similar to: Transfer dropping calls

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

I have an asterisk box using a SIP provider and IAX2 softphones clients. Audio is low and I need to apply some gain on it. How can I configure such gains - in/out on sip and iax2 channels? Thanks, Valter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101106/85417a17/attachment.htm

Hi all ! I don't know what else to do but I absolutely can't get it to work !!! :( I've installed 2.2.8a on my SGI 6.5.17 workstation and all the tests I've done failed. I'm getting quite desperate. :( Ok. Point by point I'll try to explain what I did. - Installed the 2.2.8a successfully and started it successfully. - Logged in the samba (almost) successfully from the

Hi, Just wondering if anyone else is able to reproduce this with the current * (CVS 12:15 GMT) Call Snom from any device (tested with i4l, zaptel and SIP). Answer call and try to transfer call using transfer button on Snom. After dialing new number press OK (F4). At this moment the Snom users hears dialtone, but the caller still hears the Snom user... Even hangup on the Snom doesn't

Hi ! I'm sending you all my test smb.conf and the log file generated with a 2.2.5rc4 installation. The trouble is that I cannot create new directories (I'm using a WinXX Italian version). Then under Win98 I get the most strange behaviour. In addition to the above, using win98 I can't copy files from the share to the local disk (error 1026). Please help Thanks Valter Simo Sorce

hello friends, i have to configures asterisk n my hardware details are O.S - Ubuntu 8.04 Lts Memory - 1 GB Proccessor- core 2 duo is any one having a good link or how to related asterisk. any help,support will be higly appreciated thx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) >> What interface are you using? ZapBRI? if so you might be able to do the >> hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Gr?sse

Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like " I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him "wait a sec" and push "Flash" and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to"...

Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash I hear no dialtone, and only a long and then small beep. When I try to dial digits I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've tested the S100U unit in two different computers both with different USB subsystems. The one it worked the best in used the usb-uhci USB driver. The other system uses the usb-ohci USB driver. To make sure it was not the installation, I used the same hard drive that had asterisk in both systems. And I moved the X101P and S100U between both

All, Currently I've got my Asterisk machine running smoothly on IP bases. Meaning I can reach all phones or softphones within my LAN or remote LAN's via VPN. The next step for me is connecting it to the PSTN network. After some tweaking with the modem.conf I got the i4l driver running correctly, and it appears that my Fritz! ISDN v2 card is working correctly. I have added the

>From what i've read i think that asterisk is suppose to generate the dialtone. But when i pickup my analog phone it seems that the Openswitch12 card is the one playng the dialtone. Im almost positive the card is playing the dialtone, can anyone confirm this? And if so how can i change the dialtone?

Hi We have a fairly simple Asterisk setup for a callcenter: around 10-15 operators running SIP softphones (X-Pro) and an Asterisk box connected to a E1 service using a Digium T100P, and to our legacy PBX (NEC) over a Digium TDM400P FXO interfaces. Everything is working except our Music on Hold after a transfer of an incoming call to our old PBX, that does not de-attach itself from the transfer.

Maintainer: Express Line Date opened: January 17, 2006 Status: Open Value of bounty: $5000.00 Licensing for code: We retain intellectual rights to the underlying source code. We need Asterisk (stable version) to be able to perform a 2B channel transfer for a NI2 B8ZS PRI line. We can't use a channelized T1 at the time for our work. This feature is commonly called a call transfer on analog

Hi, I would like to shape incomming traffic on eth0 and ipsec0 (binded to eth0). I need to set minimal bandwidth to some packets going via ipsec0 interface. It is running fine when I simply mark the ESP (protocol 50) packets in the PREROUTING chain - means all ipsec packets are shaped. Like: iptables -t mangle -A PREROUTING -i eth0 -p 50 -j MARK --set-mark 30 iptables -t mangle -A PREROUTING -i

Hi ! I've just finished setting up a samba 2.2.6pre2 installation on a SGI running IRIX 6.5.16. I've discovered that logging on a win2000 client, if I create a new file on a test share everything works but when I try to create a new directory I get the following error on win: Impossible to create the folder "New Folder" Impossible to find the specified file. Increasing the log

The process for transfering a call with the Bugetone is as follows.. 1. Press "transfer", you will get a dial tone.. 2. Dial in the extension to wish to transfer to.. 3. Press the "Redial".. (on the newer phones this is the "send" button) You don't need the "t" option on your dial string to do transfers with a budgetone.. Later.. > With the right

Hi! I'm trying to transfer a incomming call from a Queue to another extension. I'm receiving a call from a queue with the AgentCallbackLogin. The queu is as following: Queue(sales|t) Which should allow transfers. So as soon as the call is answered I would like to be able to transfer it When the agent presses the # I get the dialtone but as soon as I press any digit Asterisk tells me

Hello all.. i''m a newbie in LARTC... And I reallize that my problem is a clasical problem in this LARTC milist..... So if anyone dont mind.. please help me with this... Situation : LAN --> Gateway <-- Internet 10.0.0.0/8 --> eth0 eth1<-- 20.0.0.0/8 Linux Box: RedHat 8.0 /sbin/tc size is +- 700 kb #NAT to all computer to access

Hello, some problems with transfer and zap... one hfc-card in NT mode and one fritz isdn-card in server. there is one gigaset SX353 isdn phone on the hfc-card. anybody calls from external via capi and the call is bridged to the zap-device. if you want to transfer the call via R-button on the isdn-phone the caller get the music-on-hold. you get a dialtone and dial - if the called person gets on