[asterisk-users] iaxmodem, chan_capi, hylafax problem and faxing in general

Can anybody point me in the right direction please? I'm having some issues getting iaxmodem and hylafax to talk to each other. I have no doubt that someone has had this type of issue before but I can't find anything useful in the archives or on Google. Under RH9, with chan_capi 7.1, Asterisk 1.2.

Re: [asterisk-users] iaxmodem, chan_capi, hylafax problem and faxing in general

instead of iaxmodem which seems to work wonderfully but I'm having some issues with root verses uucp permissions which is spoiling my fun. Anyway, thanks again! Faris. (please excuse my top posting) -----Original Message- Faris Raouf wrote: >The problem is that when I run faxstat, i

Re: [asterisk-users] Virtual Asterisk Installation

We have been successfully using Asterisk (1.6.0.x) in a heavily loaded Virtuozzo (= commercial OpenVZ) environment for over a year. I'm sure we aren't the only ones to do so. We had some terrible problems with random "one-way audio a few minutes into some calls" to start with, which I was worried

[Asterisk-Users] Problems mixing audio in queues and playing queue positions

Hi folks, Over the weekend I finally decided to upgrade one of our Asterisk systems from 1.0.9 to 1.2.4 I had no significant problems and all is well in general - as usual Asterisk rules! However, I did run into two small issues. Can anyone help me solve them please? The first one involves

[Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis.

Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

t is presented would be 01234567894 Contact me off list if you want any further help. Peter Braidwood -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: 23 February 2006 13:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [Asterisk-Users] mpg123 alternative?

Ah! Now this is actually something I've not been able to get my head around: > Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which > has its own MP3 player. Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I use it ? I still seem to have the usual two

Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

t with e.g. 'g1', because the dialplan doesn't know which interface will be used. Armin On Thu, 23 Feb 2006, Faris Raouf wrote: Thanks for that Peter! I think your message solved my problem: I set the master number to be in group 1 (group=1) in capi.conf and called Dial w

Re: [Asterisk-Users] mpg123 alternative?

ee Archer wrote: Check out the musiconhold.conf.sample in the asterisksource/configs folder. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: 23 February 2006 18:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject

[Asterisk-Users] ISDN BRI and UK Premium Rate Numbers

Can anyone help point me in the right direction please? I'm based in the UK and I want to start using a Premium Rate number with Asterisk - I think the equivalent in the US would be a "900 number". Effectively the caller pays much more to call such a number than a normal national or local call

[Asterisk-Users] GrandStream GSX-2000 strangeness

I have a really baffling problem. A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for use with Asterisk. At first all was well. But recently I've noticed terrible sound quality problems. Basically the sound will "glitch" or stutter randomly from time to time. Now, what i

[Asterisk-Users] RE: GrandStream GSX-2000 strangeness

Thanks to all who replied on this. But amazingly I think I've solved the problem. Basically I did a factory reset (select reset via the Menu key then enter the MAC address [as shown on the white label under the phone], then press Menu key again) and re-entered the necessary config details on both

Re: [Asterisk-Users] ISDN 2e Cards

Simon Faulkner wrote: I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the UK but both seem to have drawbacks/advantages. I need to build a new Asterisk box for my tiny business (1 x ISDN2e from BT and 1 x IAX link from Gradwell) Is anyone prepared to go out on a limb and say

Re: [Asterisk-Users] ISDN 2e Cards

Avi Miller wrote: John Daragon wrote: I'd second that. For a single ISDN2e connection the AVM Fritz card is really hard to beat/ Yeah, "single" is the key word there. I have 2x ISDN2 (OnRamp2 in Australia) and the AVM Fritz cards are a nightmare. Replaced the two cards with an Eicon Diva V-4

Re: [Asterisk-Users] Problems with current chan-capi-cm

Armin Schindler wrote: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Armin Schindler schrieb: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in

Re: [Asterisk-Users] recording queue calls

Dov Bigio wrote: Hi, When I set "monitor-format=wav49" on file queues.conf for a queue, Asterisk records calls at /var/spool/asterisk/monitor. But the file names it users are the call-ids of the calls. Is there a way to change that, and use information such as date, time, agent and queue t

[Asterisk-Users] in and out recorded audio mixing in queues

Way back I was still on Asterisk 1.0.7, I configured my systems to mix the incoming and outgoing audio call recordings into one file per call for both normal calls and queued calls using: exten => _9.,1,Monitor(wav49,${TIMESTAMP}-${CALLERIDNUM}-to-${EXTEN:1}-${UNIQUEID},m) ; m option merges a

Re: [Asterisk-Users] recording queue calls

Tom Lynn wrote: Faris, Is there a way to have * send save these in an off-server location? Or have * e-mail them via smtp and then delete them from the server automatically? I'm sure there is a very technical way of doing it. For example if I remember correctly you can set your own script to

Re: [Asterisk-Users] suddenly iax calls don't work anymore

Gerald Dachs wrote: Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and n

Re: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues

[Asterisk-Users] TDM400P not detecting hangup and not hanging up.

Can anyone suggest where I might begin looking for an answer please? I have just installed a TDM400P (2x FXS and 1x FXO modules installed) The first problem is that it does not seem to be able to detect if the remote party has hung up when a call comes through on the FXO. For example, if someone

RE: [Asterisk-Users] TDM400P not detecting hangup and not hanging up

Canuck15, No, I hadn't played with the gains. But I've now done so and no difference unfortunately. Thanks for the suggestion though. I have discovered that after Asterisk has answered the call and the remote caller has hung up, if I lift the receiver on a phone connected to the line (in parallel

RE: [Asterisk-Users] FW: channel offhook state

Yes indeed. There have been huge changes to chan_zap.c in CVS-HEAD compared to 1.09. In 1.09 Stable there are a lot of problems with handling call hang-ups. CVS-HEAD, of 28/08 was much better. But even though it did improve things, it wasn't quite right. In particular I found two problems with pol

RE: [Asterisk-Users] Voice Prompts, what do you think? Good voice.

Gregory, My advice is to go for it. Allison is nice but there are times when her accent doesn't pass the International test ( e.g. everyone I've ever spoken to in the UK roll about on the floor laughing when they first hear her in the Voicemail prompt, telling you to leave a message ). Others wil

RE: [Asterisk-Users] X100p Problem, randomly hungup pstn line

>I installed this card, everything work, i can make call and receive >call with no echo and great sound quality, but after between 5 to 50 >secs the call disconnect by itself, in the log i don't see nothing >revelant. In logging.conf, try enabling debug logging to the console and/or to /var/log/as

RE: [Asterisk-Users] Hang-up Detect - Yet Again

> * answers the call, but if the incoming caller hangs up, * does not release the line. Is there a polarity reversal on hangup (those clicks you hear maybe)? If so then you may find that using the CVS-HEAD version of Asterisk will help hugely. Put hanguponpolarityswitch=yes in your zapata.conf B

RE: [Asterisk-Users] X100p Problem, randomly hungup pstn line

Right at the end of your Zapata.conf you have: #include zapata_additional.conf hanguponpolarityswitch ;Include genzaptelconf configs #include zapata-auto.conf Remove that hanuponpolarityswitch as you already have hanguponpolarityswitch=yes earlier on, and I don't know what having the second one,

Re: [asterisk-users] Polycom_acd_functions SIP trouble

Dovid Bender wrote: I am sure you prob. know this but in your configs it shows secret commented out. Also it with a softphone if it dosent work then, then its your configs. Also did you remember to reload asterisk ? - Original Message - From: "James Fromm" <[EMAIL PROTECTED]> To: Sent:

Re: [asterisk-users] Polycom_acd_functions SIP trouble

James Fromm wrote: Yeah, we tried that. Tried every combination of variables in sip.conf. Only solution that works is removing the requirement for a secret. Faris Raouf wrote: One thing to try is setting type=peer instead of type=friend. I'm a bit dazed and confused at the moment

Re: [Asterisk-Users] SPA-3000 Disconnect tone detection in Spain, Peru and Colombia ?

makevuy wrote: Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty messag

Re: [Asterisk-Users] DID setup from goiax.com

trixter aka Bret McDanel wrote: I dont know then that was cut and paste from what I have working ... maybe actual log dumps of the error? On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote: That is What I stated in the email.. my GOIAX #. not the DID #. That is not the issue. Is

Re: [Asterisk-Users] Polycom 601 XHTML microbrowser

Chris HARIGA wrote: Gary Reuter wrote: On 10/26/05, *Chris HARIGA* <[EMAIL PROTECTED] > wrote: I have a "show parked calls" php script for my Polycom IP600 phones. If U are interested let know and I can email it. Even if Sean doesn't want it, I do!

Re: [Asterisk-Users] Polycom 601 XHTML microbrowser

[EMAIL PROTECTED] wrote: At 08:38 AM 10/27/2005, you wrote: http://www.voip-info.org/wiki/view/Polycom+XML+browser+scripts+for+asterisk Best regards, Chris HARIGA Thanks. Is it possible for someone to provide a basic explanation of how to implement this for us less technical minded people

Re: [Asterisk-Users] Grandstream GXP-2000

Erick Baum wrote: We're having a rather serious echo problem using the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it every once in a while on

Re: [Asterisk-Users] Time based call direction

Rene Nelson wrote: I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? Thanks Neri Hi Neri, The command GotoIfTime() if your answer here. See http://www.v

Re: [Asterisk-Users] Fritz!Card PCI ver2.0

Stephen Arulraj wrote: Anyone knows how I can use this ISDN card for asterisk as a BRI trunk interface? Thanks, Stephen Hi Stephen, Is this a new version of the AVM card? If not (or even if it is), you may find the following pages helpful: http://www.voip-info.org/wiki/index.php?page=As

Re: [Asterisk-Users] A2Billing Authentication Refused

Sam Tam wrote: Try o reupload the mysql database again to see if that work? Sam *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Omar McKenzie *Sent:* 03 November 2005 00:27 *To:* 'Asterisk Us

Re: [Asterisk-Users] Fritz!Card PCI ver2.0

Patrick wrote: On Wed, 2005-11-02 at 19:33 +, Faris Raouf wrote: Please note, however, that somewhere in the wiki it suggests that you modify the AVM driver code slightly. I found this stopped it compiling, and that simply leaving the code as it is worked fine. Then please add a note to

Re: AW: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs

[EMAIL PROTECTED] wrote: Thanks Armin, this version is working, but I still have an undefined symbol in another module: [pbx_wilcalu.so]Nov 5 18:51:12 WARNING[11348]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Nov 5 18:51:12 WA

[Asterisk-Users] small chan_capi-cm 0.6 capicommand(echosquelch) problem?

I now have chan_capi-cm 0.6 working with Asterisk 1.2 RC2. But I have discovered a small problem. I have a mix of analog and ISDN (BRI) lines coming in to my Asterisk box. Both types of lines are fed into the same set of contexts. In the previous version of chan_capi-cm that I was using (0.53

Re: RE : [Asterisk-Users] In France asterisk never detect hang up. Why ?

[EMAIL PROTECTED] wrote: Hello everybody :-) This are my first line french zapata.conf settings. I have 3 like this, with only rx/tx gain a little bit different levels. Running well. Best Regards, Francois BERGERET, France. usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid

[Asterisk-Users] hints/subscriptions accross IAX

(I hope this isn't html - Thunderbird is so annoying) I'm new to using hints/subscriptions on * so please be patient with me. I have two * systems in different geographic locations, connected via IAX Location1 has a Polycom 600 and a GXP-2000 phone Location 2 has a single GXP-2000. With the l

[Asterisk-Users] Polycom 600 presence indication on *LED*?

I've somehow managed to battle may way through hinting issues with type=peer type=friend and various other oddities and now have presence working correctly on my Polycom 600 and Grandstream GXP-2000 phones. However, on the Polycom I have to press the Buddies softkey in order to see if an exten

Re: [Asterisk-Users] UK experts only. BT Outgoing caller ID not showing

Paul Redstone wrote: Hi guys Thanks for help on this so far. There was no typo - old exchange was System X and new one System Y. Also caller ID is enabled on the new DDI range so we get incoming caller ID. BT are looking at this - the guys I talked to is being very helpful and has referred

Re: [Asterisk-Users] Polycom 600 presence indication on *LED*?

Jerry Jones wrote: Create a contact entry with their extension and enable buddy watch on it It will then show up on an unused line key On May 27, 2006, at 3:26 PM, Faris Raouf wrote: I've somehow managed to battle may way through hinting issues with type=peer type=friend and various

Re: [Asterisk-Users] SIP Presence

Forrest Beck wrote: Does anyone have a working implementation of SIP Presence? I have a new Grandstream GX-2000 phone with the supported hardware and I am not sure how to setup presence with asterisk. I've just been through this myself. It is relatively simple once you manage to figure it

Re: [Asterisk-Users] SIP Presence

Viggiani Domenico wrote: Wonderful explanation! Just a note: So, having done all this, restart asterisk, then reboot your phones (an asterisk restart confuses hints/presence on grandstream phones sometimes) It seems that Asterisk >= 1.2.7 solved this issue. Thank you! I'll try 1.2.7 sho

Re: [Asterisk-Users] Queues Not Reporting Estimated Hold Time

I'm getting the same thing since upgrading from 1.0.x to 1.2.x - no queue hold time announcements. There are other oddities in queues in 1.2.x compared to 1.0.x too. But I'm always afraid to raise them as bugs in case they are not, and 1.0.x was going things the wrong way and 1.2.x is going th

Re: [Asterisk-Users] ISDN BRI and UK Premium Rate Numbers

This is pretty standard Asterisk behaviour exten => ,1,NoOp exten => ,2,Dial(SIP/&SIP/&SIP/) exten => ,3,Hangup The incoming ISDN call will ring the specified SIP phones, and will not be answered until one of them picks up. As simple as that? Thanks!! That's perfect. Faris. __

Re: [Asterisk-Users] Problems if GXP-2000 phones and Asterisk are not on the same network

Mimmus wrote: Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? We have GPX-2000s connecting via different network

Re: [asterisk-users] Correct settings for UK (BT) FXO

Brian Candler wrote: Is there a document somewhere giving the correct TDM400P FXO settings for use on a BT PSTN line in the UK? All I can find is http://www.voip-info.org/wiki/view/UK+Asterisk+Details A patch was written for a previous version of Asterisk that got halfway there. I found some

Re: [asterisk-users] Grandstream SX2000 attended tranfer

magnus wrote: Hi all, could anyone share how to perform attended transfers with Asterisk and Grandstream SX2000's - we are able to perform blind transfers with no problem, but attended transfers fail - is it necessary to set two line identities on the phones to be able to do this? Appreciate all

Re: [asterisk-users] Polycom 601 & Expansion Module: Light the LEDs???

Doug wrote: Hey Folks, Been wrestling with the 601 and the expansion module. Finally figured out how to populate both with speed dial entries. Also "hints" are showing in Asterisk with the "show hints" command. But how do I get the LEDs to light when one of these other extensions is either of

Re: [asterisk-users] Is 1.2.12.1 production ready

Just in case it helps anyone: We had 1.2.12.1 crashing on us on a daily basis, and sometimes several times a day. I found that by disabling all qualify lines in iax.conf and sip.conf the problem went away. Faris. ___ --Bandwidth and Colocation pr

[asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

Do any *UK* users have an SPA3102 (the newer version of the SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call has hung up? I've read everything I can find, including an SPA3000 UK setup PDF that lists UK ring etc tone settings, port impedances, disconnect tone settings an

Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

Conrad Wood wrote: On Tue, 2006-10-24 at 12:05 +0100, Faris Raouf wrote: Do any *UK* users have an SPA3102 (the newer version of the SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call has hung up? I've read everything I can find, including an SPA3000 UK setup PDF

Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

Conrad Wood wrote: It is brand new so I assume the firmware is the latest?: Software Version: 3.2.6(GWa) Hardware Version: 1.1.5. It just doesn't detect real hangups at all. If the person calling hangs up, either before and after the call is answered, the SPA will eventually timeout after abo

Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

ah. Do you have callerid from BT (bt line?). I signed up for something called BT Privacy or so which is free and gives you callerid. If you turn on logging (debug) on the sipura it'll log the received callerid via syslog. Also helpful to check under info "Last seen number" or so. There is CLI

Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

Henry.L.Coleman wrote: Just a thought ... try reversing the Tip and Ring Henry L.Coleman CEO Henry, Apologies for answering the wrong message in my last post. I thought I was answering the one from Conrad. Sorry! By reversing the Tip and Ring you mean physically in the wiring or somewhere

Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

Conrad Wood wrote: I'm not seeing any caller id in the syslog nor the last seen number thing. (which helpfully just says "," :-) I'd be pretty sure that the device doesn't detect the cli. My one does list the number under the 'last seen number thing'. What sort of line is it? Straight BT? telew

Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

Henry.L.Coleman wrote: Yep, just swop the two wires. Sometimes the "Tip and Ring" get reversed and most loop start interfaces don't really care (they work either way). It's worth a try since if the disconnect is a reverse polarity flash then the card may see not see this condition as it is alre

Re: [asterisk-users] Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )

[EMAIL PROTECTED] wrote: We have a problem where callerid works 50% of the time on both lines. What we are seeing in the logs is: Hi Phil, Unfortunately your configuration looks OK to me. Here's mine, which works 100% with CID (but not dratted hangup detection!). There are some duplications

[asterisk-users] 1.6.2.10 sounds Makefile error?

.6.2.10 source directory, all is well again and I can "make install" with no errors. I did a diff on the two Makefiles and there are what appear to be several differences, but I can't put my finger on any obvious errors. Any ideas? Faris Raouf -- __

Re: [asterisk-users] 1.6.2.10 sounds Makefile error?

> > I don't have such a centos 4.8 system handy to test with. > > What version of 'make' do you have? > > make --version > rpm -q make > > In any case, please submit a report to http://issues.asterisk.org/ > Thanks Tzafrir. GNU Make 3.80 Make-3.80-7.EL4 I'll submit a bug report. I just

Re: [asterisk-users] OT: UK PPP certification -- what is it?

They mean PhonePayPlus (formerly ICSTIS). www.phonepayplus.org.uk I am not aware of them certifying particular phone systems. Rather, they impose certain requirements and obligations on the service provider depending on the nature of the service being provided and the number range it is provided o

Re: [asterisk-users] openvz

> On Fri, Sep 03, 2010 at 03:11:39PM +0200, mattias wrote: > > Can i run asterisk on a openvz vps or do i need a kernel? > > I dont use dadi > > I don't expect any problem. > Absolutely right: 1.6.x works fine with OpenVZ and Virtuozzo out of the box as long as you don't need any hardware interfa