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Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110810/365d9d56/attachment.htm>

Hi, I would have thought that when spawning a channel using the Originate() dialplan command, variables prefixed with two underscores would be preserved. However this does not work in the following case. Dialplan code: [intern] exten => 200,1,Set(__myvar="foo") exten => 200,n,Originate(Local/123 at test_orig,exten,dummy) [test_orig] exten => 123,1,NoOp(${myvar}) exten =>

Dear, the goto function to the iax dialing, makes bill duration and call duration wrong, in cdr.they are equal to ringing time. the cdr will be produced and saved into the dbase, when the callee picks up the phone. is any way to have real duration time ? [main] exten => _1X.,1,GOTO(LOPL,${EXTEN},1) .... [LOPL] exten => _X.,1,Dial(IAX2/MAIN/${EXTEN},60)

Dear,is any way to find silent channels , and disconnect them after 30 secs? best -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090824/7709c910/attachment.htm

Dear, Faxter is an opensource email to fax gateway, please check it, let me know if any bug. best -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110130/0f418a92/attachment.htm>

Dear this note is only for fresh administrators don't think about asterisk security. I found fail2ban very useful for anti asterisk hacking, so I want to share it with fresh admins. some hackers try your sip or iax2 ip with a lot of username/password, may be after 1 million try, one username/password was accepted. so in 2-3 hours, they use all of the credit of the hacked user. fail2ban, runs

Dear, do you have any successful experience for installing SHT-8C/PCI/FAX (synway) with asterisk ? is it compatibe with asterisk (dahdi/zaptel)? best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110525/9df2050a/attachment.htm>

Hello On a uClinux-based appliance, "ps aux" shows multiple Asterisk processes: 380 root 11990 S asterisk -f 381 root 11990 S asterisk -f 383 root 11990 S asterisk -f 384 root 11990 S asterisk -f 385 root 11990 S asterisk -f 386 root 11990 S asterisk -f 387 root 11990 S asterisk -f 388 root 11990 S asterisk -f

Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer

Hi all, I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine via a T.38 enabled trunk.? I've got t38pt_udptl = yes faxdetect=no in my sip.conf file.? The ATA has all of the T.38 options turned on, echo cancellation is off, as well as silence suppression off.? The only configured codec is u711.? When the user tries to send a fax, it gets to the point where it

...s.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110510/42c6c a17/attachment-0001.htm> ------------------------------ Message: 12 Date: Tue, 10 May 2011 11:48:11 +0430 From: Pezhman Lali <lopl at lopl.net> Subject: Re: [asterisk-users] 40sec between dial execution and sending SIP request To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <BANLkTi=VJdo-wwuT36PNMRoc-1uJ30F7XQ at mail.gmail.com> Content-Type: text/pl...

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

Dear, some iax phones,(with built in router) have problem, with our asterisk server, there is no way sound if they call out, but it's ok if somebody calls them. the normal iax phones without router have'nt ny problem. can u help me? the version of kernel is 2.6.18 and asterisk is 1.4.26.2 Best -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear is any way to have a secure (encrypted) rtp line between cisco 79XX and asterisk with SCCP? best -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110206/d1a1c1a9/attachment.htm>

Dear I installed chan-sccp-b v3 on a powerful virtual machine, with 4 cpu cores and 16GB RAM(enabled in kernel by PAE) about 1,200+ clients are going to register in this machine. all data of clients are saved in ORACLE. The asterisk (1.6.2.18) connected to the database throw odbc(unixodbc). all logging are disabled( verbose, debug and sccp debug) . the asterisk was crashed every few minutes. here

Dear, Huawei softx3000 sends the dtmf with undefined content-type(sscc) and format, so the asterisk can not recognize the digits, maybe changing the source code of asterisk be a good solution, but I am looking for a better way. would you please let me know if you have a better solution. Best <--- SIP read from UDP:1.1.1.1:5060 ---> INFO sip:050111111 at 213.203.201.51:5060 SIP/2.0

Dears, There is a PSTN line shared between 2 asterisk servers, (openvox 4FXO lines) The outgoing call of the one server may be conflict with the established call of the other one, is any way to force the Asterisk or Dahdi to dial after hearing the Dial tone ? -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: