or can monitor line status/account status. So, you can

Hi all, i think my problem is trivial, but i don't get this thing done ... In our office a linux-box ("gw") is connected like this: - via eth0 to the local net (192.168.1.0/24) - via eth1 to the DSL-router (dynamic ip) - via ippp0 (isdn-card) 192.168.2.245 to the stand-alone pc 192.168.2.250 ("single") "single" is a w2k box and have an isdn-card, too and is