Re: [asterisk-users] JITTERBUFFER function

2015-01-29 Thread Matthew Jordan
On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson torbjorn.abrahams...@gmail.com wrote: Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it

Re: [asterisk-users] What conditions allow the use of dahdi native bridge?

2015-01-29 Thread Richard Mudgett
On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang lazy.char...@gmail.com wrote: Hi all, I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk 11.14.2 and DAHDI 2.8.0. I try to set callwaiting = no AND callwaitingcallerid = no in chan_dahdi.conf. But I can't find native

[asterisk-users] Semi OT - LDAP multi-valued attributes support in SIP phones

2015-01-29 Thread Olivier
Hello, I've just started to look at LDAP in IP telephony. 1. I've read parts of RFC2798 which defines inetOrgPerson class. I could find homePhone or telephoneNumber (multi-valued) attributes but nothing like phoneExtension. Did I miss something ? I not, what would you advise to store private

[asterisk-users] any valid up-to-date info about Kamailio-Asterisk integration ?

2015-01-29 Thread Kirill Marchuk
Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks about a (completely new, for me at least) approach to build scalable

Re: [asterisk-users] Investigating international calls fraud

2015-01-29 Thread Stelios Koroneos
The 25000$ @6.25/min means 4000 minutes of calls (or 66H) Not sure in how many days this has accumulated but i seriously dought this is made from a human accessing the phone. The fact that you get the calls at certain times might have to do with the timezone the calls are going If you phone

Re: [asterisk-users] subscriber absent

2015-01-29 Thread A J Stiles
On Wednesday 28 Jan 2015, Ethy H. Brito wrote: Hi all WE have some users that turns off their phones when they are not at home. We see the warning message: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) just after the Dial() command and a Everyone

[asterisk-users] JITTERBUFFER function

2015-01-29 Thread Torbjorn Abrahamsson
Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke

Re: [asterisk-users] Investigating international calls fraud

2015-01-29 Thread dk
It's very unlikely that this was an employee calling Mom for 66 hours (I'm assuming these calls appeared on a single bill). It's also unlikely that someone inside would benefit financially from making these calls. (Follow the money!) Don't discount the possibility that you've overlooked something

Re: [asterisk-users] Investigating international calls fraud

2015-01-29 Thread Michel Verbraak
Did you have a look at the phone it self already? Is call forwarding activated or something and can you call the phone/extension from externally? I have seen this in the past where an employee enabled call forwarding on the phone and once at home he or family called the phone which forwarded the

Re: [asterisk-users] Investigating international calls fraud

2015-01-29 Thread Bryant Zimmerman
If you have not done so contact the carrier immediately. Report the fraud. Have them disable international on the account until you have your security issues addressed. Ask them to pull call logs containing Source and destination IP address. for the fraud calls. If you are sure they came from

Re: [asterisk-users] any valid up-to-date info about Kamailio-Asterisk integration ?

2015-01-29 Thread Matthew Jordan
On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk 62...@mail.ru wrote: Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks

Re: [asterisk-users] JITTERBUFFER function

2015-01-29 Thread Torbjörn Abrahamsson
1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? You only need to use the JITTERBUFFER function. The jbenable option will enable a jitter buffer on every channel created for that peer (or, if global, for every peer in the system).

Re: [asterisk-users] What conditions allow the use of dahdi native bridge?

2015-01-29 Thread Charles Wang
Hi Richard, Thank you for your response. But after I remove the parameters of dial command (tTkK). The call was still not native bridge. Let me know if you have any suggestion. Best regards, Charles 2015-01-30 0:34 GMT+08:00 Richard Mudgett rmudg...@digium.com: On Wed, Jan 28, 2015 at 8:27

[asterisk-users] Dialplan for receiving faxes on Asterisk

2015-01-29 Thread Simon Humbert
Hi all, It looks like people commonly use this kind of dialplan when receiving faxes on Asterisk, with a jump to extension fax during the Wait() if a fax tone is detected: [start-here] exten = _X.,1,Answer() exten = _X.,n,Wait(n) exten = _X.,n,...do stuff... exten = _X.,n,Hangup() exten =

Re: [asterisk-users] JITTERBUFFER function

2015-01-29 Thread Torbjorn Abrahamsson
I thought this meant that jbenable alone was not enough, and that you needed to set jbforce=yes. Incorrect then Answering myself, it seems I was incorrect, as jbenable is enough to activate the buffers. I see the options different meanings now. Sorry about the buzz.. :) Second, if I