Re: [Asterisk-Users] Music On Hold troubleshooting
So the problem still persisits. What should I do? My musiconhold is not playing :) On 6/18/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: On Sun, Jun 18, 2006 at 11:50:12AM +0500, amna saleem wrote:> I have read in wiki pages that for astreisk 1.2.9.1 , you don`t have to> install this rpm package.> But Ialso read that Red hat Linux 9 and enterprise doesn`t suppport mp3> sound and song etc.> What are your views?? > Regards,> Amna SaleemUnless you need to stream mp3 music, playing an mp3 music file will be awaste of CPU.mp3 files are highly compressed, but need to be downsampled to 8khz mono(16 bit samples). If you'll convert the mp3 file to wav and downsample, chances are you'll end up with a comparable disk space and much lesswork for playing.--Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406[EMAIL PROTECTED] http://www.xorcom.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold troubleshooting
I have read in wiki pages that for astreisk 1.2.9.1 , you don`t have to install this rpm package. But Ialso read that Red hat Linux 9 and enterprise doesn`t suppport mp3 sound and song etc. What are your views?? Regards, Amna Saleem On 6/17/06, Sharon Lim <[EMAIL PROTECTED]> wrote: Did you install the sound packages such as mpg123-0.59r-1.i386.rpm ? Can download from http://rpm.pbone.net/index.php3/stat/4/idpl/516450/com/mpg123-0.59r-1.i386.rpm.html good luck! On 6/16/06, kharris <[EMAIL PROTECTED]> wrote: Can anyone point me in the direction for resources for troubleshootingno MusicOnHold with Asterisk version 1.2.9.1 and Asterisk Addons version1.2.3?ThanksKarl___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] delay in MeetMe
I am still getting delay. I have tried the q option.Did decrease the delay but not that much. Anyone having any idea why Regards, Amna Saleem On 6/14/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: The problem was fixed in 1.2.0amna saleem wrote:> No , actually I am using Asterisk-1.2.9.1 > I will try the q option though>> Thanks and regards,> Amna>>> On 6/14/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:>> >> I assume you are using 1.0.x. Add the "q" option to the Meetme>> extension. 1.0.x has a known issue where enter/exit sounds cause>> increasing delays.>>>> amna saleem wrote: >> > Hi All!>> >>> >>> >>> > I am facing some delay in conferencing.>> >>> > Using DIAX for Voip calls ,no hardware used yet >> >>> > I am using MeetMe to achieve conferencing and am having a lot of>> delays.>> >>> > Can anyone tell me how to reduce the delay>> >>> > >> >>> > Regards,>> >>> > Amna Saleem>> >>> >>> >>> >> >>> > ___>> > --Bandwidth and Colocation provided by Easynews.com -->> >>> > Asterisk-Users mailing list >> > To UNSUBSCRIBE or update options visit:>> >http://lists.digium.com/mailman/listinfo/asterisk-users>>>> >> -->> Now accepting new clients in Birmingham, Atlanta, Huntsville,>> Chattanooga, and Montgomery.>> ___>> --Bandwidth and Colocation provided by Easynews.com -->>>> Asterisk-Users mailing list>> To UNSUBSCRIBE or update options visit:>> http://lists.digium.com/mailman/listinfo/asterisk-users>>>>> >> ___ > --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kiax - iax2 softphone
Hi! I have used KIAX and have made calls from KIAX to KIAX and KIAX to DIAX soft phone which u can find on: http://www.laser.com/dante/diax/diax.html I didn`t get any MOH. Can you send me your MOH settings?musiconhold.conf also can you post me the sip.conf file which sip phone are you using? What is the asterisk version? Hope you will reply Regards, Amna Saleem On 14 Jun 2006 18:31:28 -, [EMAIL PROTECTED] < [EMAIL PROTECTED]> wrote: Has anyone on here used kiax before? I am asking because I have it installedon several computers and have been able to get it to connect and register to my Asterisk box. I can even call between them and my SIP softphones.The problem I am having is this: when I use kiax to call someone else, theyget some kind of background music playing while I am talking to them. I have called from kiax to kiax phone and get this on both ends but I cannot identifywhat it is or where it's playing from. I've watched the CLI when I make thecall and there is no indication of playing music on hold or background music. The dialplan doesn't even reference music on hold, it just dials whicheverphone I am trying to call.So, I am wondering if this is something whichI need to set in kiax and if anyone who has used kiax might be able to give me a hint on where I can disable this at. I would prefer to use an IAX softphonesince it only requires one port to be open into my firewall, but kiax is theonly one I've found for free. Once I get Asterisk up and running properly, I'll probably buy a couple of licenses for a good quality softphone, but untilthen, free is the best for me. :)So, any ideas?Undrhil___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delay in MeetMe
Hi! I am using Asterisk-1.2.9.1 Zaptel 1.2.6 And my system has Linux Kernal 2.4 Best Regards, Amna On 6/14/06, Tony Mountifield <[EMAIL PROTECTED]> wrote: In article <[EMAIL PROTECTED] >,amna saleem <[EMAIL PROTECTED]> wrote:> Hi All!>> I am facing some delay in conferencing.> Using DIAX for Voip calls ,no hardware used yet > I am using MeetMe to achieve conferencing and am having a lot of delays.> Can anyone tell me how to reduce the delayWhat version of Zaptel are you using, what version of Asterisk, andwhich Linux kernel does your system have? CheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] delay in MeetMe
No , actually I am using Asterisk-1.2.9.1 I will try the q option though Thanks and regards, Amna On 6/14/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: I assume you are using 1.0.x. Add the "q" option to the Meetmeextension. 1.0.x has a known issue where enter/exit sounds cause increasing delays.amna saleem wrote:> Hi All!>>>> I am facing some delay in conferencing.>> Using DIAX for Voip calls ,no hardware used yet>> I am using MeetMe to achieve conferencing and am having a lot of delays. >> Can anyone tell me how to reduce the delay>>>> Regards,>> Amna Saleem>>> >> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] delay in MeetMe
Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-1.2.9.1
i guess you were right. it was due to the previous version of asterisk on my PC,although i had make clean it anyway thanx for the help. can you tell me if i can use the same iax.conf and extensions.conf files that i used for asterisk-1.0.3 for this 1.2.9.1 version? thanx again On 6/11/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote: amna saleem wrote:> hi !> i have installed asterisk-1.2.9.1> but am unable to run it > i am getting this error>> "[pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325> __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined> symbol: ast_pthread_create > Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading> module pbx_wilcalu.so failed!"> can anyone help me>> i have redhat linux enterprise> zaptel version 1.2.6 > libpri version 1.2.3>> what am i missing here?I know it sounds daft, but could it be a module that was compiled aspart of a previous install of asterisk and wasn't overwritten/deletedand is being autoloaded (or directly loaded)? It's not a standard module (afaik) anymore, and certainly hasn't beencompiled with the versions I'm running.If it's for a particular card, you may need to recompile the moduleyourself from their driver source. If you don't know what it's used for, I'd move it out of/usr/lib/asterisk/modules, and see what happens/doesn't happen.Usually when you update asterisk, at the end of the make-install it willgive you a list of modules that are in the modules directory that it hasn't placed there, if it does that they are usually worth looking at.___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-1.2.9.1
hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error "[pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed!" can anyone help me i have redhat linux enterprise zaptel version 1.2.6 libpri version 1.2.3 what am i missing here? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a new asterisk version
Hi All, I need a suggestion. I want to run only IAX on two windows based PCs and asterisk Can you suggest which asterisk , libpri and zaptel versions should i use? do i need some other modules also? Also where will i find the guide to compile astreisk Actually i have installed,comnpiled and used astreisk-1.0.3 on Red hat 9 which was not that stable. Now i have Red hat Enterprise on my PC. i think there are newer stable versions which can run on Redhat Enterprise Linux. Kindly help, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SATA hard disk compatibility
Thanks alot for the help. I have not worked on fedra core .Which version should I use Also can you tell me that if I am using Red hat Enterprise, which asterisk version will be the best suited ? and will i be able to use the same .conf files which i used earlier with aserisk 1.0.3. I only need to use IAX ,and the IAX soft phones ,don`t really have to use SIP or H323. Also I want a stable asterisk version like 1.0.3 which doesn`t need to be upgraded continuously. I hope you will help me Regards, Amna On 4/27/06, Assaf Flatto <[EMAIL PROTECTED]> wrote: The Hardware support of SATA in RH9.0 is not fully integrated AFAIK , somoving to a SATA hard disk without an upgrade might not be the safest bet. on the other hand until you try you won't know for sure .have you thought of using the Fedora Core ? those have SATA support andthey should be the closest thing to RH9 you can find.why don't you want to upgrade the asterisk ? 1.0.3 is a very old versionand many fixes and features where added to the software .Assafamna saleem wrote:> Hi!> I have been using ASterisk 1.0.3 on Red hat Linux 9.0 for a long time > now on my Home PC.> I want to shift to a PC having SATA hard disk .Can I install Redhat> 9.0 on SATA hard disk ??some people are telling me that I have to go> for Linux Enterprise 4.0.I don`t want to leave Linux 9.0 because I> want to run Asterisk 1.0.3>> Can anyone help me??> Amna> >> ___ > --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>--Assaf FlattoAtelis IT ManagerCellular: +972-54-5679230e-mail: [EMAIL PROTECTED]___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SATA hard disk compatibility
Hi! I have been using ASterisk 1.0.3 on Red hat Linux 9.0 for a long time now on my Home PC. I want to shift to a PC having SATA hard disk .Can I install Redhat 9.0 on SATA hard disk ??some people are telling me that I have to go for Linux Enterprise 4.0.I don`t want to leave Linux 9.0 because I want to run Asterisk 1.0.3 Can anyone help me?? Amna ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WIP300 WiFi Phone
umm.. Can you please tell me what phone u r talking about??i mean does it support IAX. Actually i am sick and tired of my DIAX and want a new IAX phone... I am using an older version of * like 1.0.3 I hope u will not mind replying to me On 2/26/06, Me <[EMAIL PROTECTED]> wrote: How is the voice quality?> I've just plugged mine back into the charger after having used it > nearly all day. I didn't have any of the problems you've described.> Sorry you're having such bad luck with it. I'm not certain what the> phones are rated to do, but I probably got better than 3 hours talk > time on it today which is definitely the best I've gotten with any of> the WiFi phones up to this point.>> BJ>> --> Bird's The Word Technologies, Inc.> http://www.btwtech.com/> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users>___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible ?
Hi , I think i understand what you mean by your mail.I have done the same thing. You must download following modules from and asterisk site e.g. www.digium.com 1.asterisk-1.0.3.tar 2.libpri-1.0.3.tar 3.zaptel-1.0.3.tar Then there is a process you need to follow which you will find in th read me file of asterisk after you untar these. If you are unable to figure it out then mail me. Allah hafiz Amna On 1/24/06, Sohail Arham <[EMAIL PROTECTED]> wrote: Hi everyone, I am a new one for that listsactually i have final year project on VOIP & IMS ...so i want to install asterisk on my pc ...IS it possble that ...we can call on small LAN network without buying any card...i will clear my point as that...suppose i have a linux machine on which i want to install asterisk and HOW it will install...and second point is that ..i have 2 windows clients machine i want these two machice can calls each other thorough asterisk .rest of all will be think latter thanksss -- Muhammad Sohail ArhamU.E.T. LahorePhone No. 0321-4422406 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] diax not working properly
Thanx for your help. I will do some home work and then get back to you. Do you think the problem is of the asterisk version?? main issue: * after running for some time the phone logs out (gets out of registeration). Regards Amna On 12/6/05, Dan <[EMAIL PROTECTED]> wrote: Hi,>I really like DIAX and i was to stick to it so if you can help solve>my>problem with diax??? I'm not sure that the problem is DIAX relatedYou are the first one with this issue...Pls try to use the examples in DIAX help file for iax.conf andextensions.confBest regards,Dan ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] diax not working properly
Thank you gentlemen for your prompt replies and help i really appreciate that. Actually i have only used asterisk-1.0.3 :) I will definitely visit these webpages and see if they help. I really like DIAX and i was to stick to it so if you can help solve my problem with diax??? Thanx Amna On 12/5/05, Time Bandit <[EMAIL PROTECTED]> wrote: > Hi!> I have been using Asterisk-1.0.3 for quite some time now.My main aim> nowadays is to make iax-iax calls for which i am usin DIAX soft phone.The> problem is that sometimes the phone doesn`t register and at others it gets> out of the registration(after being registere for some time).Can anyone tell> me what can be the problem ,what other iax phones are available ? I don't think your problem is DIAX, Dan is making a great phone and hetest it carefully. But anyway, since you asked, here is a short list :- MediaX (my own) : http://www.marccharbonneau.com/asterisk/mediaxphone.php- Idefix : http://www.asteriskguru.com/tools/idefisk_beta.php- IAX phone : used to be at this address : http://www.sokol-associates.com/IaxPhone.htm but the site changed andI lost track of it- MozIAX : plugin for Firefox/Mozilla : http://moziax.mozdev.org/- iaxComm : http://iaxclient.sourceforge.net/iaxcomm/hth>> Thanx and Regards,> Amna> ___ > --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>> http://lists.digium.com/mailman/listinfo/asterisk-users>>>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] diax not working properly
Hi! I have been using Asterisk-1.0.3 for quite some time now.My main aim nowadays is to make iax-iax calls for which i am usin DIAX soft phone.The problem is that sometimes the phone doesn`t register and at others it gets out of the registration(after being registere for some time).Can anyone tell me what can be the problem ,what other iax phones are available ? Thanx and Regards, Amna ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX not working properly
Hi! I am facing some problems with my asterisk-1.0.3.I am using DIAX phones as clients ,but sometimes they donot register with the asterisk server.Also if I don`t restart my asterisk frequently the registration of DIAX phones expires. Anyone who can help me please reply Regards, Amna ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help on linux version
Hi! This is amna saleem.I needed to ask if asterisk-1.0.3 can run on linux enterprise edition(latest version 4)??? I have been using asterisk-1.0.3 with redhat 9 but need to change the linux edition ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need urgent help
hi! I have already memtioned the error i am getting on my cli ie"flexible rate not heavily tested". I am also getting a warning ,something like "broken -pipe" I need urgent help plz reply ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need help
hi! I wanted to ask if someone ever got the error "flexible rate not heavily tested" I am not able to dial from PSTN to iaxphones(on which agents are logged in)...I have been successfully running this for some time now..but today all of a sudden i got this error and I can`t get connected to the agent... Can anyone help,. plz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need help
hi! I wanted to ask if someone ever got the error "flexible rate not heavily tested" I am not able to dial from PSTN to iaxphones(on which agents are logged in)...I have been successfully running this for some time now..but today all of a sudden i got this error and I can`t get connected to the agent... Can anyone help,. plz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: iaxcomm
No actually i have successfully installed (from scratch) and been using asterisk for more than 4 months now...i have been using diax phone ...but i came across this iaxcomm & just thought about transfering a calljust playing around ..but i can`t really get it working ... maybe i am not getting the one hint can u help thanx On 4/13/05, amna saleem <[EMAIL PROTECTED]> wrote: > Hi! > I was using iaxcomm but due to some reason am not able to transfer > calls to some other extensionwhat maybe the problem > do i have to make some changes to my extensions.conf??or iax.conf to > be able to transfer calls > Thanks > Amna > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxcomm
Hi! I was using iaxcomm but due to some reason am not able to transfer calls to some other extensionwhat maybe the problem do i have to make some changes to my extensions.conf??or iax.conf to be able to transfer calls Thanks Amna ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] attension mark spencer
hi , I was wondering if i can get some algo or architecture of asterisk...i mean how different channels are working (specially agents,h323)and how call is established... i know i am sounding a bit stupid but i need this ...can you please guide me thanx Amna Saleem ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] invalid extension (need help)
hi! I was wondering if the "i" extension works ,i mean i have included this in my extensions.conf ie exten => i,1,Answer exten => i,2,Playback(pbx-invalid) exten => i,3,Hangup but it doesn`t seem to work,i am getting no announcement when i dial an invalid no. rather i get the invalid tone (which we usually get on our analog phones at home) can someone help??? Thanx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] invalid extension(need help)
hi! I was wondering if the "i" extension works ,i mean i have included this in my extensions.conf ie exten => i,1,Answer exten => i,2,Playback(pbx-invalid) exten => i,3,Hangup but it doesn`t seem to work,i am getting no announcement when i dial an invalid no. rather i get the invalid tone (which we usually get on our analog phones at home) can someone help??? Thanx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX phones
hi all! I am running DIAX 098c,but it closes after some timeI have tried downloading it again and from different sites but the problem persists...can anyone tell me if there is a better IAX phone available for asterisk...plz reply Amna ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing asterisk from open phone
hi ! I have compiled asterisk-oh323 successfully, is there any way that the open phone can register with asterisk??? i need my open phone to dial to asterisk ..but it gives the message "no phone running on IP $192.168.19.206". I want to call asterisk from open phone and then direct calls to iax phones or other IP addresses using asterisk. Can anyone help me in this regard Thanks Amna Saleem ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk version 0.7.1
dear list, I am using asterisk version 0.7.1,and asterisk-oh323 version 0.5.10.Both are installed successfully,and I can see the configuration file in asterisk directory.But when I changed the file h323.conf ,asterisk doesnot start..it gives the error booting.asterisk:relocation error: /usr/lib/asterisk/modules/res_features.so: undefined symbol ast_pthread_create please can anyone tell me what is the problem ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_oh323.o
have asterisk version 1.0.0,have comipled pwlib version 1.5.2 ;and openh323 version 1.12.2.When I try to build asterisk-oh323 version 0.5.9 or 0.5.10 ,I get the following error : make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 I thought that there might be some linking problem,so I searched '/' for the file chan_oh323.o but didn`t find it anywhere. NOTE: The versions that are used are according to the readme files. You might have answered this before but can you please tell me where I am going wrong. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: chan_oh323.o
I have asterisk version 1.0.0,have comipled pwlib version 1.5.2 ;and openh323 version 1.12.2.When I try to build asterisk-oh323 version 0.5.9 or 0.5.10 ,I get the following error : make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323-0.5.10/asterisk-driver' make: *** [subdirs_all] Error 1 I thought that there might be some linking problem,so I searched '/' for the file chan_oh323.o but didn`t find it anywhere. NOTE: The versions that are used are according to the readme files. You might have answered this before but can you please tell me where I am going wrong. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_oh323.o
> > I have asterisk version 1.0.0,have comipled pwlib version 1.5.2 ;and > openh323 version 1.12.2.When I try to build asterisk-oh323 version > 0.5.9 or 0.5.10 ,I get the following error : > > make[1]: *** [chan_oh323.o] Error 1 > make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' > make: *** [subdirs_all] Error 1 > > I thought that there might be some linking problem,so I searched '/' > for the file chan_oh323.o but didn`t find it anywhere. > > NOTE: The versions that are used are according to the readme files. > You might have answered this before but can you please tell me where I > am going wrong. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_oh323.o
I have asterisk version 1.0.0,have comipled pwlib version 1.5.2 ;and openh323 version 1.12.2.When I try to build asterisk-oh323 version 0.5.9 or 0.5.10 ,I get the following error : make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 I thought that there might be some linking problem,so I searched '/' for the file chan_oh323.o but didn`t find it anywhere. NOTE: The versions that are used are according to the readme files. You might have answered this before but can you please tell me where I am going wrong. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_oh323.o
I have asterisk version 1.0.0,have comipled pwlib version 1.5.2 ;and openh323 version 1.12.2.When I try to build asterisk-oh323 version 0.5.9 or 0.5.10 ,I get the following error : make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 I thought that there might be some linking problem,so I searched '/' for the file chan_oh323.o but didn`t find it anywhere. NOTE: The versions that are used are according to the readme files. You might have answered this before but can you please tell me where I am going wrong. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk 1.0.1
Hi I want to download asterisk v1.0.1,can anybody tell me where can i find this version with zaptel ,zapata ,libpri etc thanx in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: changing configuration file
hi! Thanx alot for your help,I am now able to place a call from one iax phone to another,it was really nice of you to help me out. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: changing configuration file
-- Forwarded message -- From: amna saleem <[EMAIL PROTECTED]> Date: Thu, 18 Nov 2004 22:26:11 -0800 Subject: changing configuration file To: [EMAIL PROTECTED] hi! I am a beginner at Asterisk and Linux,I am trying to place a call using IAX ,but don`t really know how to chaneg the configuration file.I open the /etc/asterisks directory ,then open the iax.conf file from there but can`t edit it .Can anyone please help me reagarding this issue.How can a configuration file be changed or edited Does the same apply if I want to change the dial plan as well? Amna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 linking with asterisk
Hi! i have to make pabx to direct calls to h323 terminals. i have an h323 gateway available and wish to use asterisk as the gatekeeper for call direction and queueing etc.I am a beginner at asterisk and to link openh323 with asterisk for my project i searched on net i found different compilation instructions from different sources. having no idea i followed two sources and issued commands as below in the ~/pwlib dir i gave the following commands ./configure make bothdepend make bothnoshared then the same in the ~/h323 dir then bak in the ~/pwlib dir i issued the command make clean opt and then the same in ~/h323 dir then in ~/asterisk/channels/h323 dir i issued the 'make' command. and in the ~/astersik dir i issued 'make install' followed by 'make samples' can u plz guide me on the linking and compilation procedure of h323. how will i know that h323 has been linked. will there appear an h323.conf kinda file?If you don`t have much time with you can you plz tell somebody else working with you to help me. plz do reply ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linking H323 with Asterisk
Hi! i have to make pabx to direct calls to h323 terminals. i have an h323 gateway available and wish to use asterisk as the gatekeeper for call direction and queueing etc.I am a beginner at asterisk and to link openh323 with asterisk for my project i searched on net i found different compilation instructions from different sources. having no idea i followed two sources and issued commands as below in the ~/pwlib dir i gave the following commands ./configure make bothdepend make bothnoshared then the same in the ~/h323 dir then bak in the ~/pwlib dir i issued the command make clean opt and then the same in ~/h323 dir then in ~/asterisk/channels/h323 dir i issued the 'make' command. and in the ~/astersik dir i issued 'make install' followed by 'make samples' can u plz guide me on the linking and compilation procedure of h323. how will i know that h323 has been linked. will there appear an h323.conf kinda file?If you don`t have much time with you can you plz tell somebody else working with you to help me. plz do reply ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] changing configuration file
hi! I am a beginner at Asterisk and Linux,I am trying to place a call using IAX ,but don`t really know how to chaneg the configuration file.I open the /etc/asterisks directory ,then open the iax.conf file from there but can`t edit it .Can anyone please help me reagarding this issue.How can a configuration file be changed or edited Does the same apply if I want to change the dial plan as well? Amna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
