similar to: Stress-Testing Asterisk

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Hello My Aster-Friends! I would like to hear if anyone out there in Asteriskland has used the Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. I have a couple of these boxes that I need to make work with Asterisk, connected with Digium TDM400P card. Anyone tried it before, and how did it go? Thank you. Jeng ___________________________________________________________ Yahoo!

Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

My Gurus! I'm still playing with asterisk in the lab here. There is a feature that I need in a production asterisk system. I was wondering if it already exists in asterisk. When we want to shutdown a production asterisk system, we would like the shutdown to happen after there are no more calls being processed. In other words, a shutdown command that does the following: - block asterisk

Hi everyone, I'm in trouble with queue. There are a little local radio station with one studio and we have to switch queued callers to the live program. Everything works fine (counting callers, periodic announcements), but while the announcement is played for 'firs in line' caller, studio gets a free line out not the caller. member = SIP/suich ;only 1 member strategy = ringall

We are in the process of implementing realtime voicemail. I was wanting to "stress-test" the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup()

Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: ---------------------------- /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field ?owner? specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: ***

Hi everyone, I'm new to Asterisk, but I like it ;o) Have a question to you; How can I limit the incoming call duration? -- Suich

Hi everybody! Can you give me a hint, how can I start a call from asterisk with some (php, bash, etc) script? I need to start two calls and bride it together. Thank you. -- Suich

Hi, I just installed Sipp 3.3?on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. SIpp output: ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- ? Start Time???????????? | 2013-05-20?22:53:08:637?1369083188.637273??????????? ? Last Reset

Hi Barry Flanagan, Barry Flanagan <barryf-lists at flanagan.ie> schrieb am Mit, 19. Aug 11:06: > SIPP is probably what you seek. http://sipp.sourceforge.net/ > > Hope this helps. That looks pretty like what I'm looking for! Many thanks! Sincerely, Dominique Haeber

Hi all, i need to test how many calls can withstand an Asterisk server. Do you know any good tools to strain the server? At best, there are scripts that I can run on a Linux server. I thank you for your tips Sincerely Dominique Haeber

Hi Asterisk Gurus! My lab asterisk box has 1 FXO and 1 FXS ports in it. I connect a GSM phone to the FXO port. I connect a regular corded phone to the FXS port. The Dial() application for both incoming and outgoing calls specifies the A(hello-world) flag. From another GSM phone, if I call the extension (corded) phone attached to the box, it plays the hello-world file when I pick it up. But

Hi All, I have a telephony project for which I need to build a prototype to demo for management. The prototype must work on a GSM phone network. In the demo system, a call from GSM phone comes into the demo box. The demo box runs CallWeaver. Callweaver picks up the GSM call, answers it and plays a sould file, then dials out to a second GSM phone somewhere and connects them so they talk. My

Hi, I've successfully setup Asterisk on my local PC and can make call using Twinkle to the server. But, I cannot call to my Asterisk server at Rackspace. I have been trying several things to figure it out, no luck. My PC is behind NAT, so I've set that up in sip.conf (nat=yes). I can ping my Rackspace server so it seems to be Public-static IP. Anyway, I tried with setting externip,

Hi, How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be "programmed" to send out digits after specified time to simulate users pressing menu items. If it can originate large number of calls simultaneously then it's great! Does any one have any recommendations ? Any other method to stress test an IVR call flow? with regards, raj

Sorry, but it seems that I have banned from list. I can reciveve, but can not send posts. Hi! When I use Dial(type/identifier, timeout, A(some_file)) CDR billsec starts when announcement ends. But I have to bill from when called party answers to phone. How can I solve my problem? -- Suich

Hi People! Is there an underground asterisk command reference manual that the Gurus here share amongst themselves only? :-) The reason I ask is that sometimes I see mention of an asterisk command and I scramble for my asterisk book (pdf) to look it up but can't find it in there. For example, I saw here last week people talking about the Set() application with the "If" conditional

I want to have a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP generation) for stress testing and precise analysis of the VoIP network equipment. Do any one knows a free program can do that . Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with

Hi, I've tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s