search for: vardan

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

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12...

Thanks Vardan, I will like to know if this scenario can work when peer is not having fixed ip and we use host = nasir.server.com ? also I have set insecure=invite,port what if i use insecure=no thanks again. Message: 24 Date: Tue, 11 May 2010 10:52:14 +0500 From: Vardan <hvardan71 at gmail.com> Subject...

...ority can be specified. NOTE: You cannot use any additional action post answer options in conjunction 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

...> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 23 > To: asterisk-users at lists.digium.com > Message-ID: > <AANLkTim3qsPCY0sY3hanDxtTg8HPbXJEmj7HWZSknO1W at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Thanks Vardan, > > I will like to know if this scenario can work when peer is not having fixed > ip and we use > host = nasir.server.com > ? > also I have set insecure=invite,port > > what if i use > insecure=no > > thanks again. > > Message: 24 > Date: Tue, 11 May 2010...

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

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

...r1/${EXTEN}) exten => _X.,3,Hangup [callfromserver1] exten => _X.,1,Noop(Call from server1) exten => _X.,2,Dial(SIP/${EXTEN}) exten => _X.,3,Hangup Try so, I think it must work. And also, look and delete any another records in both servers in sip.conf about this servers settings. Vardan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100513/01e7a560/attachment.htm

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

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

Hello, you know , when a call setup, either caller hangup first or callee hangup first , the hangupcause will set to 16(means Call Clearing Causes) My question is how could i identify whether the caller or callee hangup the phone first? Best Regards! -- Thanks for your supporting, have a nice day. Sucan

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:

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

Hi, I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/. so Asterisk server 1 (192.168.250.111) sip.conf contains: [interboxsip] type=peer host=192.168.250.112