Re: [asterisk-users] Asterisk not transcoding between installed codecs

iax.conf doesn't take canreinvite=no . It's only for sip . For iax2 its transfer=yes/no (1.4 ) or notransfer=yes (1.2) , there's one more parameter in 1.4 with which u can transfer only audio stream . Check voip-info wiki for all options . On Thu, Mar 13, 2008 at 7:48 AM, Gonzalo Servat <[EMAIL PR

[asterisk-users] Asterisk re-invites and billing

I am using asterisk 1.4.18 (server A ) and have it store records in mysql database . One of my client uses predictive dialer ( asterisk 1.2.26 based and server B ) which makes many calls . B registers with A over sip and there is no nat involved If i re-invite rtp from server B to my carrier ( s

Re: [asterisk-users] G723 on asterisk 1.4.1

That's strange , i am able to see the *url* in Martin's reply . On Sun, Mar 23, 2008 at 6:14 PM, Eric Wieling <[EMAIL PROTECTED]> wrote: > The only messages I have EVER seen Digium remove from the mailing list > archives are discussions about this unlicensed codec. > > > Martin wrote: > > Down

Re: [asterisk-users] g729 encoder/decoder

When g729 phone calls another g729 phone and you are not recording calls or doing meetme with them then license is not required ... g729 phone calling g711 will require a license to transcode the g729 side ( no license for g711 side of call ) . In short anytime u need to convert g729 into some oth

Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4

asterisk -rx "module load codec_g729.so" or "module load codec_g729" and shhow translation recalc On Nov 27, 2007 2:36 AM, Fernando Berretta <[EMAIL PROTECTED]> wrote: > Dear Mindaugas, > > Thanks for your promt response > > I've already tried this but.. it's not working,, what file do you think

Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4

:) . On Nov 27, 2007 3:33 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Tue, Nov 27, 2007 at 03:00:19AM +0530, Jaswinder Singh wrote: > > asterisk -rx "module load codec_g729.so" or "module load codec_g729" and > > shhow translation recalc > >

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

Its pretty clear from netstat that asterisk is listening on udp 5060 . It might be firewall configuration in server thats blocking it . Also you might have scanned for TCP port 5060 from outside and hence u find it closed ? On Nov 28, 2007 5:57 AM, Nick Brown <[EMAIL PROTECTED]> wrote: > Zaheer,

Re: [asterisk-users] SIP call interrupted after 64 seconds

Can you post the part of your dialplan which causes this behaviour ? On Dec 17, 2007 11:19 PM, Roger Schreiter <[EMAIL PROTECTED]> wrote: > Hi, > > some months ago, I had the problem with an asterisk-1.4.x- > Version, that some calls (but not all) were interrupted > 64 seconds after connect (a cal

Re: [asterisk-users] call-limit in database

call-limit is to set number of alternate calls . and L is to limit duration of each call . On Dec 22, 2007 2:54 PM, Pezhman Lali <[EMAIL PROTECTED]> wrote: > Dear > I am using this function with "L" > for example in the dbase. > app=Dial > appdata="SIP/[EMAIL PROTECTED]|60|L(10)" > it means di

Re: [asterisk-users] Your "favorite" Asterisk application.

I like the Echo application in asterisk ;) . Weird :P On Jan 24, 2008 7:07 PM, Mark Johnson <[EMAIL PROTECTED]> wrote: > Ken D'Ambrosio wrote: > > Hi, all. I've done some Asterisk recelling, but recently got roped into > a > > Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty m

Re: [asterisk-users] Realtime problem host='dynamic' in 1.2.26.1

Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Quite obvious .. doest sippeers have that row ? On Jan 24, 2008 6:04 PM, Torbjörn Abrahamsson < [EMAIL PROTECTED]> wrote: > Developers and maintaine

Re: [asterisk-users] Disable IAX2 call path optimization

You are usinfg sip or iax ? Its possible to prevent in both cases for sip under peer definition you can put canreinvite=no and in iax2 you can put transfer=no for asterisk 1.4 or notranser=yes in asterisk 1.2 .. Search for this on voip-info.org wiki for more info . On Jan 25, 2008 7:03 PM, <[EMAIL

Re: [asterisk-users] Meetme voice quality problems

Ubuntu has a real time kernel in repository "apt-get install linux-rt" . So you dont need to recompile . I think debian should also have one in repository , or u can manually compile a real time enabled kernel . Here's what is shows with real time patched kernel . dmesg|grep ztdummy [ 53.29307

Re: [asterisk-users] Enterprise or Fedora?

I prefer CentOS barebone install and yumming the way up for dependencies but manually compile asterisk/zaptel . Ubuntu servers are pretty good too since its repositories are quite bigger compared to CentOS . On Feb 2, 2008 11:45 PM, shadowym <[EMAIL PROTECTED]> wrote: > Actively maintained or act

Re: [asterisk-users] Call recording filename

I have figured out a way to include dialed number in recorded voicefile in freepbx . You have to edit /var/lib/asterisk/agi-bin/recordingcheck add this lines after $agi=new AGI() $temp= $agi->get_variable(DIAL_NUMBER); $agi->verbose("Number to be dialled is -{$temp["data"]}"); After this you can

Re: [asterisk-users] auto/forced call

No python code needed . Check .call files at http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out On 23/05/07, Brad Sumrall <[EMAIL PROTECTED]> wrote: Can anyone guide me to a "how to" on automating a call? I know a little piece of code (normally python) has to be place some wher

Re: [asterisk-users] Meet me

change conf => 222 to conf => 222 ( remove | ) I had same problem as freepbx always put | removing it fixed the problem On 29/05/07, Khaled Chehab <[EMAIL PROTECTED]> wrote: I am using asterisk 1.4.4 now and facing a problem with meetme,the code I was using with asterisk 1.2 is not function

Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

I think its a fair decision . 1.2 is very stable and they are not closing it all together , security issues will still be fixed . They need to concentrate more on 1.4 to make it bugfree . On 29/05/07, Stephen Bosch <[EMAIL PROTECTED]> wrote: John covici wrote: > I have an install using Rhino car

Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

Well i guess you just need a good look on logs for why and when you are getting core dumps . We are having few servers running .1.2.18 and it has turned out to be most stable in whole 1.2 branch ( had some issues with 1.2.13 and 14 ) . Except that for some users 1.2.18 is NOT stable. I've had

Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

What you say might be true for small business or home pbx systems . But if you have a production server handling sip/iax trunks over internet then you need to upgrade to avoid security related bugs and exploits that are released . You seem to miss the idea here. You work with a version that

Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

Well if you are out of luck with asterisk .. How about its fork callweaver ? I am highly awaiting its stable release to see if it holds upto what its wiki says . On 30/05/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote: > Michae

Re: [asterisk-users] channel_find_locked: Avoided deadlock

Is it over iax and there are lot of outgoing channels ? If yes then you are not the only person having this .. On 30/05/07, ram <[EMAIL PROTECTED]> wrote: Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 cha

Re: [asterisk-users] multiple host= in sip.conf

I dont think asterisk supports this . You can have host=dynamic and he can send calls from different servers . Problem will arise when you need to call him ( if registrations are enabled then latest registration will be getting call from you or you can directly send calls to his ip . ) On 30/05/0

Re: [asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!

Can you post some output from asterisk cli output while you make call ? On 30/05/07, BSumrall <[EMAIL PROTECTED]> wrote: after 18 hours, over 200 pages of reading, a complete reinstall of asterisk I am down to this. extensions.conf [globals] CONSOLE=Console/dsp IAXINFO=guest TRUNK=Zap

Re: [asterisk-users] False ring problem

Do you have 'r' in ur dial command ? 'r' makes asterisk produce ring . On 30/05/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote: Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy,

Re: [asterisk-users] Net2Phone Multiple SIP Trunk Not Working

You just have a 1 call limit on your account on net2phone side . Making 10 trunk wont let you make 10 account its restriction on your account not ip . Just change your provider . On 01/06/07, Salah Eddine ELMRABET <[EMAIL PROTECTED]> wrote: Hi, Any help regarding Net2Phone poblem? BR On 6/1

Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!

Well freepbx runs fine with asterisk 1.4.4 ( atleast for me ) i think some changes was introduced in 1.4 ( 1.4.4 ?) for some backward compatibility... like show channels now work in 1.4.4 instead of core show channels but it gives a notice that 'show channels' is deprecated bla bla .Freepbx wor

Re: [asterisk-users] yum om centos

independently install each rpm via rpm command :-/ On 04/06/07, Khaled Chehab <[EMAIL PROTECTED]> wrote: I have 2 servers, one connected to internet and the other is on a private lan have no access to internet. On the first server I update the kernel by yum update And installed asterisk pr

Re: [asterisk-users] any codec passthru mode

Asterisk by default uses the codec preferred by other device/client . Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough to check if it can avoid transcoding by forcing same codec on other side of conversation . If both sides prefer g729 then asterisk does not do transcoding bu

Re: [asterisk-users] ringback detection

It just might be that your carrier is not sending ring . You can use 'r' in asterisk dial command in extensions.conf to generate ring from asterisk . On 31/05/07, dima <[EMAIL PROTECTED]> wrote: Hello, everyone. Could anyone explain me how does ringback detection works in asterisk. Sometimes, wh

Re: [asterisk-users] g729

Are you trying to record the conversation as well ? On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote: I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just

Re: [asterisk-users] g729

=> _1NXXNXX,6,Set(CALLERID(number)=14073844200) exten => _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49) exten => _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Wedne

Re: [asterisk-users] Voip-info.org

Yep its down for me tooo . On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote: Is anyone else having trouble going into voip-info.org today? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or upda

Re: [asterisk-users] g729

Just read somewhere that you can use extension as g729 even in mixmonitor so it will record g729 stream and later you can convert it to mp3 or wav using sox . If this fails then try monitor application . On 06/06/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote: I think asterisk first co

Re: [asterisk-users] any codec passthru mode

has been patched here and there by one person or another, > but does anyone know if any of these patches to make CODEC negotiation > actually, you know, negotiate a CODEC will ever make it into the core > src? > > > Jaswinder Singh wrote: >> Asterisk by default uses the codec

Re: [asterisk-users] TCP<->UDP SIP proxy?

I think there is a patch for sip over tcp in asterisk but not sure if its stable or not try this http://bugs.digium.com/view.php?id=4903 You can also install openser as sip proxy . it supports sip over tcp . On Wed, 6 Jun 2007 19:14 +0300, Yehavi Bourvine +972-8-9489444 <[EMAIL PROTECTED]> wro

Re: [asterisk-users] Voip-info.org

Its up and working now . On 06/06/07, Compnet Bobby <[EMAIL PROTECTED]> wrote: Same in southern cali! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Wednesday, June 06, 2007 8:48 AM To: Asterisk Users Mailing List

Re: [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062

In sip.conf it should be bindport=5062 On 06/06/07, Crazy Boy <[EMAIL PROTECTED]> wrote: Hi Friends, I want to use 5062 port for SIP protocol. I made the below modifications in my server to use 5062 port. Polycom phone: port=5062 Trunk settings: port=5062 sip.conf: bindaddr=5062 Extension con

Re: [asterisk-users] IAX trunk with dynamic IPs

In your no-ip client set it to update ip every 2 minutes or so . and /etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt its 300 ( 5 minutes) On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote: Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary fo

Re: [asterisk-users] IAX trunk with dynamic IPs

s for Asterisk. Do you have any? Ronaldo. Jaswinder Singh wrote: > In your no-ip client set it to update ip every 2 minutes or so . and > /etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt > its 300 ( 5 minutes) > > On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTE

Re: [asterisk-users] No audio after Dial with G option

Remove Answer() and try . On 12/06/07, Rosalinda Trevino Cadena <[EMAIL PROTECTED]> wrote: I'm using the Dial application in the extensions file with the G option to execute an AGI script after the Dial (I need to track the call status) as follows: exten => _X.,3, Dial({DIAL_STRING},,G(_X.^4)

Re: [asterisk-users] Qualify renders all SIP peers unreachable

What does "sip show peers" output ? Also set a timeout in millisec like qualify=200 instead of qualify=yes On 14/06/07, randulo <[EMAIL PROTECTED]> wrote: I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT. All SIP peers are working properly to place or receive calls. Any SIP

Re: [asterisk-users] SIP Peering--call terminated prematurely

Please do not post same thing again and again . It wont help you get better replies , Post you asterisk cli output while call is in progress and when it disconnects prematurely . On 18/06/07, Don Kelly <[EMAIL PROTECTED]> wrote: I am attempting to establish SIP peering between Asterisk and an A

Re: [asterisk-users] Binding to multiple addresses

You can use bindaddr=0.0.0.0 to bind to all interfaces in sip.conf and iax.conf . On 23/06/07, Jordan Novak <[EMAIL PROTECTED]> wrote: I have a simliar problem as the port binding question. I have a four port parelell processing NIC, I would like to team them together. Can I do this in asteri

Re: [asterisk-users] Use of ChanSpy

exten =>*76,1,Answer exten => *76,2,Chanspy(|qb) ; q for quiet and b for only bridged calls exten => *76,3,Hangup Now you can spy on any call ,. All you need to do is press * again and again to change calls . Like if 3 calls are going then you can switch between calls by pressing * and # increa

Re: [asterisk-users] CDR Records "s" as dst

This is due to changes in cdr in asterisk 1.4.5 so in all outgoing dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if it was intended to be that way . On 25/06/07, Troy - Purple Oranges <[EMAIL PROTECTED]> wrote: > I am using VoiceOne http://voiceone.it/ as my management interf

Re: [asterisk-users] Asterisk + hinting presence + macro

It was due to changes in cdr in asterisk 1.4.5 previous version does not do it .there is a fix on bugs.digium.com or you can wait till next release or use asterisk 1.4.4 On 28/06/07, Rob Schall <[EMAIL PROTECTED]> wrote: I currently have about 50 polycom 501 phones on my asterisk setup. The dia

Re: [asterisk-users] Asterisk + hinting presence + macro

Sorry i didnt read your mail properly . I thought your problem is with cdr's. Here's link to cdr problem :) http://lists.digium.com/pipermail/asterisk-dev/2007-June/028085.html see the next message for patch . On 29/06/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Rob Schall wrote: >

Re: [asterisk-users] v1.4.x ready yet?

I jumped into asterisk 1.4 and its pretty stable .. i never got a core dump but it did halt while reloading a few times . I am back on asterisk 1.2 now but i think asterisk 1.4 is stable . On 29/06/07, Bruce Reeves <[EMAIL PROTECTED]> wrote: While I have not jumped all my systems to 1.4, there

Re: [asterisk-users] Question about dnsmgr

Are you sure calls were dropped with change in IP ?? I think it should let current calls run and use new IP for new connections . However if destination serv drops calls then it's a different story . On 03/07/07, Henry J. Cobb <[EMAIL PROTECTED]> wrote: Asterisk 1.4.5 full log: [Jul 2 09:31:16

Re: [asterisk-users] Google acquires Grand Central

Think about voicesense which will sense what you are talking and pop in a *relevant* voice ad to spice up conversation :P . On 03/07/07, Dean Collins <[EMAIL PROTECTED]> wrote: Ooops did Google just become a carrier :) http://googleblog.blogspot.com/2007/07/all-aboard.html I hear stocks cru

Re: [asterisk-users] Upgrade Asterisk

Yes that is write order . libpri then zaptel then asterisk . Remember that zaptel compilation is not required if you are using asterisk for voip only environment .But it's always good to install it before asterisk if you want to use conferencing abilities of asterisk . Regards, Jaswinder

Re: [asterisk-users] Asterisk console filtering and logging

extension/peer ( please correct me if i am wrong ). Regards, Jaswinder Singh. On 05/07/07, Eugene Prokopiev <[EMAIL PROTECTED]> wrote: Hi, Is it possible to filter messages on asterisk console, which was started with -, to see messages only for one extensions? By default there are all me

Re: [asterisk-users] Upgrade Asterisk

Yes just download new version of asterisk,zaptel,libpri . "make install" for all 3 ( first libpri , then zaptel, then asterisk ) . It is recommended to stop asterisk b4r doing "make install" of new version . Do not do "make samples" or it will overwrite you config's . After installing newer zapte

Re: [asterisk-users] Simple CDRs w/Asterisk/OpenSER.

Asterisk is poor with codec negotiation . It does not check if it can avoid transcoding by forcing codec available to both sides .. instead it will read it's config file and will select first allowed codec that is also available on other device on each leg of call and happily transcode between t

Re: [asterisk-users] Which features are lost when canreinvite is turned on ?

If you manage to get everything working with canreinvite=yes ( i suppose u figure out nat issues ) then you cant play music on hold , can't record calls , and can't do most of pbx stuff asterisk is capable of .. but dont worry asterisk doesnt disable all this features if canreinvite=on .. like if

Re: [asterisk-users] Best and easiest soft phone for my Dad..

Get portsip ( www.portsip.com ) its realtively easy to configure ( just push in user/password and server name at startup ) .. there might be NAT issue so make sure you have nat=yes in ur asterisk's sip.conf for the peer definition . If it still doesnt work then you need to find a iax phone like z

Re: [asterisk-users] asterisk is not sip proxy

Asterisk is not a sip proxy but it *can* partly act as a sip proxy if reinvites are enabled ( canreinvite=yes in sip.conf ) only then asterisk connects 2 end points directly and does signalling between them . Asterisk is a PBX now suppose u need to record all calls ..do conferencing stuff then rt

Re: [asterisk-users] SIP IP Trunk, between Asterisk and Softswitch

In your case it will send calls without registering to softswitch . Btw what does your softswitch expects from asterisk ? like is it configured to authenticate by username alone , user/pass or ip address ?? People here can help you better if you post that info . On 24/07/07, bilal ghayyad <[EMA

Re: [asterisk-users] What is the best softphone work with Asterisk

Idefisk/zoiper softphone is for IAX2 and it works fine almost everytime . However there is more variety in sip softphones . I think zoiper is much better than other iax2 softphones . On 25/07/07, bilal ghayyad <[EMAIL PROTECTED]> wrote: Hi All; Thanks for all replies :) - But that means, sof

Re: [asterisk-users] What is the best softphone work with Asterisk

Idefisk is now renamed to zoiper . http://www.zoiper.com/ :) On 26/07/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote: Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad: > Hi BaharatSamaria; > > Thanks for your kindly email. > > Are (Xlite and phoner) IAX or SIP? From where

Re: [asterisk-users] Lines Not being Hung UP Major

Btw are the phones behind NAT ?? Also you can try some softphone and make sure that this problem is caused by snom phones or some other factors .. On 25/07/07, Michael J. Liberatore <[EMAIL PROTECTED]> wrote: I thought it was the fios service but now I realize it's the snom 360! It doesn't hang

Re: [asterisk-users] Locking a device to a codec

in ur sip.conf under the device definition you can set it for example device name is asterisk is pap2 [pap2] username=pap2 secret=blabla type=friend disallow=all allow=g729 Then asterisk will only use g729 for incoming as well as outgoing calls on this device . On 27/07/07, Matt <[EMAIL PROTECT

Re: [asterisk-users] SIP "Max Channels" Setup

http://www.voip-info.org/wiki-Asterisk+config+sip.conf * call-limit * = number : Number of simultaneous calls through this user/peer On 27/07/07, Nicholas Blasgen <[EMAIL PROTECTED]> wrote: > > I'm running Asterisk without Free

Re: [asterisk-users] 1and1 dedicated servers have been down for a few hours .

Is this a web hosting forum or mailing list ? On 31/07/07, Asterisk guy <[EMAIL PROTECTED]> wrote: > > 1and1 dedicated server's service has been down for a few hours , unable > to reach them by phone or email. do anyone know what is going on there ? > > Mario > > ___

Re: [asterisk-users] Silly MeetMe() question.

Do you have proper version of zaptel installed corresponding to your asterisk version ? On 31/07/07, Knud Müller <[EMAIL PROTECTED]> wrote: > > Alex Balashov schrieb: > > On Mon, 30 Jul 2007, Knud Müller wrote: > > > >> what does your modules directory contain? Can you find a file > >> /usr/lib/as

Re: [asterisk-users] partial ChanSpy

Chanspy() app allows spying live channel but you will get 2 way voice in it . I dont think any other app allows to spy on one side of call . On 03/08/07, nik600 <[EMAIL PROTECTED]> wrote: > > Hi > > is it possible to spy (not record, spy) partially on a channel? > > for exaple, i'd like to listen

Re: [asterisk-users] iax2 registration being rejected

> > > Yes, since IAX2 only uses one port, this is correct. Another thing to > keep in mind is to set a low qualify value in Asterisk since some > routers will tear down the connection pretty quickly. The qualify acts > as a keep-alive and prevents the router from closing the port and losing > the

Re: [asterisk-users] CDR/MySQL basic config

sock=/tmp/mysql.sock Is this path for socket correct ? In some distro it is /var/lib/mysql/mysql.sock . Type "locate mysql.sock" in shell . Also remove uncomment port=3306 if using socket to connect . On 07/08/07, Alessandro Russo <[EMAIL PROTECTED]> wrote: > > Hi, try to login as asteriskcdru

Re: [asterisk-users] Use of context=... in [default] section of sip.conf

When you make calls then context=xxx of the peer you are using ( your extension ) will matter , the context=yyy line of your trunk wont matter . If you dont specify a context= for a peer then it is considered to be in [default] context . On 07/08/07, Jared Smith <[EMAIL PROTECTED]> wrote: > > On

Re: [asterisk-users] asterisk wait for traling digits

This should be configured in phone system instead of asterisk :) . On 08/08/2007, Michael Rice <[EMAIL PROTECTED]> wrote: > > This is part f the phones dial plan. Our aastra phones do the same > thing. Most phones allow you to configure the dial plan on them. > > satish patel wrote: > > i have onl

Re: [asterisk-users] les.net losing DID's

Please stop advertising your forums/services on every single chance u get on users list . On 08/08/07, Al Bochter <[EMAIL PROTECTED]> wrote: > > That is why you need to start posting info about the providers at > > http://www.bochterservices.com/phpbb/ > > so everyone knows > This is a FREE SERVI

Re: [asterisk-users] pick sip channel whn two party talking

google for ASTERISK CMD CHANSPY and follow voip-info link in search results . On 08/08/07, satish patel <[EMAIL PROTECTED]> wrote: > > Dear all > > i need this feature in asterisk whn 2 party calling that > time i pickup call and listen conversation of that party spoofing like is

Re: [asterisk-users] CDR-CSV Processing

Enable mysql loggin of cdr's by installing asterisk-addons and use asterisk-stat http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54 On 13/08/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > On Mon, Aug 13, 2007 at 08:49:11AM -0500, Jeremy Mann wrote: > > Does anyon

Re: [asterisk-users] CDR billsec greater than duration

I made same thread few months ago and many people said that they dont have such records in plain asterisk install ( no freepbx ) . I was also using freepbx when i had this problem . Heres mine : mysql> select count(*) from cdr where billsec > duration; +--+ | count(*) | +--+ |

Re: [asterisk-users] CDR billsec greater than duration

e from one of my customer who has a > > very bad connection . > > > > On 16/08/07, *Edoardo Serra* <[EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>> wrote: > > > > I noticed that fpbx calls ResetCDR on call hangup (don't know why > this > &g

Re: [asterisk-users] asterisk multiport

What i actually do is make asterisk listen on some other port like 5097 and redirect port 5060 to it with iptables like this /sbin/iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5060 -j DNAT --to YOURIPHERE:5097 This works very well . If i make asterisk listen on 5060 and redirect say 5097

Re: [asterisk-users] A102d sangoma's card and ztdummy

Sin you have sangoma card , it will act as timer . You need to install meetme ( app_conference is not very stable last time i read ) . On 01/09/07, fateme fatah <[EMAIL PROTECTED]> wrote: > > Hi: > I want to have conference call service and I use A102d sangoma's card.Do I > should install ztdummy

Re: [asterisk-users] Configure extension by software

You can use asterisk realtime which can read sip config from database ( mysql/pgsql) . Your application can just write info to database and asterisk will read it and make peers . You can also include a custom config file within sip.conf and make your application write peer settings to that file an

Re: [asterisk-users] Difference in show channels

'show channels' shows only running calls while 'sip show channels' shows all running sip sessions including phones trying to register . On 09/09/2007, ram <[EMAIL PROTECTED]> wrote: > > Hi all > > what is the difference between > > show channels > > sip show channles > > i see the difference in b

Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

I prefer centos , debian/ubuntu are also a good option . It just depends on which distribution you are comfortable with . We also have asterisk running very stable on slackware . On 12/09/2007, Gordon Henderson <[EMAIL PROTECTED]> wrote: > > On Wed, 12 Sep 2007, Euler Pereira wrote: > > > Hey all!

Re: [asterisk-users] errors messages in asterisk CLI

Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away This part is more like mistake in /etc/asterisk/cdr_mysql.conf . Check it once and relaod asterisk , then you can type "cdr mysql status" in cli to check if it conn

Re: [asterisk-users] prepaid application recommendation

A2billing is very versatile and good solution for asterisk prepaid/postpaid billing . On 22/09/2007, Apa Minerala <[EMAIL PROTECTED]> wrote: > > You should make sure you know how to install it yourself. > > And you should also test it very very VERY carefully. > > I can't underline very enough. >

Re: [asterisk-users] SIP and Firewall

Here you go http://www.voip-info.org/wiki/view/Asterisk+firewall+rules . You can also set your rtp.conf properly and open very few rtp ports instead of all 1-2 udp ports . On 22/09/2007, Guenther Sohler <[EMAIL PROTECTED]> wrote: > > Hallo, > > I'd like to correctly set up my firewall in

Re: [asterisk-users] Multiple Home system with SIP

since asterisk is only using operating system's routing ability , you can always set static routes using route command in linux . On 26/09/2007, Jeremy Mann <[EMAIL PROTECTED]> wrote: > > Why did you waste time with this reply? You do realize some users don't > have control over their Exchange se

Re: [asterisk-users] echo problems

Also many people using softphone turn's on mic boost in windows xp which also makes echo if it is set to very loud . On 30/09/2007, Philipp Kempgen <[EMAIL PROTECTED]> wrote: > > http://linux.sgms-centre.com/misc/netiquette.php#threading > http://linux.sgms-centre.com/misc/netiquette.php#toppost >

Re: [asterisk-users] Injecting a sound file into a bridged call

See chanspy in asterisk 1.4 , it also has a whisper mode and you can talk to one party http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy . But i dont know how to play a recorded file in it . On 08/10/2007, Girts Graudins <[EMAIL PROTECTED]> wrote: > > Hello everyone, > > > > I'm looking f

Re: [asterisk-users] asterisk.conf and it's impact on CLI

astrundir => /var/run Change this to astrundir => /var/run/asterisk on 1.4 server and chmod /var/run/asterisk to 777 . make sure u create that directory as well . On 20/10/2007, Al lists <[EMAIL PROTECTED]> wrote: > > this message is basically tells you asterisk is not running. > can you check a

Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite

Latest version of X-lite does not have GSM codec . Downgrde your versiona dn you will get gsm codec . I read on their forums that next version will again be including GSM codec . On 03/11/2007, Julio Tejera <[EMAIL PROTECTED]> wrote: > > Latest version of X-Lite does not > support GSM codecs any

Re: [asterisk-users] Asterisk unable to register to tnet.it

Check dns server entries in asterisk box . /etc/resolv.conf . Put opendns servers ip there just to test . opendns ip's are 208.67.220.220 and 208.67.222.222 On Tue, Jul 15, 2008 at 2:19 PM, map <[EMAIL PROTECTED]> wrote: > Hi Giorgio, > > RE my point 2: > You should test a sip client, whatever you

Re: [asterisk-users] DID number

[442033553] user=442033553 type=pusers secret=1234 host=dynamic context=users nat=yes make it context=stations , i am assuming this is how your DID provider is sending u calls ? Let us know if your DID provider is just sending calls to your ip address or you are registering asterisk server with

Re: [asterisk-users] A request for your input.

ITS Open source related . On 22/03/07, Bill Hackensack <[EMAIL PROTECTED]> wrote: On 3/22/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hello > > > P.S The program that I am using is open source, of course > ( www.phpsurveyor.org)! What part of the survey is running Asterisk? __

Re: [asterisk-users] Outbound SIP call from asterisk extension

[outgoing] exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]) Whats the dialplan number to ring to userA on server (ONDO) ? if u know that try exten => 1234,1,Dial(SIP/[EMAIL PROTECTED] ) since ur sip.conf has [*192.xxx.xxx.xxx*-out]. i am not sure why you use Sip/test to call to userA . Is that how

Re: [asterisk-users] Failure acknowledgement time

You can use qualify=(time in ms) option in sip.conf but its phone's configuration that should register to asterisk everytime its reconnected . On 26/03/07, cimsi <[EMAIL PROTECTED]> wrote: Hi, I've noticed that if I disconnect or reconnect a phone from the net, Asterisk take long time to realiz

Re: [asterisk-users] error in FreePBX

disable voicemail for that extension .. apply settings .. re-enable voicemail .. re-apply settings . this helped me once before. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options vis

Re: [asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

Wow i need a tftp client to download it now . Nice April 1 joke :P . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-us

Re: [asterisk-users] Make an iso image or a kickstart-Really its too urgent

Why not use a asterisk specific live cd distribution like www.astlinux.org ? It is also installable on usb . You can copy your whole dialplan and settings ( all files in /etc/asterisk ) on a pendrive . On 25/04/07, Khaled Chehab <[EMAIL PROTECTED]> wrote: Dears its too urgent Can anyone guid

Re: [asterisk-users] Calls in ulaw, not gsm as desired

Try ilbc if the phone supports (free) or g729 ( better but your asterisk will need license if you want to transcode calls from g729 to other codecs or want to record calls ) . Also check your phones config if its support multiple codecs . . On 02/05/07, Rob Schall <[EMAIL PROTECTED]> wrote:

[asterisk-users] 0 duration but non-zero billsec in mysql cdr

I was just going through my call records ( stored in mysql database by cdr_MYSQL module ) and saw a record having duration = 0 and billsec of more than 50 seconds . I did a query on cdr where duration < billsec and saw that there were infact some 250 records with duration less than billsecond ( ta

Re: [asterisk-users] 0 duration but non-zero billsec in mysql cdr

Someone in -biz list pointed out that this could be a freepbx problem so i think i will go check there . @ Salvatore Giudice: how can i intentionally do it ? Damn i need a app that can make sure customer phone doesnt hangup for the time i specify .. even if customer breaks his phone . lol

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