search for: rining

Hi, I have an asterisk installation that has been happily working in production for some time (E100P and UK BT ISDN30). Recently I upgraded to HEAD-07/29/04. Now, incoming callers don't hear ringing while calling in. As far as I can tell, my config files haven't changed from what was working before. Can anyone please help before my boss shoots me? JC zaptel.conf

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

...gure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming phon from mobile ( i hangup my cell phone ) still my sip phone rining not disconnect notification reached to my sip phone so what is the problem and one more thing some time i got cross talk on phone on Zap channel so is it timeing problme of card or any...

Dear all I have asterisk with TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme or FXO signalling problme i have configuraed singalling=fxs_ks ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org...

...us to integrate our ISDN house telephone system with VOIP. Preferably I would like to run a setup, so that our internal ISDN phones on an S bus are not aware that * is sitting in between. With the configuration below I run into the following problems: 1. On outbound calls, I get the normal rining call progress tone althought the the other party has not even been reached. This then changes from normal ringing suddenly to busy when the other party is sending a busy signal. I'd rather have the call progress send a busy signal right away. 2. Internal calls between to ISDN phones on the...

...ect: Re: [Asterisk-Users] Dialplan to Pick up calls that are ringing onother extensions? On Thu, 2004-10-07 at 13:42 -0400, James Freire wrote: > Hi All, > > I have a qestion that I am sure is not too out there. I would like to > setup my phones so that if I have an extension that is rining I can > call that extension thats ringing from another phone and have it > connect to the phone that I am on. This would help if someone is not > picking up their phone and I cant get to that phone to pick it up. > Anybody encounter a dialplan that would do this? > > Is there any...

I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Answer...

Hi, Here is the configuration of the server that I currently have extension 100 (SIP) =>(SIP)asterisk server 1.8.18(IAX trunk) <===>(IAX trunk)asterisk server 1.4.32(SIP) ===> SIP Providers The issue is while dialing out from extension 100(sip) if the providers sends back 180 Rining the SIP extension(100) won't hear the ringback tone, where as if the providers send 183 session in progress extension(100) will hear the ring back tone. I tested registering to the main gateway server (by passing iax trunk) and it plays ring back tone every time for 180 Rining and for 183...

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat /etc/asterisk/queues.conf | grep -v ';' [general] persistentmembers = yes autofill = y...

...alled SIP/103 -- SIP/103-09142868 is ringing -- SIP/101-090fb6a0 is ringing -- Stopped music on hold on SIP/74.63.41.218-b6036ae0 == Spawn extension (trunkinbound, 8888651085, 3) exited non-zero on 'SIP/74.63.41.218-b6036ae0' other erro is when i call to my tollfree number is rining 2 extencion the 101 and 103 -- Bayardo S?nchez Garc?a Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanchez at gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bj...

...signalling=fxo_ks channel=1 context=local-access signalling=fxs_ks channel=4 ; When I try to dial outside from a SIP phone I get the following error Nov 3 11:14:12 NOTICE[3337]: Unable to create channel of type 'Zap' When I try to dial in, it appears as if the line is answered, as the rining stops, but ther eis only static, and no indication in the asterisk logs that it has received an event from FXO. I have searched the asterisk mailing list archives and posted questions, but to no avail. Can you provide any insight? Cheers Cian

...cator light, which is implied to work as a voice mail indicator. Is this true? What extra setup is needed? Or do I just need to add `mailbox` to the sip config entry and be done with it? -Do these phones still have issues where one side hears the conversation while the other hears nothing or rining? Do they still need special sip file entries ( 2 per phone )? I apprecaite any and all note/advise/rants/raves/information that the asterisk crowd can volunteer about this phone. Thank you. Sean

...n they stop. If he doesnt choose a option and lets those phones dial time out then they stop ringing but whats weird is that queue doesnt go and ring the next priority. I dont want it to ring the next priority but i find it weird that the queue knows that sombody has the call, but doesnt stop rining those other extensions. Now here's another way i found to do this, rather than using dial with a gosub i found that i can put the gosub as part of the queue() command. [ example: Queue(mainqueue,,,,300,,,screencallee) ]. This will run that gosub when the member pics up and it DOES stop r...

I have a TDM400P with 2 FXO cards and I'm using Asterisk@Home 2.8 I noticed that when I place a call to the analog lines from outside, Asterisk takes a while to actually ring the extension the call is being sen to. I've been doing some tests, calling from my cellphone and here is what I see... - After the first ring on my cell, Asterisk logs to the CLI that is has an incoming call -

Hello, I'm using a dial command to make several phones ring. I use this format : Dial(SIP/4029&SIP/4030,15,tTr) As soon as one of the phone is picked up, all the others should just stop rining. But the fact is that they continue to ring for several seconds (4-5s), and this is quite annoying as all phones are in the same room. Phones are Siemens Gigaset C450IP and Linksys SPA921. Do you have any idea what I could do to fix this ? Regards, Yves.

...is dialing fast but without # wait for few number is there any configuration for dialplan I have setup asterisk with avaya system i have 5 avaya system on 5 location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when i press 1627 then it is wait for 5 second and then rining start alternative press '#' what is the method to break this space of waiting after dialing my extention.conf ;North Delhi NOC Extention exten => _16XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _16XX,2,Playback(vm-nobodyavail) exten => _16XX,102,Playback(all-allbusy) ;Mumbai NOC...

...xcept when I get onto voicemail. Then, I hear nothing. It doesn't matter whether I dial 333 (my voicemail extension) and enter VoiceMailMain or get bounced to voicemail on busy/no answer: I just don't get to hear anything. If I ring and extension and let it go over to voicemail, I hear the rining and then when Asterisk picks up the call and diverts to VM, I get nothing again. At first, I thought this might be a firewall issue but then took the phone to an untrusted network allowing it talk without any firewalls or NATs in between directly to the Asterisk machine. Version: Asterisk CVS-HEA...

is it possible to do with asterisk something like this how ? maybe some copy of extensions and some other file ..pls. call to asterisk server -> user 100 is busy -> recorded msg say "all lines are busy , pls wait" ,"you are second caller in 'queue', pls wait" -> caller is on hold till user 100 busy then ring user 100 (ext100) another call to asterisk

...es, just bearly noticeable) - but we can still hear the ringing sound from the calling phone until the timeout of Dial() has reached, i.e. about 20 seconds later. A trace via etheral reveals the following: 7.37s Asterisk-----invite----->GS01 7.48s GS01-----trying----->Asterisk 7.49s GS01-----rining----->Asterisk 7.51s GS01-----OK----->Asterisk 7.52s GS01-----487 request cancelled----->Asterisk 7.53s Asterisk----->ACK----->GS01 I have no idea why the 487 request cancel appeared here. Does that mean there's something wrong with the GS or the network in the remote site? I hav...

Hi all, i have just finished my installation of wine as normal user (see below). Notepad is running fine, so i decided to run something more fun, starcraft. The installation program works, even with sound and all, exept when i decide to run the previews. The previews run rine, but whithout sound. When i return to the installation program the sound is gone. But anyway, it does it's yob,