Re: [asterisk-users] asterisk-users Digest, Vol 36, Issue 25

Hi Thanks for your help. What do I need to change in Makefile for ooh-323 to be loaded? Regards -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de clive.chan(Alpha Trilogies Networks) Envoyé : mardi 10 juillet 2007 07:17 À : asterisk-users@lists.digium.com

Re: [asterisk-users] Very bad TDMF tone !

Hi, Yup, I agree. You have to post out your Zapata.conf for discussions, and Have you tried out the cable terminations and also the incoming DTMF signal voltage level? It has so many possibilities for this case. > >> i am using tdm400P in my office. i tested that TDMF generated by asterisk

Re: [asterisk-users] List delays

Most of the users using this list do not experience the issue you are having, rather than insult the admins, please trouble shoot and if you cannot, at least post headers so others can. -- Original Message -- From: Dimitri Volski <[EMAIL PROTECTED]> Reply-T

Re: [asterisk-users] asterisk-users Digest, Vol 36, Issue 25

Hi Stephane, Solutions remove all the Asterisk modules as well as the addon modules, and re-install Asterisk and Addon again. # rm -rf /usr/lib/asterisk/modules/* # rm -rf /usr/include/asterisk Then reinstall it again. It works for me. Have you change the addons Makefile so that the ooh-323 fil

Re: [asterisk-users] Very bad TDMF tone !

OK, tel me put my share of experience. I have a PRI & TE212P which had onboard VMP & echo cancellation. I am using Asterisk-1.2-18. the DTMF issue was really bad. It rang the wrong extension some tings rang invalid extension. In the begining I didnt knew that it was an DTMF issue. Th

[asterisk-users] ZAP TDM and DTMF issue

Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not

Re: [asterisk-users] callback and bridge problem

Are you behind NAT ? Do you have canreinvite=yes ? - Original Message - From: "Adam KOSA" <[EMAIL PROTECTED]> To: Sent: Monday, June 25, 2007 6:37 PM Subject: [asterisk-users] callback and bridge problem > Hi guys, > > sorry for the long e-mail, i'm only trying to give as much informat

Re: [asterisk-users] call fail from audiocode to sip trunk

What error are you getting on the Audio Codes side ? Set verbose to 5 on the Audio codes box and try running Syslog. - Original Message - From: satish patel To: asterisk-users@lists.digium.com Sent: Tuesday, June 26, 2007 2:14 PM Subject: [asterisk-users] call fail from audioc

[asterisk-users] Asterisk 1.4.7 and MOH

I just installed the newly released Asterisk 1.4.7 and I cannot get music on hold. I am using the default settings with the wav files. Here is what I get on the cli from any sip phone: -- Executing [EMAIL PROTECTED]:1] NoCDR("SIP/1120-084e6010", "") in new stack -- Executing [EMAIL

[asterisk-users] how to register several clients with different number but using single authentication account ?

Hi all, I'm running some performance tests over my Asterisk,to simplify the test,I want to configure Asterisk to allow several clients registered with different nubmers but using single authentication account,so that Asterisk would not check the "to" "from" head when challenging registartion & in

Re: [asterisk-users] Help. Cannot compile version 1.4.6 with the following error

Wai Wu wrote: > > Hi all, > > I need the zap channels going, but got the following error. What do I > need to change in my configuration? Thnx. > > chan_zap.c: In function `zap_send_keypad_facility_exec': > chan_zap.c:2309: warning: implicit declaration of function > `pri_keypad_facility' > cha

Re: [asterisk-users] DTLS availablity?

Noah Miller wrote: >> Is DTLS available for Asterisk on any Linux distro? >> > > Nope. > > I've read that the reSIProcate SIP stack has DTLS support. I found out that DTLS is in openSSL 0.9.8. This is available with Redhat/Centos 5. So the code is there. Perhaps just configuring it to some p

Re: [asterisk-users] List delays

There is definitely something wrong with this list. I have my emails sorted by date, and every day, the emails do not just come on top, but get slotted in. Today (10 July 2007), I received about 6 emails from 29th of June, couple from 30th, up until the 5th of July, nothing of today's, or, well

Re: [asterisk-users] ipv6 patch

Hans Witvliet wrote: > I intended to ask, wether it would remain for the time being a > bleeding-edge-patch, or already included into the svn-tree. > Either way, i presume that i shouldn't hold my breath while waiting for > the first 1.6 ;-)) (six-months, a year?) As far as I know, the patch is re

Re: [asterisk-users] OpenSer/Asterisk PBX solution

Thank you for your input it is very helpful - Original Message - From: "Dovid B" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] OpenSer/Asterisk PBX solution Date: Mon, 9 Jul 2007 17:05:57 +0300 I have some clients using Enswitch (Pai

[asterisk-users] Meetme delay?

I recently installed 1.4.5 and I've noticed a recurrence of a problem that I thought was solved long ago, namely a very long (2-4 seconds) delay on meetme calls. That means with two people in the conference room, it takes 2-4 seconds for what one person says to reach the other person. Is anyone e

[asterisk-users] Asterisk 1.2.21, 1.4.7 and Libpri 1.2.5, 1.4.1 released

The Asterisk development team is proud to announce a new batch of releases. There are new releases of Asterisk and Libpri for both the 1.2 and 1.4 series. The development team has been working especially hard on fixing bugs in our existing release branches. These releases are regular maintenance

Re: [asterisk-users] ipv6 patch

On Sun, 2007-07-01 at 18:27 -0500, Russell Bryant wrote: > Hans Witvliet wrote: > > Before taking a plunch into the code > > Marc Blanchet wrote that he's making code ip-version independant. > > How much of these improvements have already made it into the 1.4 > > branche? > > None, and they ne

Re: [asterisk-users] List delays

On Wed, 2007-07-04 at 09:57 -0500, John Faubion wrote: > >Is it just me? After the mail list server upgrade, the average delivery > >time for messages to the users list is between 4 and 5 days. The Dev > > I've seen several people mention it taking a few days to send messages. I've > usually see

Re: [asterisk-users] Queue Status

Arun Kumar wrote: > Hi > > I already tried asterisk manager but Im not able to get status for each > queue member. > > thanks > That must be a problem with your configuration. I get QueueMemberStatus on my AMI interface (1.2): Event: QueueMemberStatus Privilege: agent,all Queue: support L

Re: [asterisk-users] Basic asterisk Autodialer?

Call files and app_amd (Answering Machine Detection) come to mind. app_amd can take a little time to tune, but you can get it to be pretty reliable in most cases. See: http://www.voipinfo.org/wiki/index.php?page=Asterisk+cmd+AMD http://www.voipinfo.org/wiki/view/Asterisk+auto-dial+out Mat

Re: [asterisk-users] DTLS availablity?

> Is DTLS available for Asterisk on any Linux distro? Nope. I've read that the reSIProcate SIP stack has DTLS support. - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or updat

[asterisk-users] Digium cards for sale in Pakistan

Hello Users, I have 2x2 port T1/E1 cards for sale in Pakistan. Cards are in warrenty and going cheap as i have purchased additional cards. regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To

Re: [asterisk-users] Early Media Handling

Hi Arun - > using php script and Asterisk manager I'm dialing numbers and once gets > connected send to an exten in my dial plan that plays an automated message > but some time without answering even it goes to my exten. How can I handle > early media in Asterisk that is I want only when user answ

Re: [asterisk-users] Very bad TDMF tone !

> >> i am using tdm400P in my office. i tested that TDMF generated by asterisk > >> is so bad. the sound is very soft and quality is so bad. i am using > >> asterisk 1.2.18. most of time, the # key can not be detected correctly. > >> Does anyone has that problem? > >> please give me a hit for that

Re: [asterisk-users] Basic asterisk Autodialer?

Shawn, > Just call and play the message and move on. Trying to find a way to > notify a couple hundred customers that their service has been changed. > > Anyone have any easy ways to do this? I already have a functioning > asterisk server with a POTS interface, etc. Set up a dial plan ext

Re: [asterisk-users] Setting Appearance on Outbound Calls?

Hi Matt - > What do I need to do to set the outbound appearance on a call so that > it shows up as Unavailable or Private? In most cases, I think you'd need to arrange this with your provider. If you want to do it on a call-by-call basis (in the US), dial *67 before you dial the number. If you ha

Re: [asterisk-users] Basic asterisk Autodialer?

[EMAIL PROTECTED] wrote: > I'm looking for an easy way to make asterisk perform as a basic > (broadcast)autodialer. > > Basically all I want to do is give it a list of phone #'s and a > pre-recorded message and have it call each one and play the message or > leave it on the person's answering machi

Re: [asterisk-users] Asterisk Help

Hi Arun - > I need help in configuring a auto dialer system using Asterisk. I'm holding > my customers number in MySQL want to fetch 10 numbers one time and dial if > gets connected and answered by customer wants to play a sequence of message > > I've tried here is my code to place calls but in th

[asterisk-users] Basic asterisk Autodialer?

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling ar

Re: [asterisk-users] Polycom multiple registrations

> The 430's have two line appearances. I'm trying to get the second line > registered to a different extension but for some reason it's not > allowing me to do this. The first line will register fine but the second > line never seems to register no matter how I swap the device ID's and > permission

Re: [asterisk-users] Call Waiting curiosity...

Is your incoming context using chanisavail, while your internal-dialing context is not, and just sends the call, without checking? Mojo Michael Wareman wrote: > Hi, > > I have (to me) an interesting problem. > > There are 3 physical extensions, 11, 12 and 13. All hang off Sipura > adapters. >

[asterisk-users] allow third party registration/invitaion

Hi all, I'm running some performance tests over my Asterisk,to simple the test,I want to configure Asterisk to allow third party registration and invitation,so that Asterisk would not check the "to" head when challenge registartion and "from" head when challenge invitation,and I can only create on

[asterisk-users] Setting Appearance on Outbound Calls?

What do I need to do to set the outbound appearance on a call so that it shows up as Unavailable or Private? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: h

Re: [asterisk-users] List delays

I received your message just a few minutes after you sent it; however, it sometimes takes 3-4 days before I see messages I post coming back to me on the list. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message---

Re: [asterisk-users] Monitor events?

James FitzGibbon wrote: > On 7/9/07, *Daniel Gradecak* <[EMAIL PROTECTED] > > wrote: > > > are you sure the monitor is started and sotoped via the dialplan ? > > > > If you're using Monitor() or MixMonitor(), then just add a UserEvent() > call just before it in the

Re: [asterisk-users] Google acquires Grand Central

On Mon, 9 Jul 2007, Wendell Hamilton wrote: > GrandCentral doesn't do anything you can't do with asterisk. What it > does do is put those features within reach of an average person by > providing a superb user interface for the end user, which allows them to > self-administer all of these wond

Re: [asterisk-users] Google acquires Grand Central

- "Alex Robar" <[EMAIL PROTECTED]> wrote: > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-us

Re: [asterisk-users] ipv6 patch

On 01:06, Mon 02 Jul 07, Hans Witvliet wrote: > On Sat, 2007-04-07 at 10:57 +0200, Michiel van Baak wrote: > > > Read > > http://svn.digium.com/view/asterisk/team/blanchet/v6/README-IPV6.txt?view=markup > > before running this code. > > > > Before taking a plunch into the code > Marc Blanche

Re: [asterisk-users] Polycom 301 - Problem with AMI Originated Calls

Lee Jenkins wrote: > Hi all, > > I'm having an odd problem with my polycom 301. I am initiating a call > to it with AMI Originate() function: > > Action: Originate > Channel: local/[EMAIL PROTECTED] > Context: to_meetme > Exten: s > Priority: 1 > Variable: dropped_conf=111 > > The "to_meetme"

[asterisk-users] Problems sending more than 2 SMS with asterisk / smsq

When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled

Re: [asterisk-users] Monitor events?

Hi Stefan, actually you probably know i am using your java-asterisk :) Yes the best solution i found till now it was to add those events to res_monitor.c. I wonder why it was not yet done, may be there was a reason or nobody needed it yet. Anyhow this would be a cool feature that others should

Re: [asterisk-users] Monitor events?

On 7/9/07, Daniel Gradecak <[EMAIL PROTECTED]> wrote: are you sure the monitor is started and sotoped via the dialplan ? If you're using Monitor() or MixMonitor(), then just add a UserEvent() call just before it in the dialplan. If you're doing monitoring of queues, it's a bit trickier - yo

Re: [asterisk-users] List delays

John Faubion wrote: >> Is it just me? After the mail list server upgrade, the average delivery >> time for messages to the users list is between 4 and 5 days. The Dev >> > > I've seen several people mention it taking a few days to send messages. I've > usually seen mine in a few minutes. We'

Re: [asterisk-users] Google acquires Grand Central

GrandCentral isn't about hiding your number, it's about reachability. Grand Central gives you a single number that rings your home, office, cell, etc... And provides a single voicemail box for all of those numbers. As Asterisk users, these features do not seem very "ground breaking" to us, as most

Re: [asterisk-users] Monitor events?

Anthony Francis wrote: > There are no events generated when the monitor stops and starts, but > since you are implicitly recording in your dialplan one way or another > you can just add a userevent step before recording and after. You can also start monitoring through the Manager API in which ca

Re: [asterisk-users] Monitor events?

Hi Anthony, are you sure the monitor is started and sotoped via the dialplan ? Anthony Francis wrote: > Daniel Gradecak wrote: > >> Hi all, >> I would like to know if there is any possibility to send an event when a >> call is monitored? >> For both start and stop monitor. >> >> There is no e

Re: [asterisk-users] Google acquires Grand Central

I don't see the point of the service provided by GrandCentral. Party A calls party B through GrandCentral. Party B know party A's number and calls party A back, now party A can call party B directly, and party A has party B's directly number. -Original Message- From: [EMAIL PROTECTED] [ma

Re: [asterisk-users] Monitor events?

Daniel Gradecak wrote: > Hi all, > I would like to know if there is any possibility to send an event when a > call is monitored? > For both start and stop monitor. > > There is no event sent on asterisk 1.2 for that monitor case. I did not > find any changes regrding that on 1.4. Am I wrong? > Is

Re: [asterisk-users] OpenSer/Asterisk PBX solution

I have some clients using Enswitch (Paid solution). They are real happy with it. - Original Message - From: Bob Gibson To: asterisk-users@lists.digium.com Sent: Wednesday, June 27, 2007 7:37 PM Subject: [asterisk-users] OpenSer/Asterisk PBX solution We have been working a O

Re: [asterisk-users] Very bad TDMF tone !

- Original Message - From: "Eric "ManxPower" Wieling" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, July 09, 2007 4:40 PM Subject: Re: [asterisk-users] Very bad TDMF tone ! > lizhong zhu wrote: >> hello, all of asteriskers: >> i am usi

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

Olivier wrote: > Hi, > > My setup is : > PSTN - ISTP Network --- Router - Asterisk > -- SIP Phones > > Phones are located in the same location. > I'm thinking about installing new phones in other locations (small > agency, home workers), registering those phone

Re: [asterisk-users] Very bad TDMF tone !

lizhong zhu wrote: > hello, all of asteriskers: > i am using tdm400P in my office. i tested that TDMF generated by asterisk is > so bad. the sound is very soft and quality is so bad. i am using asterisk > 1.2.18. most of time, the # key can not be detected correctly. Does anyone > has that prob

Re: [asterisk-users] Need Advice/Suggestion

Or, if you can have a trigger of some type. If you have say, a database, that stores the current "night service" status, then you can query that to determine if you should send the call to the after hours steps, or to dial into the phone. Then set up another extension that the internal people c

[asterisk-users] Very bad TDMF tone !

hello, all of asteriskers: i am using tdm400P in my office. i tested that TDMF generated by asterisk is so bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18. most of time, the # key can not be detected correctly. Does anyone has that problem? please give me a hit fo

Re: [asterisk-users] Need Advice/Suggestion

> One of my client requested that he wants to > manually shift the dial > plan like above as he has flexiable timing sometime he finishes at 3:00pm > some > time 8pm. I can > not give him freepbx access. How about ignoring the time element completely and just telling the client to divert his/h

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

[asterisk-users] DTLS availablity?

Is DTLS available for Asterisk on any Linux distro? I am most interested in Centos ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.

[asterisk-users] Monitor events?

Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event whe

[asterisk-users] Background transfers with callback

Hello list, I have successfully set up Asterisk, but girls from our office complain to me that when they hit Flash to transfer a call and pick the number, they need to wait until the call is answered, and only then they could hangup. On the analog PBX we had before the transfer was in "backgroun

[asterisk-users] Monitor events?

Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event wh

Re: [asterisk-users] installing * from source

On Sun, Jul 08, 2007 at 05:58:18PM -0400, EdPimentl wrote: > Have you also consider adding adding the uBuntu steps in addition to CentOS? > -E Ubuntu steps, due to popular demand: apt-get install asterisk zaptel-source m-a a-i zaptel Untested yet. Should work on 7.04 . Bug reports are welcom

Re: [asterisk-users] Xorcom Bri and asterisk crashes

On Mon, Jul 09, 2007 at 04:02:06PM +1000, Nathan Dennis wrote: > Thanks for the input, but I still don't seem to have any luck with the > devices locking up. The trace you posted before mentioned tasklets. Those are not in use in the Astribank driver code (unless you set the optional parameter pc

Re: [asterisk-users] Fax and Asterisk

- "Lee Howard" <[EMAIL PROTECTED]> wrote: > Andrew Nowrot wrote: > > > I am trying to build reliable fax solution with asterisk, iaxmodem > and > > hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium > 3 > > 1.2 GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. Afte

[asterisk-users] CTI application controling Asteridk

Hi, I have two boxes : - A asterisk server. - A Python Server doing CTI and call control. If a call come on the Asterisk, a sound will be played continually Then, If somebody want to pick up this call, he will click on a Webpage (using the Python server) that will ask the Asterisk box to j