RE: [Asterisk-Users] Softphone Audio problem
-Original Message- From: Michael Van Donselaar Sent: Thursday, May 20, 2004 6:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Softphone Audio problem On Thu, 20 May 2004 14:18:16 +0100, Andy Farnsworth [EMAIL PROTECTED] wrote: As a test, I was trying to use Iaxcomm and Iaxphone to connect to Asterisk and dial out to my other line. Using either of these soft phones, I can connect to Asterisk and listed to audio just fine. I can even connect across the net to another asterisk server and hear audio just fine, however, when I dial out to my second land line the audio that is transmitted is horribly broken up. It is as if the audio stream is broken into 8 parts every second and then every other part is dropped. I then tried the asterisk echo test and got the same thing. I am running Asterisk under RH9 on an AMD 2600+ 512 Mb RAM Desktop and the soft phones on my laptop running Windows XP (Laptop is Sony Vaio PCG-GRT815E, 2.8 Ghz processor, 512Mb Ram). Is this an asterisk problem or a soft phone problem? If asterisk, any ideas on how to fix it? What kind of PSTN interfaces are you using? I'm not sure from the description: are you seeing the problem of both lines, or only the second line? Do you get the same kind of results when using a SIP softphone? I'll be posting new binaries to sourceforge this weekend, because there have been some library changes related to jitter, but I haven't heard or seen anything as drastic as you describe. BTW what version of asterisk? - End Original Message- Looking back at this, I realized I gave very little information. I am using an X101P card to connect to the PSTN. I have tried connecting asterisk to my land line and calling to/from my cell phone and 2nd land line, changed to the second land line and called to/from my cell phone and my 1st land line. I have not tried one of the SIP softphones yet but will do so this weekend. I downloaded and compiled the latest asterisk from cvs two weeks ago and have done no updates since. I will check the exact version when I get home. I suspect it is a problem with my laptop as opposed to the software as I have found that it has started to ding repeatedly when connecting to the IAXPhone test line. This dinging happens on my end and goes away as soon as I hit the mute button. It does not seem to be sent out across the network as if I go to the echo test and then unmute and remute there I do not hear any dings come back and usually there is at least a slight delay in the echo. I'll keep trying to resolve this and will try it on my desktop machine (booted from RH9 in to Win2k) to try it out. I'll also try my wife's laptop and the old laptop to see if they do the same thing. Andy Farnsworth ___ 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] Softphone Audio problem
As a test, I was trying to use Iaxcomm and Iaxphone to connect to Asterisk and dial out to my other line. Using either of these soft phones, I can connect to Asterisk and listed to audio just fine. I can even connect across the net to another asterisk server and hear audio just fine, however, when I dial out to my second land line the audio that is transmitted is horribly broken up. It is as if the audio stream is broken into 8 parts every second and then every other part is dropped. I then tried the asterisk echo test and got the same thing. I am running Asterisk under RH9 on an AMD 2600+ 512 Mb RAM Desktop and the soft phones on my laptop running Windows XP (Laptop is Sony Vaio PCG-GRT815E, 2.8 Ghz processor, 512Mb Ram). Is this an asterisk problem or a soft phone problem? If asterisk, any ideas on how to fix it? Thanks, Andy Farnsworth ___ 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] x100p / Answer- Flash - Dial
Title: Message Dan, This will probably not work. Once Asterisk tells the Panasonic PBX to transfer the call, the call will no longer go through the extension the Asterisk PBX is attached to. It seems the only solution to this (doing it the way you are) is to have two zaptel cards in the Asterisk PBX, attache them to two extensions of the Panasonic PBX and the calls come in one extension and then are sent out the other KEEPING BOTH BUSY as long as the call is in progress. As Sam said, you will really need to have an intelligent connection to the Panasonic PBX to do this on much more than a single call at a time basis. Disclaimer: I am an Asterisk Newbie, though I have been following it for several years, my hands on experience is abouta week old. Andy Farnsworth -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam BingnerSent: Saturday, May 08, 2004 10:54 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] x100p / Answer- Flash - Dial Even if you could get that to work properly, which I dont know... the callprogress detection is horrible; if you want to do that reliably you need a T1,ISDN or IPinterface to the switch (something that actually provides proper call progress) Sam -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan FernandezSent: Saturday, May 08, 2004 11:44 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] x100p / Answer- Flash - Dial I have an X100P connected to an extension of aPanasonic PBX.When a call from the PSTN comes in,it is routed directly to theextension where the x100p is .I want* to answer the call, play amessage and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then todialan IAX channel. The problem is that when I do a Flash on thezap channel, and then try to dial a new extensionvia that zap channel I get the following error "can't createzap channel". If I do a SendDTMF()thecalldoes get transfer to the new extension but then * gets out of the callloop and don't know it is answered or not by the new extension. AmI missing something? Why am I getting the "can't creatza channel" Thanksin advance. Dan
