similar to: IAX clients

Hi, I try to use ISDN4Linux drivers with Asterisk. In modem.conf i put /dev/ttyIO. Everything is OK when i lauch asterisk but, when i call Asterisk nothing happen. Someone can help me ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030407/13e78d8e/attachment.htm

Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030804/af4203ca/attachment.htm

Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 00001 [192.168.1.22:4569] Tx-Frame Retry[000] --

hi everybody, Is there SIP client which work with Asterisk and can be embedded in a HTML page ? Thanks Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/226a8b1b/attachment.htm

hi, how can I add or remove this line "include => context" by the command CLI ? regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030827/979ddd76/attachment.htm

Hi, I try to use conferences with Asterisk but i'm not succeed in doing this. I have set in meetme.conf : conf => 4000 and add an extension in extension.conf. But i have Asterisk WARNING Unable to open pseudo channel. Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, Someone know how to install conference (MeetMe) without a zaptel devices ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030414/f8a9140a/attachment.htm

Hi, I have a little question, I use asterisk with Netmeeting client. When I call netmeeting client with a phone. I don\'t have his ID in netmeeting window i have something like : ???;..dhz instead of 28. Someone know a way to display this ID ? Thanks you so much Rattana

hi, Does anybody knows how to set variables which by asterisk command CLI ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030513/59fc6610/attachment.htm

Hi, In the file Master.csv (var/log/asterisk/cdr-csv) we have stats of call. But Call from H323 client doesn't here. What sould have do in order to have this. Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030804/ed74cc69/attachment.htm

Hi, I try to use asterisk with PHP. But I can't execute asterisk -r when i am not in root. But I have lauch asterisk in an users mode. I have this message when i try this : ERROR[1024]: File asterisk.c, Line 1222 (main): Unable to connect to remote asterisk Someone konow how to use asterisk in user mode ? Regards Rattana -------------- next part -------------- An HTML attachment was

Hi, Does anyone use EICON Diva Server Card ISDN and CAPI ? I need some help for drivers installation. I use RED HAT 8.0 (kernel 2.4.18-14) I want to know which drivers should I use and how can I install it. Thanks Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a compilation errors with chan_capi 0.1.0. I have : chan_capi.c:33:20 capi20.h: No such file or directory and a lot of errors with this error. The capi20.h is in /usr/srx/linux-2.4.18-14/drivers/isdn/eicon/capi20.h. So I don't know where the problem. Someone have a solution ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI, Anyone knows the current status of JIAXclient? I tried to recompile the sources available in sourceforge but they reference a old java package that I was not able to find. I tried to e-mail the author but seems that his account is no longer valid. I in need of a java IAX client that could be loaded as an applet. I know that is a lot of viable SIP alternatives, but due to NAT/Firewall

Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I have tried JIAXClient, but it allows people to call anywhere, and what I want is just a configurable

Not sure if this is the best place but does any one have any used IAXy's they are interested in selling? I am looking to pick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at asterisk@matraex.com Thanks Michael

I am wondering how to interpret the parameter estimates that lm() reports in this sort of situation: y = round(rnorm(n=24,mean=5,sd=2),2) A = gl(3,2,24,labels=c("one","two","three")) B = gl(4,6,24,labels=c("i","ii","iii","iv")) # Make both observations for A=1, B=4 missing y[19] = NA y[20] = NA data.frame(y,A,B) nonadd = lm(y ~

We build a Voip client with Iaxclient(Jiaxclient+Iaxclient2.1beta1+Theora1.0+libvidcap0.21),when we established video call with resolution 640X480(bitrate:500000,frameRate:11,fragsize:1472) in both sides,the CPU is consumed about 100%,the workload of Asterisk server is good at that moment.This issue does not exist with the resolution 320X240(bitrate:204800,frameRate:11,fragsize:1500), the