[Asterisk-Users] weird zttest result

in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% Anyone has any idea why this happens? Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com

Re: [Asterisk-Users] weird zttest result

Is this result indicates no problem at all? 8192 samples in 27554 sample intervals -136.352539% Regards, Stevanus C F wrote: These are actulay not strange, but good results. On 1/23/06, stevanus [EMAIL PROTECTED] wrote: Hi, I have these strange results : 8192 samples in 8192

[Asterisk-Users] jitterbuffer causes no sound?

this beast again :)... Currently, I'm rollback to asterisk 1.0.10 to avoid any unprecedented issue... Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] asterisk 1.2.3 call problem

configuration change issue in 1.2.3 cause I've just used my configuration that worked in asterisk1.2.2 ? Please shed me some light, thank you.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

Re: [Asterisk-Users] asterisk 1.2.3 call problem

=g723.1 ;allow=g729 ;allow=ulaw ;allow=alaw ;allow=gsm mailboxdetail=yes #include iax_additional.conf #include iax_custom.conf Regards, Stevanus stevanus wrote: Hi, I've tried to upgrade my asterisk to 1.2.3 again after disastrous bug incident yesterday but when I called and the phone was picked

[Asterisk-Users] intel 536 EP as x100p clone?

wiki and there I found that people are success using Intel 537EP as x100p clone. While mine is merely Intel 536EP, but I think both are modems made by Intel.. Maybe there's a way making it function like x100p too like someone in asterisk channel (irc) told me :).. Thanks.. Regards, Stevanus

[Asterisk-Users] intel 536 ep as fxo - possible?

not detect it. I just need more information before I throw this intel 536 EP to the garbage can :P. Any information would be appreciated.. Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

[Asterisk-Users] delay problem

between 1-10 calls. Is my Duron overwhelmed by the load? The delay exists in queue, local sip-to-sip call, and zap-to-sip call. It's so annoying :( Anyone has a solution or maybe some clue for me? Totally clueless here... Thanks... Regards, Stevanus

[Asterisk-Users] sccp help

issues? Any suggestion will be greatly appreciated.. Thanks Best Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] sccp help

Hi, I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp. Is it the same as chan_sccp from chan-sccp.berlios.de? Best Regards, Stevanus Stefan Gofferje wrote: Hi, On 9:04:57 August 19, 2005 stevanus [EMAIL PROTECTED] wrote: Hi, I tried to connect cisco 7910

Re: [Asterisk-Users] sccp help

Hi, Haven't noticed that there exists one :P Thanks for the pointer anyway ;). Gotta sign up pretty soon :) Best Regards, Stevanus Stefan Gofferje wrote: On 10:10:54 August 19, 2005 stevanus [EMAIL PROTECTED] wrote: Hi, I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp

[Asterisk-Users] tdm04b hangup problem

that this is caused by improper setting in rxgain or txgain? Currently, I set rxgain = 15.0 and txgain = 5.0.. Thanks.. Best Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo

[Asterisk-Users] ivr not working?

stand there about 7 seconds, playing no sound at all and then hung up.. I use AMP version 1.10.007a.. Has anyone known the solution for my problem? Any help would be appreciated a lot.. Thanks Best regards, Stevanus ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Phone always busy after caller hangup

Hi, If you use digium card, then maybe you set wrong signaling on fxs... Best regards, Stevanus Tim P wrote: I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one and answering it and then hanging up results

[Asterisk-Users] bypass incoming ring..is it possible?

Hi, Is it possible to bypass incoming ring on asterisk so that incoming calls come to asterisk box will be directed straight into did? Is anyone able to give me any clues or pinpoint me where I can get more information about it? Thanks for your attention.. Best regards, Stevanus

Re: [Asterisk-Users] bypass incoming ring..is it possible?

it will take some weeks, perhaps months to learn the source codes Perhaps I should take a C course as well :P Anyone can help me? Best regards, Stevanus Alexander Ilyushin wrote: You can first answer to call, and then provide playtones(ring) to caller. 2005/6/8, stevanus [EMAIL PROTECTED]: Hi

Re: [Asterisk-Users] ivr not working?

Finally I've got the ivr to work.. The workaround I've found so far is record and edit the voice file through Adobe Audition or cool edit or sound recorder, etc and then convert it to gsm using sox.. Hope that might help someone ;) Best regards, Stevanus stevanus wrote: Hi, Recently

[Asterisk-Users] tdm04b slow response

Hi, After days tinkering with this digium card (TDM04B), I notice that this card has a slow response in detecting ring signal from pstn and hanging up when the call is over. The delay can consume up to several seconds... Is this normal? Best regards, Stevanus

Re: [Asterisk-Users] Thank you for the timely suggestion

Hi, try xlite if you have enough bandwitdh for G711 codec requirement.. try firefly if you want to use G729 codec freely (linked via dll).. both of them are the best freeware softphone for windows. Best regards, Stevanus infra struct wrote: I have been searching for the necessary

Re: [Asterisk-Users] bypass incoming ring..is it possible?

Hi, I've tried your suggestion but the result is still the same... Have another suggestion? Best regards, Stevanus Wilson Pickett wrote: Is it possible to bypass incoming ring on asterisk so that incoming calls come to asterisk box will be directed straight into did

Re: [Asterisk-Users] Softphone for Linux desktops

Hi, try xlite, it has linux version.. Best regards, Stevanus Eric Bishop wrote: Hi all, We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating

Re: [Asterisk-Users] G.729AB codec support

cannot afford it. [In my country, there are still many IT professionals that is paid under $200 per month :( ] Best regards, Stevanus Zoa wrote: Im saying that the code is only an implementation of g729. The intel sources clearly states that you need a license for g729, not from intel

Re: [Asterisk-Users] How to allow multiple codecs in A@H

hi, just put those lines (allow=bla) in peer details box in AMP GUI. at section "add sip trunk". best regards, stevanus Erdem HAK wrote: I wonder how to allow more then one codec in AMP ([EMAIL PROTECTED]) GUI? For example I want to configure like this

[Asterisk-Users] zeroconf help

writing audio data: : Broken pipe it's weird since I've double checked the library and header from zeroconf and it seems that everything has been in the right place.. Is there anyone can help me? Well, it seems I hit another dead end this time... Best regards, Stevanus

Re: [Asterisk-Users] cisco 7940 + sccp issue

y in order to make my 7940 talk with asterisk using sccp. I had bad experiences in flashing devices therefore I want to avoid this as much as possible :). Best regards, Stevanus Stefan Gofferje wrote: Hi, On 9:20:51 July 06, 2005 stevanus [EMAIL PROTECTED] wrote: I've set the con

Re: [Asterisk-Users] No sound

Hi, That would probably be a problem with nat. Just read this on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+NAT+solutions Best regards, Stevanus Ronald Wiplinger wrote: I have an asterisk box installed, but all connections to outside of the private network do

[Asterisk-Users] asterisk hang

asterisk 1.2.5, zaptel 1.2.4 and 3 TDM04B on my machine.. Is it possibly caused by interrupt sharing? I admit I get some interrupt sharing problems and cannot be solved right now because of lack of funds :(.. Thanks Regards, Stevanus ___ --Bandwidth

Re: [Asterisk-Users] asterisk hang

when asterisk is hung :(. I could even connect to the CLI using asterisk -r as if asterisk worked properly. But when asterisk is in this state, sometimes I cannot make outgoing call (not even sip-to-sip call) and sometimes I lose the dial tone at all.. Pretty weird.. Regards, Stevanus Tzafrir

[Asterisk-Users] Avoiding initial deadlock on iax?

this http://bugs.digium.com/view.php?id=4045 , but from the link I read that it is just for H323 not for iax. Will that patch cure my asterisk problem since the symptom are the same? Anyone has any ideas? Thanks Regards, Stevanus ___ --Bandwidth

Re: [Asterisk-Users] Codec Problem

It seems that your asterisk cannot transcoding from ulaw to g729 and vice versa. What is the output from 'show translation' ? Did you allow both codecs in sip.conf ? Regards, Stevanus Il Neofita wrote: I have the license for G729, however I need to use a different codec for the prepaid

[Asterisk-Users] multiple asterisk process ?

-c asterisk 5126 0.0 5.1 25932 12240 ? S13:57 0:00 asterisk -vvvg -c asterisk 6487 0.0 5.1 26016 12336 ? S14:23 0:00 asterisk -vvvg -c Is this normal? Does anyone experience this? Thanks.. Regards, Stevanus

Re: [Asterisk-Users] multiple asterisk process ?

that haven't been solved like "avoiding deadlock on iax" problem which I had mentioned before.. Unfortunately, I don't know how to recreate the problem so all I can do if the problem is happened just do some killall - 9 asterisk :(... Regads, Stevanus Moises Silva wrote: Thanks for

Re: [Asterisk-Users] multiple asterisk process ?

I've tried cat /proc/*asterisk proc number*/environ | strings | grep LD_ASSUME_KERNEL and it returns nothing..:( And just for confirmation : I had the same problem as Lee had (unable to make calls out) :( Regards, Stevanus Lee Archer wrote: Yes it is a problem cos after a while of just

Re: [Asterisk-Users] Sangoma Card Question

will be enough).. Regards, Stevanus Matt wrote: By the system you mean the phone company? Or asterisk? So what you are saying is I hang up... the sangoma hangups... but the phone company sees it as a flash... then says.. HEY DUDE! YOU JUST HUNG UP ON YOUR CALLER.. Here they are back

[Asterisk-Users] asterisk hung again

returned -1: Network is unreachable These messages above exist for each devices that are registered to asterisk server. After I restarted the asterisk server, the problem was gone. What did caused this? I'm running asterisk 1.2.7.1 on Redhat EL4 Regards, Stevanus

[Asterisk-Users] iax2 disconnect problem

] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL Do I have to post this on mantis? Is this a bug? Anyone can confirm this? Thanks.. Regards, Stevanus

Re: [Asterisk-Users] ip phone

googling it. Don't ask me as I don't know any information about it. Good luck.. Regards, Stevanus trixter aka Bret McDanel wrote: looking for ip phones for an office setting. The client wants about 15 phones initially. Not counting volume discounts, does anyone have any recommendations

Re: [Asterisk-Users] tdm04b hangup problem

Any thought anyone? stevanus wrote: Hi, I have severe problem here.. My asterisk server use tdm04b from digium and is often incapable of detecting hangup signal. It is happened occasionally in incoming call so I have to watch fop all the time and hangup the channel manually

Re: [Asterisk-Users] tdm04b hangup problem

nobody like to share any comments? Just curious :P Best Regards, Stevanus stevanus wrote: Any thought anyone? stevanus wrote: Hi, I have severe problem here.. My asterisk server use tdm04b from digium and is often incapable of detecting hangup signal. It is happened occasionally in incoming

Re: [Asterisk-Users] tdm04b hangup problem

-PIC i8042 14: 163276 XT-PIC ide0 15: 997605 XT-PIC ide1 NMI: 0 ERR: 0 Best Regards, Stevanus stevanus wrote: Hi, Yesterday, the asterisk machine was crash :S. But after the crash, it seems the previous problems were eliminated. I will notice

Re: [Asterisk-Users] TDM400P not detecting hangup and not hanging up.

. (I've asked this in another thread, but got no respon :( ) Best Regards, Stevanus canuck15 wrote: This may or may not be related but have you tried adjusting your RX and TX gains? I see both are at the default (0.0) which leads me to believe you have not. Search the Asterisk Wiki

[Asterisk-Users] asterisk frequently dead

Asterisk. Anyone has any idea of the cause? Thanks.. Best Regards, Stevanus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] asterisk frequently dead

in process of learning it...Gotta be careful cause the system is used by more than 10 person (well, I'm getting tired of apologizing anyway :P) Thanks, Best Regards, Stevanus Andrew Kohlsmith wrote: On Wednesday 07 September 2005 22:56, stevanus wrote: My asterisk is frequently dead

Re: [Asterisk-Users] asterisk frequently dead

() from /lib/ld-linux.so.2 (gdb) ... Actually the information giving by gdb is far more detail..Just keep it brief here to keep the space small. If anyone want to help me, then I'll send it entirely.. Any comments/thoughts will be greatly appreciated. Thanks, Best regards, Stevanus

Re: [Asterisk-Users] asterisk frequently dead

Hi, Thanks for the reply.. But the biggest problem here is that people using asterisk will get dissapointed, sometimes mad because their call being dropped off when asterisk is dead.. Any suggestions anyone? Regards, Stevanus Moises Silva wrote: this is not a solution, more a workaround

Re: [Asterisk-Users] Outgoing busy

=Asterisk+config+extensions.conf Regards, Stevanus Anders Svensson wrote: I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se

Re: [Asterisk-Users] Outgoing busy

Hi, Yeah, you're right Olle. The connection to another sip provider is set in sip.conf. I thought Anders was trying to make outgoing call to pstn. Too much tinkering with Digium card make me think that :P. Regards, Stevanus Olle E. Johansson wrote: stevanus wrote: Hi

[Asterisk-Users] 1.0.10 to 1.2.1 upgrade..is it worth it?

story with asterisk 1.2.1 please share :) Thank you... Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Asterisk 1.2.2 Released!

Just curious, will this version be supported by AMP 1.10.010? Anyway, I am going to upgrade mine in saturday..:) Joe Pukepail wrote: Perhaps I'm an idiot, but I looked through the readme and changelog but can't figure out what asterisk-netsec is all about? Anybody figure it out? On

[Asterisk-Users] speex in asterisk 1.0.10

Hi, Does anyone know how to configure speex in asterisk 1.0.10? I've successfully installed it but cannot get any idea how to set the quality, etc.. Thanks Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk

Re: [Asterisk-Users] speex in asterisk 1.0.10

Yeah, Paul. I guess you're right.. Just tested speex and got complains from my customer :S..Maybe this codec is not suited for our network ;).. Regards, Stevanus [EMAIL PROTECTED] wrote: Quick question - what is the point of speex? Do we really need it as an option? PaulH