similar to: Choose IAX or SIP trunking?

Hi all, Hopefully this will be quick, I'm looking for pointers to packages/ functions that would allow me to calculate the power of a t.test when the DV has measurement error. That is, I understand that, ceteris paribus, experiments using measure with more error (lower reliability) will have lower power. Mike -- Mike Lawrence Graduate Student, Department of Psychology, Dalhousie

I can use DSMARK to mark on the Egress side. Is there a way to mark/change the DSCP value of an incoming packet on the ingress side? Thanks. Jon Flechsenhaar Boeing WNW Team Network Services (714)-762-1231 202-E7

I have a private trunk with a peer. A call comes in from the trunk, and Asterisk calls a peer agent. If the agent transfers the call (this is a blind transfer using REFER), I want Asterisk to send a REFER back to the trunk, and essentially stay out of the loop. As set up, Asterisk initiates a new call using INVITE to the trunk and then bridges the original incoming and the new outgoing calls.

https://bugzilla.mindrot.org/show_bug.cgi?id=2246 Bug ID: 2246 Summary: PAM enhancements for OpenSSH server Product: Portable OpenSSH Version: 6.6p1 Hardware: Sparc OS: Solaris Status: NEW Severity: enhancement Priority: P5 Component: PAM support Assignee: unassigned-bugs at

Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped? -kkm

I have an Asterisk 13 that only processes app Transfer with PJSIP, to the tune of 60 per second. No voice calls. After like 2 hours, I can no longer get into Asterisk. This command, asterisk -r, fails, and also "asterisk -rx core show channels", etc. I am returned to the bash prompt. I checked the handles and lsof | grep asterisk |wc -l 7098126 I think there is a kind of handle leak

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what

Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source site<http://openingyourmind.wordpress.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

Hi All; I used A2Billing, basically it is nice and fine, but management possibilities is not that rich, so a lot of staff are need to be repeated that let the admin facing a problem of the needed time to do the task. Anyone advise for another open source prepaid billing that is rich by the management features? Also, I hope to find an open source Billing (prepaid and postpaid) that can work with

Hi everyone Anyone please explain me How Account code is use for billing., Thanks Nikhil

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part

The Asterisk Development Team has announced the release of Asterisk 13.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

The Asterisk Development Team has announced the release of Asterisk 14.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch Thanks

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more