similar to: Slow hangup - Australia - analog - incoming calls

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Hi, I have little problem and it is so embracing when u r talking to some one and line get hang-up. When some one calls from out of state or out of country my calls gets randomly hang-up with in few seconds and it happens with most of the calls. It's happening randomly I got few calls, which worked. I have also observer that with quite few call I can't hear the person and person can

Hello, I've commited the new busydetect routine to CVS. You need to cvs update asterisk of course and then choose it in asterisk/Makefile and recompile asterisk. All you X100P users that had the problems with false hangups or the card not being able to detect the busy tone please check that. In the asterisk/Makefile you need to find a line BUSYDETECT = and uncomment what you want/ comment

Thanks for your suggestion Steve. I have done as you advised and set busypattern=300,200 to match the sample I recorded. This hasn't worked though, asterisk doesn't seem to detect the busy signal. Does asterisk require a the signal to be in a certain power range? The signal I get is very quiet. Thanks for your help Regards Jonathan On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s

> Hi Gareth Blades & Doug, > > Thanks so much for for the feedback. I have searched on lot of documents > but couldn't able to find clear answer regarding it. > > I hope you guys replies are very much help all in aterisk community. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Engineer, > > Debug Solutions > > Sri Lanka .

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

Hello, I've got the following configuration: 2 X101Ps Asterisk built with BUSYDETECT_MARTIN busydetect=yes busycount=10 callprogress=yes signalling = fxs_ks With this setup, the best I can do is get voicemail with 17 to 19 seconds of silence tacked on at the end. Ideally, I'd like at most 2-5 seconds. Has anyone had any success with this? It seems that hangups are indeed detected,

Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not

Hi list, I'm wondering if it's possible to transfer a call that is still ringing??? Actually, the problem is that my telco provider doesn't offer an uniform method for answer/disconnection supervision, and by that I mean, some of it's line (I think) offer a polarity reversal, but other lines (of the same service provider) do not offer anything at all, so the answer of a call

Hi ! I have some issues, i don't know exactly if it's a busy detection issue. When i dial into the Asterisk box, and if i hang up before the Asterisk answers with the IVR Welcome message, the Asterisk goes on with the call. But, if i wait for the Asterisk to answer, and if i hang up, the Asterisk hangs up too. I have this parameters on zapata.conf: busydetect=no

thankyou both very much for your swift responses and helpful insight... While my knowledge of administrating Asterisk is fairly decent, i must say my knowledgebase and ability in troubleshooting it is fairly lousy... all these wonderful suggestions you have had about turning this log on here, etc sounds great, but i don't know where to begin on that! how do you recommend i turn these on or

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hello people, this is my first mail to this list. I'm new to asterisk and trying to set up an IVR. So far my dialplan works nice connecting with softphones, but I'm having problems to detect hangups on the analog line. Here are the details: -Clone pc with Ubuntu server 10.04 64 bits -Asterisk 1.6 -Openvox A400P, lspci -v says: 01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX

Is there a way to specify multiple email addresses in voicemail.conf for a specific user? I seem to remember that it was possible, but can't remember the character to separate the email addresses. (I tried '&', but that didn't work...) later, PaulH

I am trying to send 'codes' over an isdn2 link - such as *#24# - to activate call forwarding. But it doesn't work. I have tried sending it as a straight dial, and also as a DTMF string...but no luck... I spoke to a telco tech and he said I had to send a facility code....huh? Anyone with any ideas on this one? PaulH

I need to carry a variable over into the 'h' priority - so I can go back and clean up DB entries in a mysql database (time of call and so on) I tried using UNIQUEID but it seems that 'h' generates a new one. Anyone have any ideas? What can I use to carry a variable over into 'h'?? later, PaulH

When a caller hangs up (e.g. after leaving a voicemail), my British Telecom exchange sends a continuous tone for about 15s and then silence. I can't get asterisk to recognise this tone as a hangup indication. I have tried indications.conf with both country=uk and country=us. My zapata.conf has busydetect=yes, callprogress=yes and I've tried setting busycount from 1 through 7 I am using

Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? I got * server 1.2.5 and running on Centos 4.3, I hock up to two PSTN lines and each time some one call in and my phone delay 1-2 sec (this is Asterisk delay nothing to do with Sangoma) and it rings on my phone, however, end on the day I got not less that 10 empty messages. I found out that Sangoma FXO