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...nction with this option. Also, pbx services are not run on the peer (called) channel, so you will not be able to set timeouts via the TIMEOUT() function in this routine. Thanks -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: info at eif.am www.eif-it.com

Dear sir or madam, I am a 4-th year student at Yerevan State University, Armenia; and I am studying LLVM in order to write my Bachelor thesis. I am trying to write an llvm pass that just removes all "Add" commands and gives some statstics. Nevertheless, I get this segmentation fault: ................some rows about functions, that are not ch...

Hello, We have a strange situation (asterisk 1.6.2.14), where we get a result for DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL. This is the (relevant) test dialplan: -------------------------------- [incoming-private] exten => _X., n, Dial(SIP/1001,30) exten => _X., n, NoOp(${DIALSTATUS}) exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1) [incoming-status] exten

Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto

Hello all, I need that Asterisk Always use Reason in a CANCEL. How to do? thank you *Fran?ois * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100521/f3a91f36/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: francois.vcf Type: text/x-vcard Size: 400

Hi All; Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is working fine and it is same as 1.4? Appreciate ur kindly help. Regards Bilal

...enter the parameter and don't want to change anything in the code which is quite difficult/. Thank you in advance, Hrach -- Hrachya Astsatryan Head of HPC Laboratory, Institute for Informatics and Automation Problems, National Academy of Sciences of the Republic of Armenia 1, P. Sevak str., Yerevan 0014, Armenia t: 374 10 284780 f: 374 10 285812 e: hrach@sci.am skype: tighra [[alternative HTML version deleted]]

Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/xxxxxxxxxxxx ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound

Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 11/3/2011 4:55 AM, Arshak Nazaryan wrote: > Dear sir or madam, > > I am a 4-th year student at Yerevan State University, Armenia; and I > am studying LLVM in order to write my Bachelor thesis. Hi Arshak. > I am trying to write an llvm pass that just removes all "Add" commands > and gives some statstics. > Nevertheless, I get this segmentation fault: The problem is most li...

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These