similar to: sip and iax2 audio volume gain

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,

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 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'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

I am investigating an issue with IAX2 extensions in Asterisk 1.8.x. My application connects to Asterisk via AMI and attempts to run an Originate command between an extension (such as SIP/5555 or IAX2/8888) and an application (in my case it is AgentLogin). This works correctly for SIP extensions, in all Asterisk versions. With IAX2 extensions, this worked correctly in Asterisk 1.6.2.20, but

We're encountering no audio issues in ConfBridge. Only the moderator and the 1st invited have audio. When the 2nd invited number picks up the phone, only the announcement is heard, then followed by silence. This occurs frequently. Only 1 out of 5 test calls do not have this problem. We were hoping that switching from MeetMe to ConfBridge will solve the no audio issue, since there's no

Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

Hi, I'm having some questions related to echo cancellation configuration on a Digium board enabled systems (B410P, TE420, TE420B, ....) for cases when a hardware ech canceller is present or not. I read in TEXXX manual that when setting echocancel=yes in chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo cancellation was enabled. 1. I'm correct thinking that it is then

Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Thanks! Regards, Ronald -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. P.S. Are both options part of the configs of Asterisk or need modules to be selected and installed before doing the

Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!!!!!!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e99afa31/attachment.htm>

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->

Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -------------- next part -------------- An HTML attachment was

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

Howdy All, I'm considering Asterisk / Digium as a replacement to my existing phone switch. I need to continue to be able to push analog lines between multiple buildings in a campus environment. The Digium Analog 410 Card manual states it's not recommended to go beyond 1500 feet distance for an FXS card, and no line should leave the building or be bundled. The 2400 Series Manual does

Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000 at mydomain to call 9000 at my domain(which is the call center)