[asterisk-users] loop detected

I have an asterisk 1.4, that was working properly, but from last week, without any changing in the config of asterisk, all of calls,fall in loop detected error. there is two strange actions: 1-the first call after restarting the asterisk, is done successfully . 2-no packet , was sent to the

[asterisk-users] running twice

Dear I am using an asterisk 1.2.7.1 , with postgres and safe_Asterisk, for running, asterisk. but there is a problem, after 2-3 hours after restarting any things, top shows me, that, two asterisk, are now running, and one of them, gets 99.7 percent of cpu. Do you have any idea? Best Mani

Re: [asterisk-users] running twice

:01.97 httpd 31092 root 20 0 40092 3324 2904 S 0.0 0.7 0:00.03 ser --- Benjamin Jacob [EMAIL PROTECTED] wrote: show us the output of ur top command Pezhman Lali wrote: Dear I am using an asterisk 1.2.7.1 , with postgres and safe_Asterisk, for running, asterisk

[asterisk-users] psql

Dear I am using ast 1.4.19 with postgres. the realtime extension was done properly, but the two following warning was reported, 1)realtime_pgsql: Postgresql RealTime: Could not find any rows in table extensions. 2)realtime_multi_pgsql: Postgresql RealTime: Could not find any rows in table

[asterisk-users] max retry

my new asterisk server 1.4.19, disconnected the established calls after the 6 times, retries, when the quality of Bandwidth between cisco(2600) and server(asterisk) is not well. but there is no problem, with asterisk 1.2.7 please help me ___

[asterisk-users] 3 ways

Dear, after a lot of searching and testing I can not find a total solution for nat, with ser -- asterisk. now I have 3 selections: 1)using iax-phones instead of sip phones with asterisk 2)using sip phones registered in asterisk, 3)using sip phones with ser/openser and, searching for new ways,

Re: [asterisk-users] Asterisk Database Handling

using odbc+( postgres or mysql) is more stable, but at all odbc + postgres is recommended --- Sherwood McGowan [EMAIL PROTECTED] wrote: Steve Prior wrote: Tilghman Lesher wrote: Correct; it's actually a workaround for a bug in the MySQL drivers. It was discovered long after 1.2

[asterisk-users] iax test

Dear, is any test for using iax-phone with asterisk in larg system? for example cpu-users, ram-users, cpu-call, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] asterisk was discunnected suddenly

Dear, your hardware is good for more than 200-300 calls, configure asterisk for more details in debug, the output in console is more useful. also plz attach your main configurations for conference, viewing consumed ram and cpu during conference, can help --- On Mon, 6/16/08, fateme fatah [EMAIL

[asterisk-users] disconnection from caller did not recognized

Dear, I am using ser + asterisk, for outgoing calls, my problem is that the session was not closed if the caller says bye. can u help me ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22

Re: [asterisk-users] sendmail file

your mail is not clear at all. if you want to change the path of sendmail ,do this with mailcmd, in the voicemail.conf, if you want to send a voicemail to a class of emails, using dbase is more easier. let me to know more, about your problem. --- On Sun, 6/29/08, fateme fatah [EMAIL

[asterisk-users] realtime outgoing

Dear, Is any configuration for using outgoing via database(realtime)? Best Mani ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net

[asterisk-users] realtime outgoing

Dear, is any solution for replacing .call files into the database? best ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net

[asterisk-users] play remote file

Dear, do u have any idea to playback a remote file (with url address) ? for example : exten = _X.,1,playback(http://www.test.com/test.gsm;); best Mani ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 -

[asterisk-users] meetme without zaptel

Dear, I have some limitations to install zaptel because of kernel reinstalling. also there is'nt any zaptel device installed in the server. but I need to install meetme,  for conferencing . can u help me ? Best Mani ___ -- Bandwidth and

Re: [asterisk-users] meetme without zaptel

Of Pezhman Lali Sent: September-11-08 5:59 AM To: asterisk Subject: [asterisk-users] meetme without zaptel   Dear, I have some limitations to install zaptel because of kernel reinstalling. also there is'nt any zaptel device installed in the server. but I need to install meetme

[asterisk-users] codec of channels

Dear, is any command to show the codecs of  channels , in asterisk 1.4? Best ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net

Re: [asterisk-users] codec of channels-solved

solved with sip show channels best --- On Wed, 9/17/08, Pezhman Lali [EMAIL PROTECTED] wrote: From: Pezhman Lali [EMAIL PROTECTED] Subject: [asterisk-users] codec of channels To: asterisk-users@lists.digium.com Date: Wednesday, September 17, 2008, 5:42 PM Dear, is any command to show

[asterisk-users] app_confrence with loud voices

Dear, I have a little  problem with app_conference, the very low power voices, were amplified, too much, and normal voices were destroyed. codec=g729 asterisk=1.4.19 app_conference =last released best Mani ___ -- Bandwidth and Colocation

[asterisk-users] appconference low quality g729

Dear, compiling appconference 2.0. with g729 enabled, makes the quality of voices too low, for low voices , there is'nt any problem, but normal voices have alot of noises. best Mani ___ -- Bandwidth and Colocation Provided by

[asterisk-users] any format

Dear can Background, plays wav format , for any incomming, codecs, best Mani __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and

[asterisk-users] ser problem

Dear I am using ser + asterisk, for setting up land line calling. only probelm, each unregistered soft phone can places the call only with callerid, this is critical problem, because any number(soft phone) , has a limit time to use this system, best Mani

[asterisk-users] asterisk and snmp

dear is any snmp access , for asterisk 1.2.* ? Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

[asterisk-users] realtime_extensions

Hi now, I am using, realtime connection(mysql) for dialplan, but the following line must be added ,manualy to extensions.conf, before reloading.for each new context. [NEW_CONTEXT] switch = Realtime/@extensions is there any idea, to add this line to dbase too? thanks in advance Best MAni

[asterisk-users] sendmail problem

Hi We have a SER + asterisk server, on the same computer. after starting sendmail service , the ser will be confused. we need sendmail to send voicemails . best Mani Never Miss an Email Stay connected with

[asterisk-users] web based sipphone

Hi dear is any web based sip-phone?opensource? best Mani Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA. http://answers.yahoo.com/dir/?link=listsid=396545367

[asterisk-users] sip tunnel

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani

[asterisk-users] h323

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI -- Executing Dial(SIP/2.2.2.2-086f5ac0, H323/[EMAIL PROTECTED]|60) in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No

[asterisk-users] web based sip phone

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and .. best MAni Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel

RE: [asterisk-users] web based sip phone

thanks Yuan I was search the best result is sipfoundary.org but it's client is not spesific for my purpose, but it will be. is any better answer for this searching? best Mani --- Yuan LIU [EMAIL PROTECTED] wrote: From: Pezhman Lali [EMAIL PROTECTED] Date: Fri, 30 Mar 2007 02:05:35 -0700 (PDT

[asterisk-users] xten web phone

hi xten.de produced an activex for web phone. but I can not find any link for download. can u help me ? best Mani Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo!

[asterisk-users] realtime problem

Dear the following is the asterisk's dbase(Mysql5). if the extension =17171000 asterisk run appdata=22, but I prefer to run appdata=333. let me know how I can run the appdata=3 best Mani mysql select * from ext;

[asterisk-users] calls bridging

dear can asterisk dial two numbers, then bridge them.(like jah jah) best Mani Looking for earth-friendly autos? Browse Top Cars by Green Rating at Yahoo! Autos' Green Center.

[asterisk-users] no real ring back

Dear I am using Ser+Asterisk, for sip providing. there is a problem, the asterisk does not return back the busy tones to the sip phones. for example, if the destination number is busy, we are hearing waiting ring from sip phones, and after 60sec(timeout) the call will be terminated. thanks

[asterisk-users] a2billing

hi My a2billing adds |HrL automatically to dial string, I can not find the source of this task, I need to remove r from all dial strings, Thanks for your help. Best Mani __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection

Re: [asterisk-users] changing the size of voice packets

, November 10, 2008, 3:00 PM Hi! On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali [EMAIL PROTECTED] wrote: Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. You can specify size of voice

[asterisk-users] changing the size of voice packets

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure.   thanks in advance Mani ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] changing the size of voice packets

is any command , shows the current rate of each channel?   --- On Mon, 11/10/08, Kristian Kielhofner [EMAIL PROTECTED] wrote: From: Kristian Kielhofner [EMAIL PROTECTED] Subject: Re: [asterisk-users] changing the size of voice packets To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Dial outside number using the E1 Link

Dear Fateme two good refrences: http://articles.techrepublic.com.com/2415-1035_11-94140.html and http://www.trixbox.org/forums/vendor-forums-certified/sangoma/solved-sangoma-101d-card-trixbox-asterisk-1-4-19-1 hope to help u best Pezhman --- On Tue, 11/11/08, fateme fatah [EMAIL PROTECTED]

Re: [asterisk-users] play file from url

mp3player, is just for your need, use it this like exten = _X.,1,mp3player(http://www.test.com/test.mp3;) try this page http://www.voip-info.org/wiki-Asterisk+cmd+MP3Player best --- On Wed, 11/12/08, Singer X.J. Wang [EMAIL PROTECTED] wrote: From: Singer X.J. Wang [EMAIL PROTECTED] Subject:

[asterisk-users] echo cancellation for sip phones

Dear, the sip phones that registered, in to the asterisk 1.4.x have the echo in their callings to pstn. how this echo can be canceled? Best ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] echo cancellation for sip phones

@lists.digium.com Date: Thursday, November 20, 2008, 12:01 PM Pezhman Lali wrote: Dear, the sip phones that registered, in to the asterisk 1.4.x have the echo in their callings to pstn. how this echo can be canceled? H - you don't give much to go on... What is the connection to the PSTN (i.e

[asterisk-users] reducing iax packet size

Dear, is any way to change the iax packets? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] asterisk 1.2 and openser 1.4

asterisk 1.2 , is enough old to make a lot problems, upgrade to 1.4 or 1.6 and enjoy it. integration opensips( ser) and astersik, is the best solution for the big voip systems. --- On Sat, 12/27/08, Mike Trest m...@trest.com wrote: From: Mike Trest m...@trest.com Subject: Re:

[asterisk-users] no busy here

Dear, I have combined asterisk 1.4 with cisco 2600 connected to PRI, the biggest probelm is that, the cisco does not send busy her sip_486 to asterisk, for busy callee . can u help me to find the solution? ___ -- Bandwidth and Colocation

[asterisk-users] local dialing

Dear, because of using dial(local/...) each incoming calls (_12X.) makes 4 ports on asterisk. I can not use goto , because of some limitations. is any way to decrease it? Best, [MAIN] exten = _12X.,Dial(LOCAL/${ext...@test/n,60) [TEST] exten _X.,1,Dial(${ext...@next_gateway,60)

[asterisk-users] custom cdr userfiled

Dear, I added new field to cdr table , named service and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten = _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Re: [asterisk-users] custom cdr userfiled

To: Asterisk Users asterisk-users@lists.digium.com Date: Monday, January 26, 2009, 1:18 PM Pezhman Lali schrieb: I added new field to cdr table , named service and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten = _X.,1,Set(CDR(service)=OUT

[asterisk-users] goto iax problem

Dear, the goto function to the iax dialing, makes bill duration and call duration wrong, in cdr.they are equal to ringing time. the cdr will be produced and saved into the dbase, when the callee picks up the phone. is any way to have real duration time ? [main] exten =

[asterisk-users] iax clients were unregistered after 30sec

Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys |

Re: [asterisk-users] iax clients were unregistered after 30sec

by using rtcachefriends=yes it was done. --- On Sat, 1/31/09, Pezhman Lali pezhman_l...@yahoo.com wrote: From: Pezhman Lali pezhman_l...@yahoo.com Subject: [asterisk-users] iax clients were unregistered after 30sec To: asterisk-users@lists.digium.com Date: Saturday, January 31, 2009, 7:34

[asterisk-users] analysing tools

I have problem with packet size of voip packets, in a big network. what is the best monitoring tools and analyzer for this purpose? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To

[asterisk-users] freemin managment for sim cards

is any program , to manage freemin on sim cards ,for gsm gateways that connected to the asterisk, for termination? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or

[asterisk-users] route by caller id

Hi I has installed, asterisk , with postgresql. it 's the view of extensions table: didex=# select * from extensions order by id desc limit 5; id | context |exten | priority | app | appdata |

[asterisk-users] new route by caller id

Hi I have installed, asterisk , with postgresql. it 's the view of extensions table: didex=# select * from extensions order by id desc limit 5; id | context |exten | priority | app | appdata |

Re: [asterisk-users] new route by caller id

Dear Mathew I found that u can setup astersik for routing by caller id, with dbase. I have installed, asterisk , with postgresql. it 's the view of extensions table: didex=# select * from extensions order by id desc limit 5; id | context |exten | priority | app |

[asterisk-users] anti ex-girlfriend

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 |2 | hangup | | 455 | DID

[asterisk-users] google talk

hi How does asterisk can act as google talk's client. for mapping, received calls , to google talk. tanx Mani Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com

[asterisk-users] pipemedia

dear I am searching for the company like pipemedia (legend.co.uk) in USA, or other european countris. I tested, didex.org, but pipemedia is more advance tele-communication company. please tell me, if you know. thanks best Mani

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

Dear I am using this function with L for example in the dbase. app=Dial appdata=SIP/[EMAIL PROTECTED]|60|L(10) it means dial 1 thru 1.1.1.1, with limitation=10 mili-second, and time out=60 sec best Mani --- Bhrugu Mehta [EMAIL PROTECTED] wrote: hi, all proble: I have add

[asterisk-users] disconnection silent channels

Dear,is any way to find silent channels , and disconnect them after 30 secs? best ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net

[asterisk-users] iax no way sound

Dear, some iax phones,(with built in router) have problem, with our asterisk server, there is no way sound if they call out, but it's ok if somebody calls them. the normal iax phones without router have'nt ny problem. can u help me? the version of kernel is 2.6.18 and asterisk is 1.4.26.2 Best

[asterisk-users] iax no way sound

Dear, some iax phones,(with built in router) have problem, with our asterisk server, there is no way sound if they call out, but it's ok if somebody calls them. the normal iax phones without router have'nt ny problem. can u help me? the version of kernel is 2.6.18 and asterisk is 1.4.26.2 Best

Re: [asterisk-users] Why does my IAX2 trunk between two office hangup a channel after 30 seconds? Can you share your IAX2 trunking configuration? URGENT HELP much appreciated

Dear Please send us, your iax configurations. best On Mon, Jul 5, 2010 at 7:10 AM, bruce bruce bruceb...@gmail.com wrote: Hi guys, I have two Asterisk servers (with FreePBX) connected together with IAX2 trunking. When I call from server A-B call connects but hangs up after 30 seconds. What

Re: [asterisk-users] Problem in establish call from a2billing users.

add the a2billing configurations to the sip.conf best On Thu, Jul 1, 2010 at 7:34 PM, bruce bruce bruceb...@gmail.com wrote: Yes, you are missing a whole bunch of configurations from creating SIP users to making sure they show as peers on Asterisk to making sure you use dnid, etc.You

Re: [asterisk-users] Problem with extensions in IVR and queues

please send your extension.conf 2010/6/30 Anahi Ludueña a_ludu...@hotmail.com Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the

Re: [asterisk-users] SendFAX dialplan example

don't forget to install spandsp, and replace the value of Channel with true value. best On Fri, Jan 28, 2011 at 4:26 PM, bakko asannu...@gmail.com wrote: Hello, you have to use a callfile http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Create a callfile, for example test.txt,

Re: [asterisk-users] Anybody ever see this before?

did u compile lib_pri ? On Thu, Jan 27, 2011 at 7:30 PM, William Stillwell will...@stillwellsoft.com wrote: [Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405 dahdi_pri_error: Should have only transmitted 0 frames! [Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405

Re: [asterisk-users] Reducing number of Asterisk processes?

check your /etc/asterisk/asterisk.conf and post it here best On Sat, Jan 29, 2011 at 2:22 PM, Gilles codecompl...@free.fr wrote: Hello On a uClinux-based appliance, ps aux shows multiple Asterisk processes: 380 root 11990 S asterisk -f 381 root 11990 S asterisk -f 383

Re: [asterisk-users] Reducing number of Asterisk processes?

= apache ;astctl = asterisk.ctl [compat] pbx_realtime=1.6 res_agi=1.6 app_set=1.6 On Sat, Jan 29, 2011 at 4:32 PM, Gilles codecompl...@free.fr wrote: On Sat, 29 Jan 2011 15:47:53 +0330, Pezhman Lali l...@lopl.net wrote: check your /etc/asterisk/asterisk.conf and post it here Here goes: root

[asterisk-users] faxter

Dear, Faxter is an opensource email to fax gateway, please check it, let me know if any bug. best -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] faxter

sorry for no url https://code.google.com/p/faxter/ https://code.google.com/p/faxter/best On Sun, Jan 30, 2011 at 12:51 PM, Pezhman Lali l...@lopl.net wrote: Dear, Faxter is an opensource email to fax gateway, please check it, let me know if any bug. best

Re: [asterisk-users] [newbie] Conference call

Dear, Meetme is a default conference application, but you can try conference or konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference

Re: [asterisk-users] Callback through extensions.conf?

Dear a2billing also provided call_back daemon, try it best On Sun, Feb 6, 2011 at 12:57 AM, Paul Belanger pabelan...@digium.comwrote: On 11-02-05 06:07 AM, Gilles wrote: I'd like to configure Asterisk so that... 1. I ring it from my cellphone with CID number displayed, just to notify

[asterisk-users] secure sccp

Dear is any way to have a secure (encrypted) rtp line between cisco 79XX and asterisk with SCCP? best -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Using files .call or AMI

as you know you have 2 ways. using ami or .call files. if you have experience, the AMI is more powerful. you must have a context in your extensions.conf to manage agent procedures, it looks like a simple context, that you must have, for managing queues. with .call file or ami dial your customers,

Re: [asterisk-users] Fax for Asterisk SIP-TDM

Dear I had good experience with asterisk + spandsp for sending and receiving fax, if your ip phone supports fax, you need asterisk only as g711(no vad) gateway. best On Sun, Feb 13, 2011 at 7:00 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/12/2011 10:53 PM, Mark Willis wrote: Is it

Re: [asterisk-users] trunks and phones registered from the same IP

really it's too difficult to understand, please explain more clear On Tue, Feb 15, 2011 at 5:17 AM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives

Re: [asterisk-users] further action after caller in a queue hangs up

you can run any function in your hangup extension, exten = h,1,... best On Tue, Feb 15, 2011 at 12:21 PM, Richard Zheng rzh...@gmail.com wrote: Hi, In ACD queue, is it possible for the agent to take some actions when the caller hangs up? For example, to let the agent to enter some

Re: [asterisk-users] unregistered trunks and registered phones coming from the same IP

please send your sip.conf, is any NAT procedure implemented in your network? On Mon, Feb 14, 2011 at 10:16 PM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound

Re: [asterisk-users] Dial command

this command will not work. what is your main purpose? do u need to have a conference with a group of sip phones? best On Tue, Feb 15, 2011 at 3:13 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: I am wondering if its possible to have sometime like this: exten 100 = Dial

[asterisk-users] changing logo of 7905

I know there is not a good place for ask this question. but I can not find in other ways. Dear, Do you have any experience for changing the logo of cisco 7905 on sccp firmware? best -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Cisco 7945G phone with asterisk

dear I have a good exp in setting up 79xx on sccp, with sccp-b library, and tftp server, which part is the main problem for you? best On Wed, Feb 16, 2011 at 3:10 PM, Andrew Latham lath...@gmail.com wrote: On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote: Hi, Anyone who

Re: [asterisk-users] DTMF not detected, time out

some outside sip provider does not accept dtmf, if you have not this problem in your local, ask your outside carrier best On Wed, Feb 16, 2011 at 7:27 AM, asterisk asterisk aster...@ck-lee.comwrote: In the past it was set as auto and worked. I change to RFC2833 but did not work. How can I

Re: [asterisk-users] Two Asterisk machines for redundancy

hi using database as realtime functions solves your first problem, for second try by using dns best On Mon, Feb 28, 2011 at 1:54 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I would like to have two Asterisk machines to have redundancy between them, so if first machine failed then we

Re: [asterisk-users] Two Asterisk machines for redundancy

help? Regards Bilal --- On *Mon, 2/28/11, Pezhman Lali l...@lopl.net* wrote: From: Pezhman Lali l...@lopl.net Subject: Re: [asterisk-users] Two Asterisk machines for redundancy To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: bilal ghayyad

Re: [asterisk-users] Prepaid Billing other than A2Billing

I think a2billing is the best billing opensource system, but try astbill, new url http://astbss.org/ http://astbss.org/but if you want to setup a large system select enterprise system, these systems are useful for small and med networks. best On Sat, Mar 5, 2011 at 8:56 PM, bilal ghayyad

[asterisk-users] fail2ban + asterisk

Dear this note is only for fresh administrators don't think about asterisk security. I found fail2ban very useful for anti asterisk hacking, so I want to share it with fresh admins. some hackers try your sip or iax2 ip with a lot of username/password, may be after 1 million try, one

Re: [asterisk-users] ignore this test

you can not see what you send, change the config in the mailing list options On Sun, Mar 6, 2011 at 6:36 AM, sean darcy seandar...@gmail.com wrote: I can't seem to send anything. Let's see if this shows up. -- _ --

Re: [asterisk-users] Fax

for your network it's optional to receive the fax on your server, you can pass the received fax to the destination, like a voice call with g711 and no VAD. ask if you need more info. best On Wed, Apr 6, 2011 at 4:55 PM, Bert Van Kets mail...@vankets.com wrote: On 1/04/2011 13:04, Khaled W.

Re: [asterisk-users] Iptables configuration to handle brute force registrations?

fail2ban(opensource) is a good choice for you best On Wed, Apr 6, 2011 at 1:16 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 5 Apr 2011, Steve Edwards wrote: On Tue, 5 Apr 2011, Gilles wrote: I'm no expert of iptables, and it seems like it can handle banning IP's

Re: [asterisk-users] agi create mailbox

using the realtime functions for voicemail solve this problem. you can insert a query from your agi to add new voicemail box. is it what you need ? On Tue, Apr 5, 2011 at 10:17 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 5 Apr 2011, vip killa wrote: Is it possible to create a

Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the h extension?

Dear there is some problem. the true way for running php script, is using agi not system. second after 5 sec, a lot of channel variables were removed, it makes your program wrong. with some little experience you can add your script to a2billing, try it. best On Sat, Apr 9, 2011 at 7:22 PM, Bruce

Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the h extension?

extension even if it was only run in x extension. Regards, On Mon, Apr 11, 2011 at 6:34 AM, Pezhman Lali l...@lopl.net wrote: Dear there is some problem. the true way for running php script, is using agi not system. second after 5 sec, a lot of channel variables were removed, it makes your

Re: [asterisk-users] accessing currents calls from outside asterisk

yes, ami is your unique answer. what is msisdns ? On Wed, Apr 13, 2011 at 3:18 PM, Albert alber...@wp.pl wrote: Hi, I am working on integration of 2 systems: asterisk and messaging platform. What I need is to access somehow information about current calls. Should I do it over AMI ? I

Re: [asterisk-users] Nat=yes

check this http://www.voip-info.org/wiki/view/Asterisk+sip+nat On Thu, Apr 21, 2011 at 2:12 PM, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can

Re: [asterisk-users] IAX2 codec selection and video

check this url, let me know if any problem http://www.voip-info.org/wiki/view/Asterisk+video http://www.voip-info.org/wiki/view/Asterisk+video http://www.voip-info.org/wiki/view/Asterisk+videobest On Thu, Apr 21, 2011 at 9:00 PM, Steve Davies davies...@gmail.com wrote: Hi, Can anyone let

Re: [asterisk-users] odbc error - server is gone

check your odbc connection with isql best On Fri, Apr 29, 2011 at 9:33 PM, Warren Selby wcse...@selbytech.com wrote: You're using 1.4.2. Why not try upgrading to a more recent release of 1.4 (I believe 1.4.41 is current) and see if your issue has been resolved. Thanks, --Warren Selby,

Re: [asterisk-users] SIP bad request

may be the ip phone has the problem, try reset as factory On Fri, Apr 29, 2011 at 8:03 PM, Mike l...@net-wall.com wrote: What I am looking for? Here is a snippet, with some info obfuscated. I can see the bad request, but why there is such a message isn’t obvious. --- SIP read from

Re: [asterisk-users] AGI WAIT FOR DIGIT - key press BEFORE command

Dear try phpagi. it has a lot of useful functions. in this scenario you will lose your digit, set a check point between each digit gathering best On Wed, Apr 27, 2011 at 6:17 PM, David asterisk@spam.lublink.netwrote: Hi, Consider the following situation : SIP/asterisk-001dAGI Rx

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