similar to: Transfert call

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 everybody, I'm trying to use msn with * and for that, I'm reading all information on the mailing list. You used to recommend the page http://mcleod.pbx.nq.net/msn/, but I always get an error while opening. Has it changed? Is there another one? Thanks cmayor ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y

Hi, Talking about messenger,,, it's still necesary to do HKEY_CURRENT_USER\Software\Microsoft\MessengerService\Corp2PC_Phone equals to '1' ??? But it's still sending the '+' digit, so it's necesary to stripMSD? Thanks a lot cmayor ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y m?s...

Hi, I asked a time ago what were callgroup and pickup group used for. I have done some proofs and all, and I'm not sure if I have pick the idea up well!! That's what I understand: For example: group=1 callgroup =2 and pickupgroup=2 and my phone is a membership of the group 1. that's mean that when a phone that belong to group 2 is ringing, I'll be able to answer this call dialing

what is the format for an h323 entry in the dialplan? can I use chan_h323 without compiling anything else or should I compile oh323? basically what's the best way :) cheers Dave

Hi since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? thanks regards bk

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

Hi, I try to use Libiax in order to put un transfert button in my iax softphone. Is there a way to make a call transfert ? Best regards rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031210/e1e6acf3/attachment.htm

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

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

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

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

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

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

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

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, 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! Is it possible to merge two ext3/ext2 partitions into ONE ext3/ext2 partition? -- Ralf Hildebrandt (Im Auftrag des Referat V a) Ralf.Hildebrandt at charite.de Charite - Universit?tsmedizin Berlin Tel. +49 (0)30-450 570-155 Gemeinsame Einrichtung von FU- und HU-Berlin Fax. +49 (0)30-450 570-916 IT-Zentrum Standort Campus Mitte AIM. ralfpostfix

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