similar to: Transfert with IAX

Displaying 20 results from an estimated 1100 matches similar to: "Transfert with IAX"

2003 Jul 08
2
Transfert call
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:
2003 Oct 29
6
SIP client
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
2004 May 13
1
How to improve transfert rate with rsync
Hello, 1) I am using rsync with gentoo and all emerge are very fast 400 kb/s ADSL connections. When I am using rsync with two computers with the same bandwith connection (ADSL 400 kb/s) transfert is very low (40 kb/s). options are "rsync -avzub". How can I improve the rate of transfert ? I saw That it use sftp. Is there a configuration file for sftp that improve the transfert ? 2) How
2003 Nov 04
1
Call Transfert with SwissVoice IP10S in MGCP mode
Hello, Now that I have a nearly working configuration for my IP10S with * I wonder if anyone has done call transfert with this Phone. In the IP10S documentation they talk about the 'service key' wich is the key with the white dot on it. With this Key, it should be possible to have a menu with call transfert entries. This menu should (accordingly to the documentation) depend on the
2005 Jul 21
1
attended transfert
hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks
2003 Sep 24
3
Call transfert with dial plan
Hello, As I have problems getting transfert call working with my grandstream SIP Phones, I woul like to know if it is possible to do it with a proper dial plan in exten.conf. I haven't found any information about that in the docs. Regards, Daniel ANDRE -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
2008 Jan 17
1
Iax Encryption
Hello, from what I've understood Iax2 should support aes128 encryption. I've found this old info: http://www.voip-info.org/wiki/view/IAX+encryption and this (unanswered?) post http://lists.digium.com/pipermail/asterisk-security/2005-August/000060.h tml Is this the libiax used currently on asterisk http://ftp.digium.com/pub/libiax/ ? I would like to understand if someone is using this in
2003 May 20
1
Error Connecting using libiax
I have been working on a cross platform library for connecting to asterisk along with Steve Kann. I recently was testing the library against the recent code for asterisk from CVS and got the following error when using the command line client (similar to miniphone). Text Based Telephony Client. TeleClient> call 192.168.2.40 TeleClient> Unknown event: 15 dump Dumping call! TeleClient>
2005 Mar 03
3
Audio pausing over IAX trunk
I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to
2003 Apr 07
6
ISDN4Linux problems
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
2008 Feb 28
1
[Samba to samba transfert] Timestamp problem
/[Same as previous mail cause previous filtered and empty]/ Hello all, I recently did some transfer of files located on a samba server to another samba server. To do this I used nautilus from the destination machine to connect to the source computer and initiate the transfer. I realized then that the /date created/ and the /date modified/ info of the transfered files were the same as the
2003 Aug 04
2
H323 CallerID
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
2003 Sep 01
2
gnuGK + h323 Caller ID
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:
2012 Jun 01
0
r-project, votre plateforme transfert de fichier
[1]transfert [2]transfert [3]fichier [4]fichier [5]s??curis?? [6]s??curis?? Bonjour, Vous ??tes plus de 200 000 utilisateurs ?? ??changer vos fichiers professionnels en toute simplicit?? et de mani??re s??curis??e avec notre [7]plateforme. Saviez-vous que nous
2003 Dec 08
9
IAX clients
Hi, Is there IAX client in Applet JAVA which can be embeded in a web page ? Best regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/c388ef61/attachment.htm
2004 Jan 30
2
IAX call problems
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] --
2003 Sep 05
2
Transfer (again!)
Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice
2002 Aug 18
0
rsync stops files transfert unexpectedly
Hello, i used rsync for some years ago...Well fine because i synchronize laptops and servers or laptops and PCs using rsync AND between both unix/linux machines and unix/linux filesystems. Ok. Now, i have to synchronize files that are on unix/linux filestems AND on win filesystems (NTFS and FAT)...I thought: "don't worry, mounting win filsystems with samba and so on". Yes
2003 Aug 27
2
include context
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
2013 Jan 10
0
Samba is slow and crash when i transfert one file
Hi at All Samba stop my transfert when i want to copie one files or is very low. I don't for why in my log, i have when my copie crash : [2013/01/10 21:40:56.442652, 3] smbd/oplock.c:895(init_oplocks) init_oplocks: initializing messages. [2013/01/10 21:40:56.442712, 3] smbd/oplock_linux.c:224(linux_init_kernel_oplocks) Linux kernel oplocks enabled [2013/01/10 21:40:56.445978, 3]