similar to: How to kill AMI ORIGINATE on-the-fly

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

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:

We relay on 'failed' extensions after AMI ORIGINATE command. When moving from Asterisk 1.8.22 to Asterisk 13.2, it has stopped to work. I belive that it is due to a change in pbx.c => ast_pbx_outgoing_exten. Thanks, Valter -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -------------- next part -------------- An HTML attachment was scrubbed... URL:

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

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

We redirect calls to another pbx - which are not under our control. So, we need to distinguish when an operator answer from the time the call stays on queue. To do so, we use dial cmd with M option. M calls a macro that repeats a digit (saydigit) until someone dial it. When operator dials the correct digit we finish the macro making it to be bridged (caller and called). It almost works

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

When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse event with a Failure Response, Reason 3. My phone continues to ring. If I answer the phone at this point, it attempts to answer, but does not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

Hi all, my domU guests stop booting in my new 3.2 (kernel 2.6.18.8) xen installation when run perfectly in my other 3.0.2 (2.6.16) xen server. whe i start the domU with "xm create" i get these messages: Using config file "./debian.4-0.xen3.cfg". Started domain debian.4-0 IP route cache hash table entries: 4096 (order: 2, 16384 bytes) TCP established hash table

Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk <-> SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything

Hello everyone, I am having a bit of problem getting MusicOnhold to play. I am running Asterisk 1.4 with MPG123 0.59 installed. And here's what i see in the debugging window of asterisk: -- Started music on hold, class 'default', on channel 'SIP/x123-082043d0' -- Stopped music on hold on SIP/x123-082043d0 Any idea why it is not playing the file at all? thanks

I have incoming analog and SIP DIDs that all ring multiple sip extensions with a Dial command as the first exten. I am curious to know if it's possible for the incoming caller to transfer out of the Dial command while in progress and dial a single extension? Thanks! jlc

I have a TE210P digium card that has 2 E1/T1 ports. the code in my extensions.conf file for span 1 is : [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=Zap/g1 ; Trunk interface TRUNKX=Zap/g2 ; 2nd trunk interface ... ... ; dial a long distance outbound number to SPAIN ; This

Hey, all. I was just wondering if there were any tools/utilities/what-have-you out there that would allow a user to click on a contact in Outlook, and have their phone dial it? (Or, I guess, have Asterisk dial both their phone and the destination number, and put the two into a conference.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is

Hello may situation is the next: Asterisk <--> NAT1 (router)<---> internet <--> NAT2 (router) <--> x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone

Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r