Re: [asterisk-users] fxotune: Unable to set impedance

Hi On Thu, Jul 10, 2008 at 01:45:02AM +0200, Udo Schacht-Wiegand wrote: > Hi, > > we have a strong echo on an Astribank running on > Asterisk 1.2.29-BRIstuffed-0.3.0-PRE-1y-s > The echo is on the opposite site of the analogue phones > connected to the Astribank. So this is an FXS module. > > I

Re: [asterisk-users] READ application

On Wednesday 09 July 2008 22:18:58 John Millican wrote: > Tilghman Lesher wrote: > > On Wednesday 09 July 2008 09:08:50 John Millican wrote: > >> Can anybody tell me what I am doing wrong or why the Read application > >> does not accept the # key as input? My read statement: > >> exten => s,n,Read

Re: [asterisk-users] Performance issues

On Wednesday 09 July 2008 19:15:03 Steve Edwards wrote: > On Wed, 9 Jul 2008, Tilghman Lesher wrote: > > On Wednesday 09 July 2008 16:50:09 Steve Totaro wrote: > >> On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher < > >> > >> [EMAIL PROTECTED]> wrote: > >>> DeadAGI is not recommended and is not supp

Re: [asterisk-users] READ application

Tilghman Lesher wrote: > On Wednesday 09 July 2008 09:08:50 John Millican wrote: >> Can anybody tell me what I am doing wrong or why the Read application >> does not accept the # key as input? My read statement: >> exten => s,n,Read(uchoice|thankyouforcalling|3||1|1); >> >> In the prompt thankyouf

Re: [asterisk-users] Click to Dial Service Providers in Australia

Hi Paul, yes you could build it but the client I have in mind really just wants a preformatted solution eg, someone has already set it up and just gives them a cut and paste job to make it work on their website. Cheers, Dean -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED]

Re: [asterisk-users] Click to Dial Service Providers in Australia

H...you could build a site that used a voip service provided to hook into people's mobiles or home phones That way there's very little software development needed, and it 'just works'. You could deliver the full 'net phone' stuff as phase 2. later, PaulH Dean Collins wrote: > >

Re: [asterisk-users] Dial function exit, go to line n+1

Jared Smith wrote: On Wed, 2008-07-09 at 13:42 +0200, Jerome Poggi wrote: I use them before some patch. But this example work : exten => s,5,ChanIsAvail(SIP/604,s) exten => s,6,Dial(SIP/604,15,wotr) exten => s,7,NoOp(Nopnopnopnopnop) exten => s,10,NoOp(Matthieu) and this not : exten =

Re: [asterisk-users] change E1 link from ISDN to Q.SIG

Klaus Darilion wrote: > Hi! > > I want to test Asterisk<-->Siemens HiCom integration using Q.SIG instead > of ISDN. I did not find any documentation about Asterisk und Q.SIG. > Thus, I wonder is it sufficient to set "switchtype" from "euroisdn" to > "qsig" or are there any other things which I h

Re: [asterisk-users] Performance issues

On Wed, 9 Jul 2008, Tilghman Lesher wrote: > On Wednesday 09 July 2008 16:50:09 Steve Totaro wrote: >> On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher < >> [EMAIL PROTECTED]> wrote: >>> DeadAGI is not recommended and is not supported for channels which are >>> not already hungup (and invoked fro

Re: [asterisk-users] Performance issues

On Wed, 9 Jul 2008, Steve Totaro wrote: > On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher < > [EMAIL PROTECTED]> wrote: >> On Wednesday 09 July 2008 14:06:56 Thameem Ansari wrote: >>> There is no performance impact if you use AGI or DeadAGI. There is no significant difference (in Asterisk 1.2)

[asterisk-users] fxotune: Unable to set impedance

Hi, we have a strong echo on an Astribank running on Asterisk 1.2.29-BRIstuffed-0.3.0-PRE-1y-s The echo is on the opposite site of the analogue phones connected to the Astribank. I tried to use fxotune. It is a production system, so I cannot shutdown Asterisk for long, I did: CLI> zap destroy ch

Re: [asterisk-users] Performance issues

On Wednesday 09 July 2008 16:50:09 Steve Totaro wrote: > On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher < > > [EMAIL PROTECTED]> wrote: > > On Wednesday 09 July 2008 14:06:56 Thameem Ansari wrote: > > > There is no performance impact if you use AGI or DeadAGI. > > > > There is a performance impact

[asterisk-users] changing inbuilt sound messages

Hi all, I am wanting to change the sound files from the standard ones to a New Zealand voice pack. I have copied the files into the /var/lib/asterisk/sounds directory and chowned them to asterisk:asterisk and chmod 420 to match the existing files but the system is still using the original files

Re: [asterisk-users] asterisk 1.4.21.1 seg fault

Jerry Geis wrote: > I am seeing the following seg fault when using a SIP connection > to Console/Dsp. It takes quite a long time to happen but it eventually > happens. > nothing else is on this box. just alsa and asterisk running sip and > console/dsp. > > What should I do now? > > Jerry > > Prog

Re: [asterisk-users] cell phone hangup not getting recognised by system

Yes this solution worked well. Thanks [EMAIL PROTECTED] wrote: > I used to run into this when I first started palying around with some T1 > cards in asterisk. I fixed it with the busydetect and busycount options as > already mentioned. > Try adding: > > busydetect=yes > busycount=5 > > to your z

[asterisk-users] Simple Call Screener

I'm trying to build a simple accept/reject screening app for inbound calls that * forwards to my cell phone. Basically I want * to announce the caller ID and then let me press 1 to accept the call or 2 to reject the call and send the outside party to voicemail. I've been messing around with va

Re: [asterisk-users] cell phone hangup not getting recognised by system

I used to run into this when I first started palying around with some T1 cards in asterisk. I fixed it with the busydetect and busycount options as already mentioned. Try adding: busydetect=yes busycount=5 to your zapata.conf for the group thats having trouble. When the cellphone hangs up aster

Re: [asterisk-users] Performance issues

On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher < [EMAIL PROTECTED]> wrote: > On Wednesday 09 July 2008 14:06:56 Thameem Ansari wrote: > > There is no performance impact if you use AGI or DeadAGI. > > There is a performance impact, in terms of the time it takes for > the process to start up. It m

Re: [asterisk-users] Proper Hangup message

It looks like it's 19: http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause Nhadie wrote: > Hi, > > How do i send proper message when hanging up? > > [from-trunk] > exten => _1234,1,Dial(SIP/${EXTEN}|30|t) > exten => _1234,n,Hangup > > With that, the other end receives a "call

Re: [asterisk-users] Performance issues

On Wednesday 09 July 2008 14:06:56 Thameem Ansari wrote: > There is no performance impact if you use AGI or DeadAGI. There is a performance impact, in terms of the time it takes for the process to start up. It may be measured in fractions of a second, but there certainly is a performance penalty.

Re: [asterisk-users] cell phone hangup not getting recognised by system

Hi Giorgio, Thanks for the suggestion. It works a treat. Kate Giorgio Incantalupo wrote: Hi Kate, have you tried the busydetect parameter in zapata.conf? Take a look here for other useful parameters: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf Giorgio. Lists wrote: Hi al

[asterisk-users] ** app_swift v1.2.2 released for Asterisk 1.2.x code-base **

2008-07-09 - app_swift v1.2.2 released for Asterisk 1.2.x code-base --- Added support for handling multiple dtmf input Added support for input timeout and max input digits (similar to AGI's get_data) Ignores DTMF if no timeout and max digits args

Re: [asterisk-users] cell phone hangup not getting recognised by system

Hi Matt, It is only cell phone calls. We have two POTS lines coming in. I am using TDM400 digium card. This is my zapta.conf. Looks like i'm using signalling fxs_ks is this right? Thanks for your help. Kate ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language

Re: [asterisk-users] cell phone hangup not getting recognised by system

is it only cell phone calls that don't work? or is it any external call coming in over your lines? What type of inbound lines do you have? I;m guessing analog lines... if thats the case what type of signalling are you using? if its only cell calls and not all external calls then I have no idea

Re: [asterisk-users] Zap Bridged Channels

Then see if you can set the speed of your fax to something very low and work up from there until you get to the fastest reliable speed. On Wed, Jul 9, 2008 at 3:44 PM, Jeremy Mann <[EMAIL PROTECTED]> wrote: > I set it up in general because my voice lines(ports 1-4) had very low > volume, and cal

Re: [asterisk-users] Zap Bridged Channels

I set it up in general because my voice lines(ports 1-4) had very low volume, and callers complained about outgoing as well, upping both to two seemed to resolve them. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, July 09, 2008 2:39 PM To: Asterisk

Re: [asterisk-users] Zap Bridged Channels

On Wed, Jul 9, 2008 at 3:28 PM, Jeremy Mann <[EMAIL PROTECTED]> wrote: > I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS > for modem connectivity. > > > > I have Zap/8 as a Fax Machine > > > > Zap/5 is my outside line. When a call rings in on Zap/5 it immediately > calls

[asterisk-users] asterisk 1.2.21.1 seg fault

I am seeing the following seg fault when using a SIP connection to Console/Dsp. It takes quite a long time to happen but it eventually happens. nothing else is on this box. just alsa and asterisk running sip and console/dsp. What should I do now? Jerry Program received signal SIGSEGV, Segmenta

[asterisk-users] Zap Bridged Channels

I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS for modem connectivity. I have Zap/8 as a Fax Machine Zap/5 is my outside line. When a call rings in on Zap/5 it immediately calls Zap/8 and bridges the channels. I see it doing a native bridge on the two. I have echo

Re: [asterisk-users] 2 AVM ISDN Fritzcards

Hi, >> Has anyone managed to get 2 AVM ISDN Fritzcard's working in with >> a 2.6 kernel system? > Yes, with Suse 10.2/10.3 and chan_misdn. are there any cheap (< 100 EUR) PCI-express ISDN-cards out that are supported by Asterisk 1.4.x? I would like to replace my SFF Pentium III with two PCI HFC-I

[asterisk-users] e911/CAMA/MF

Does anyone have any experience getting inbound ANI information from a CAMA/MF/E&M Wink trunk on Asterisk? Is this only do-able with a PRI interface? Any information would be helpful. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.c

Re: [asterisk-users] Performance issues

There is no performance impact if you use AGI or DeadAGI. The only difference is, if you use AGI it will not continue executing the dialplan if the calling party hangsup the call. DeadAgi, will continue executing the dialplan and its upto the applications responsibility to hangup the channel. So, t

[asterisk-users] ** app_swift v1.6.2 released for Asterisk 1.6.x code-base **

2008-07-08 - app_swift v1.6.2 released for Asterisk 1.6.x code-base --- Added support for handling multiple dtmf input Added support for input timeout and max input digits (similar to AGI's get_data) Ignores DTMF if no timeout and max digits args

Re: [asterisk-users] Zap Bridged Calls do not continue dialplan

i just discovered something that would work. exten = 205,1,Dial(Zap/g1/205,,gr) the g and r option make it work the way it should. i dont quite know why r would help, but it did! On Wed, Jul 9, 2008 at 10:22 AM, Kevin Leinenweaver <[EMAIL PROTECTED]> wrote: > Shouldnt it be able to detect the ri

Re: [asterisk-users] Zap Bridged Calls do not continue dialplan

Shouldnt it be able to detect the ringing ringer and a busy tone and differentiate? i'd think this would be a feature that would be implemented. but atleast with the option g i saw that i says it was answered when the line first picked up. On Wed, Jul 9, 2008 at 9:59 AM, Tilghman Lesher <[EMAIL P

Re: [asterisk-users] music on hold realtime

Hi additional question on music on hold. scenario: both using x-lite two extensions 101 and 102 101 calls 102 102 click onhold button 101 hears music (the music is what 101 uploaded) 102 offhold call 101 press onhold button 102 hears music (the music is what 102 uploaded) call put offhold call h

Re: [asterisk-users] OT: DNS security

On second note after reading CERT it looks like thats exactly what it is. Another case where the media is over dramatizing something. On Wed, Jul 9, 2008 at 1:17 PM, C F <[EMAIL PROTECTED]> wrote: > I don't think that this is the exploit that they are talking about. > What you say is too simple an

Re: [asterisk-users] OT: DNS security

I don't think that this is the exploit that they are talking about. What you say is too simple and requires too much to achieve (do it the right time when a request is asked and quicker than the intended DNS server). On Wed, Jul 9, 2008 at 12:01 PM, Alexander Lopez <[EMAIL PROTECTED]> wrote: > Sni

Re: [asterisk-users] READ application

On Wednesday 09 July 2008 09:08:50 John Millican wrote: > Can anybody tell me what I am doing wrong or why the Read application > does not accept the # key as input? My read statement: > exten => s,n,Read(uchoice|thankyouforcalling|3||1|1); > > In the prompt thankyouforcalling it says press pound

Re: [asterisk-users] Zap Bridged Calls do not continue dialplan

On Wednesday 09 July 2008 08:24:25 Kevin Leinenweaver wrote: > Hello all! > I'm having problem with the calls that come through my asterisk box > and back out to our legacy pbx, it seems to be that even if the call > is ringing and not picked up yet, zap reports the line as answered, > why is it do

Re: [asterisk-users] Dial function exit, go to line n+1

On Wednesday 09 July 2008 09:03:07 Jared Smith wrote: > On Wed, 2008-07-09 at 13:42 +0200, Jerome Poggi wrote: > > I use them before some patch. But this example work : > > > > exten => s,5,ChanIsAvail(SIP/604,s) > > exten => s,6,Dial(SIP/604,15,wotr) > > exten => s,7,NoOp(Nopnopnopnopnop) > >

Re: [asterisk-users] Zap Bridged Calls do not continue dialplan

Well, i tried to see a noop with: exten = 205,1,Dial(Zap/g1/205) exten = 205,2,NoOp(After Dial ${DIALSTATUS}) exten = 205,102,VoiceMail exten = 205,103,NoOp(After VM ${DIALSTATUS}) to see i got any dial status changes, i got nothing. Let me explain my path and mabye it might help. A User calls i

Re: [asterisk-users] Proper Hangup message

Nhadie wrote: > Hi, > > How do i send proper message when hanging up? > > [from-trunk] > exten => _1234,1,Dial(SIP/${EXTEN}|30|t) > exten => _1234,n,Hangup > > Ask the telco what cause code they expect: asterisk*CLI> core show application hangup -= Info about application 'Hangup' =

Re: [asterisk-users] OT: DNS security

Snip On Wed, Jul 9, 2008 at 10:50 AM, C F <[EMAIL PROTECTED]> wrote: Very interesting article. I guess we won't know much more for another few weeks: http://www.breitbart.com/article.php?id=080709124916.zxdxcmkx&show_artic le=1 I thought this was common knowledge. I remember hearing about the

[asterisk-users] question about fxo cards

Hi, has anyone worked with nxtvox(www.nxtvox.com) fxo cards? What is their quality? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.n

Re: [asterisk-users] transfers only work when voicemail enabled

>> Hi all, >> >> when enabling blind and attended transfers in features.conf, these only seem >> to work when I enable voicemail for a particular user. How can this be? Can >> I >> have transferrring without voicemail? >> >> Using Asterisk 1.4 by the way. >> >> >> Thank you! >> >> >> Bart

Re: [asterisk-users] OT: DNS security

C F wrote: > Very interesting article. I guess we won't know much more for another few > weeks: > http://www.breitbart.com/article.php?id=080709124916.zxdxcmkx&show_article=1 Interesting! I just went there and the Check your DNS link failed. Anyone else? Rod -- > > _

[asterisk-users] disable DTMF on a particular channel

Hi to all is it possibile (via AMI or dialplan) to disable the DTMF tone on a particular channel? Thanks in advance -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___

Re: [asterisk-users] asterisk sip problem

I don't see anything obvious right away other than have you confirmed that the phone is actually working? Can you get it to ring? With my Sipura adapters that use Linksys software I can view the call status in the "Info" section which if you have that panel might tell you if the adapter think

Re: [asterisk-users] OT: DNS security

On Wed, Jul 9, 2008 at 10:50 AM, C F <[EMAIL PROTECTED]> wrote: > Very interesting article. I guess we won't know much more for another few > weeks: > > http://www.breitbart.com/article.php?id=080709124916.zxdxcmkx&show_article=1 > > I thought this was common knowledge. I remember hearing about t

[asterisk-users] OT: DNS security

Very interesting article. I guess we won't know much more for another few weeks: http://www.breitbart.com/article.php?id=080709124916.zxdxcmkx&show_article=1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - Septe

Re: [asterisk-users] The S word: Asterisk security

Tzafrir Cohen wrote: > On Tue, Jul 08, 2008 at 09:34:44PM -0700, Trevor Peirce wrote: > >> I was recently introduced to fail2ban. It's a nice tool that will watch >> log files and when it notices too many failed authentication attempts >> (SSH, FTP, Password protected web sites, asterisk) it wi

Re: [asterisk-users] READ application

On Wed, 2008-07-09 at 10:08 -0400, John Millican wrote: > Is it because Read exits with a # terminated string Absolutely. The Read() application stops reading digits when the user enters a #. -- Jared Smith Training Manager Digium, Inc. ___ -- Ban

Re: [asterisk-users] transfers only work when voicemail enabled

Bart Coninckx wrote: > Hi all, > > when enabling blind and attended transfers in features.conf, these only seem > to work when I enable voicemail for a particular user. How can this be? Can I > have transferrring without voicemail? > > Using Asterisk 1.4 by the way. > > > Thank you! > > > B

Re: [asterisk-users] asterisk sip problem

They are on the same lan the adapter is registered sip show peers Name/username HostDyn Nat ACL Port Status sippyskypeuser/sippyskype 192.168.2.765070 OK (1 ms) 1000/1000 192.168.2.76 D 5061 OK (1 ms) freephonie-o

Re: [asterisk-users] Zap Bridged Calls do not continue dialplan

> Hello all! Hello! > I'm having problem with the calls that come through my asterisk box > and back out to our legacy pbx, it seems to be that even if the call > is ringing and not picked up yet, zap reports the line as answered, > why is it doing that? could be that the PBX *answers* the line

[asterisk-users] READ application

Hello, Asterisk version 1.4.21.1 Can anybody tell me what I am doing wrong or why the Read application does not accept the # key as input? My read statement: exten => s,n,Read(uchoice|thankyouforcalling|3||1|1); In the prompt thankyouforcalling it says press pound for a company directory along

Re: [asterisk-users] Dial function exit, go to line n+1

On Wed, 2008-07-09 at 13:42 +0200, Jerome Poggi wrote: > I use them before some patch. But this example work : > > exten => s,5,ChanIsAvail(SIP/604,s) > exten => s,6,Dial(SIP/604,15,wotr) > exten => s,7,NoOp(Nopnopnopnopnop) > exten => s,10,NoOp(Matthieu) > > and this not : > > exten => s,5

Re: [asterisk-users] Asterisk dimensioning

Conrad Wood wrote: > Unless I am mistaken and there *is* some way to run 400 simultaneous > calls over 2 PRIs... Traditionally, there hasn't been. But now that they've got that Large Hadron Collider going... :-) -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel:

Re: [asterisk-users] asterisk sip problem

Are asterisk and the phone on the same lan? I see you have nat=no. Do you see the phone adapter registered? Emmanuel Favre-Nicolin wrote: > Hi, > > I'm having a problem to receive inbound call from my sip provider. I used to > be OK, I may I have change something (for example I switched from a

[asterisk-users] Zap Bridged Calls do not continue dialplan

Hello all! I'm having problem with the calls that come through my asterisk box and back out to our legacy pbx, it seems to be that even if the call is ringing and not picked up yet, zap reports the line as answered, why is it doing that? Also it wont stay connected for the correct amount of rings,

[asterisk-users] Proper Hangup message

Hi, How do i send proper message when hanging up? [from-trunk] exten => _1234,1,Dial(SIP/${EXTEN}|30|t) exten => _1234,n,Hangup With that, the other end receives a "call reject" if i don't answer the phone, but the telco said they need something like "No Answer" instead of "Call Reject

[asterisk-users] asterisk sip problem

Hi, I'm having a problem to receive inbound call from my sip provider. I used to be OK, I may I have change something (for example I switched from asterisk 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a configuration problem on my side!) I have basically a sip accou

Re: [asterisk-users] Dial function exit, go to line n+1

On Mon, 07 Jul 2008, Matt Riddell wrote: > > Example : > > > > exten => s,5,ChanIsAvail(SIP/604,s) > > exten => s,6,Dial(SIP/604,15,wotr) > > exten => s,106,NoOp(Matthieu) > > exten => s,n,ChanIsAvail(SIP/605,s) > > > > Won't work because Dial exit to 7, and line 7 don't exist > > > > but > > >

[asterisk-users] cron jopb

Hy, i have a big problem with asterisk instance. Sometimes my instace fall down on zerro and i must manually enter asterisk in console. Can someone eyplain me how to make cron job and make conditions if asterisk fails? Thx ___ -- Bandwidth and Colocation

[asterisk-users] H.323 <-dtmf->

Hi All, would Asterisk 'transcode' H.245 alphanumeric DTMFs to an H.245 signal / rfc2833 H.323 device over G.729 codec ? Thanks for supporting, .TF ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22

Re: [asterisk-users] Asterisk dimensioning

voip crazy wrote: > Hello all, > > I need to install asterisk for 900 sip users with 2 PRI ports. > > Is this correct? 60 channels (assuming an E1 connection, not a T1) between 900 extensions means only 1 in 15 people can be on the phone at once - which is a pretty low ratio. If this is ind

[asterisk-users] change E1 link from ISDN to Q.SIG

Hi! I want to test Asterisk<-->Siemens HiCom integration using Q.SIG instead of ISDN. I did not find any documentation about Asterisk und Q.SIG. Thus, I wonder is it sufficient to set "switchtype" from "euroisdn" to "qsig" or are there any other things which I have to take care of? Thanks Klau

Re: [asterisk-users] Asterisk dimensioning

On Wed, 2008-07-09 at 10:17 +0200, voip crazy wrote: > Maybe 400 calls at one time. By the momento there aren`t voip trunks > maybe in the future. > [snip] > > > > I need to install asterisk for 900 sip users with 2 PRI ports. > > It is posible to handle this number of calls/extensions with only

[asterisk-users] Default table layout for cdr logging with Mysql

Hi guys, I've been looking for a table layout that I should be using for cdr logging to a Mysql database. Everything I find seems to be a little different. What should I be starting with before I start adding custom fields in the future? Also note I want to import some Master.csv files in to this d

[asterisk-users] Simple call accept test

Hi all, I want to write a script to test my asterisk server at predefined intervals. The script will make a call to asterisk server. For that the extension is defined which will answer the call wait for a second and then hangup. If asterisk server receives the call successfully then it will re-ex

Re: [asterisk-users] Asterisk dimensioning

Maybe 400 calls at one time. By the momento there aren`t voip trunks maybe in the future. About cluster, Which cluster solution will could be good option? Which solution could I use to do load balancing between two asterisk machines? Thanks again. Voipcrazy 2008/7/9 Tom Moore <[EMAIL PROTECTED

Re: [asterisk-users] DTMF on iax channel is not interpreted by asterisk

On Tuesday 08 July 2008, Florian Hackenberger wrote: > On Tuesday 08 July 2008, Matt Riddell wrote: > > Maybe the feature digit timeout? The problem turned out to be related to several bugs in iaxclient with alsa support. The softphone was unable to open the audio device and that seemed to cause

[asterisk-users] transfers only work when voicemail enabled

Hi all, when enabling blind and attended transfers in features.conf, these only seem to work when I enable voicemail for a particular user. How can this be? Can I have transferrring without voicemail? Using Asterisk 1.4 by the way. Thank you! Bart __

Re: [asterisk-users] Distinctive Ring for SIP?

> > Does anyone have experience with setting distinctive ring in SIP in such > a way? > I have done this for internal call with ;;;grandstream;;; exten => _12X,1,Set(_ALERT_INFO=\;info=internal) exten => _12X,2,Dial(SIP/${EXTEN},30,tTr) http://www.grandstream.com/asteriskfaq

Re: [asterisk-users] Asterisk dimensioning

How many calls do you expect to be going at one time? Do you have any sip trunks for the users to call out on? Unless this ratio really works for you I'm not sure a 15 to 1 ratio works for most people. I wouldn't just depend on a single server for this purpose. I'll leave it to the cluster guys to

Re: [asterisk-users] Newbie Dialplan: Best Practice in usingContext - Do not use Default??

> With an ISDN10/20/30/etc, I would just put all the lines into an > 'incoming' context - and make sure that incoming context doesn't have > any includes (unless you really need them...) Can someone please have a look at below to see if this would be the best and secure practice of using context i

Re: [asterisk-users] The S word: Asterisk security

On Tue, Jul 08, 2008 at 09:34:44PM -0700, Trevor Peirce wrote: > Steve Totaro wrote: > > For security, how about an authentication retry setting in the sip > > configuration? After X amounts of failed auth or registration > > attempts, block IP for Y amount of time. It would seem fairly easy to >

Re: [asterisk-users] cell phone hangup not getting recognised by system

Hi Kate, have you tried the busydetect parameter in zapata.conf? Take a look here for other useful parameters: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf Giorgio. Lists wrote: > Hi all, > > When I do a test call into the box (which is running latest version of > Trixbox) it all

[asterisk-users] Asterisk dimensioning

Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser . Is it necesary run a SER server on thi