similar to: how to detect pickup...

Hello asterisk-users, There are .call files, with their own syntax, ant they works. But I have a problem. The voip-info.org says: "... If the call answers, connect it here ..." that means, if the called people picks up the phone, he/she hear ringing, until the "caller" picks up the phone. But what can I do, to connect the call before it answers, so the when the called

Hello Asterisk, This is only a test, because I can't start new thread in this list... -- Best regards, Gergo mailto:csibra at gmail.com

Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in

Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone. Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone. <-- SIP read from xxx.yyy.142.234:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP

Hello I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine. I am using the OpenVox B200P ISDN card. My problem is that even though chan_misdn module seems to be loaded correctly with Asterisk (I can see it using 'module show' command) the misdn commands are not available to me in the CLI so I cannot tell if my box is correctly interfacing with the ISDN card Any ideas

Good Afternoon, Can someone please explain what Y-cords are available out there and how they can be used with Aastra or other VoIP phones? Maybe with or WITHOUT headsets? Isn't a Y-cord traded for soft Barge in these days? Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've got a device that seems to become unreachable for about 2 minutes, every hour. From what I can tell, it isn't due to network or server issues. Any ideas? TIA. -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

Hi all, With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Regards Kyle -- Kyle Gordon kyle@lodge.glasgownet.com http://lodge.glasgownet.com

Q: What is the most annoying thing in e-mail? Spam and useless replies when I've already asked for this topic to be closed *sigh*. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Gergo Csibra -->> Sent: 05 December 2008 14:41 -->> To: Asterisk Users

Greetings list, We recently installed some Linksys SPA962 + SPA932 sidecars into a client's offices. The BLF functionality works fine, but call pickup using the BLF is something of an issue. Following the advice on voip-info.org, I configured part of their dialplan as follows: exten => _**2XX,1,Pickup(SIP/${EXTEN:2}) exten => _**2XX,n,Dial(SIP/${EXTEN:2},15,tw) exten =>

Hey guys, just thought I'd let you know that I'm giving away a copy of "TrixBox Made Easy" on The Asterisk Blog <http://www.asteriskblog.com>. Check it out. -- www.AsteriskBlog.com Your home for easy to learn Asterisk stuff. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've read alot of comments on the SPA-3000, many if not all saying they had echo issues, but I've not seen anyone comment on the SPA-3102. Does anyone have any comments or issues with these? Tim

Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? Thanks, Jose

Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/1001 at from-sip|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten

Hi, I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement. My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345. Can someone help me to add this dialplan. Thanks in advance Dan -------------- next

Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. Tnx Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.0/639 - Release Date: 18/01/2007 18.47 -------------- next part

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee => #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten => s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able

Hello ! I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18 The things work up to 80%, I can transfer the call by BLF button and I can see the green (free) status and red (busy) status. What I cannot do is to accept the call when someone rings a remote extension. The BLF button starts to blink in red telling me that the call is ringing on remote extenson, but if I press it,

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]