similar to: Extracting custom headers from SIP REFER

How is the billsec field calculated in CDRs? I have a situation where billsec is being reported as 0 despite the call being answered and a conversation occurring. An example record follows: '2007-11-06 21:36:50', '6495566778', '6495566778', '0116495566778', '1100012_1', 'Local/0116495566778 at 1100012_1-887b,2',

We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is

Hello I want to unlist the attached element getting only the first element in each element of the list. The last element of the list looks as this: [[5065]] [[5065]]$Pluv3Meses [1] 274.4 [[5065]]$PluvMesesMedio [1] 378.2667 [[5065]]$Pluv2UltimosMeses [1] 23.33333 So I would like to get for each element of the list the element called Pluv3Meses. The whole list has 5065 elements but when I try to

Hi folks, To provide MWI, * will send out a sip:notify message to the UA. The originator of this message is asterisk, as shown below; NOTIFY sip:1001@www.mysipproxy.com:5065 SIP/2.0 Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21 From: "asterisk" <sip:asterisk@66.121.xxx.yyy>;tag=as0ffb1bdc <=============== To: <sip:1001@www.mysipproxy.com:5065> Contact:

hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the

Hi, Does anyone has configured Asterisk to connect to BabyTel (a SIP Provider in Canada) ? Here is my sip.conf (I'm behind a firewall and I already opened port 5060 and 5065 (udp and tcp) to my Asterisk server): [general] port = 5065 context = Test insecure = very register => 1514XXXXXXX:password@sip.babytel.ca When starting Asterisk, the sip registration failed after 5 connecting

Hi, ALL: When I use astcc to do the prepaid function, but if I want to enable "call forward". The result of CDR seems not correct. UA 1011 make a call to UA 9999, and UA 9999 forwards this call to a PSTN number. I think we shall charge the credit from UA 9999 not UA 1011 because UA 1011 don't know where UA 9999 forwards to. But in CDR, I can only find the from(1011) and

I have seen a number of people in this newsgroup asking for information regarding asterisk interworking with Vocal. I was able to configure Vocal and Asterisk so that calls originating from vocal can land on an extension in Asterisk. I would like to share this info with the group The scenario that I tested was as follows. A call was originated from extn. 1001 on Vocal and the call was made to

Hi All, I am using rsync to backup our office server to our Internet server (RHE). As an association for doctors we are looking at providing a backup service for their practices using rsync. As it would be patient data it would need to be encrypted. I have found a few options, namely esync wurt rsyncrypto Does anyone have experience with the above and perhaps like to recommend one? On the

Hi, We're deploying a small VoIP solution for a group of teleworkers. Naturally, this exposes us to all sorts of fun, most of which we seem to have working properly. However, some NAT issues are still bugging us and we have noticed that often these situations didn't exist when users were connected directly to our VoIP provider, voiptalk.org. They have something which they call a

Hi all I fancied playing with SER and * on the same box. So i thought i'd just change the default sip port for * in sip.conf [general] port = 5065 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) restarted * and now when i issue a > ]# netstat -anp |grep 5060 > udp 0 0 0.0.0.0:5060 0.0.0.0:*

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

Hi all, First of all sorry for writing such a big mail.I am facing problem while implementing Piranha Load balancer on CENTOS 4 for my two oracle 11i application server running on linux. Oracle Real server details Instance Name - test url's - 1 . dev.xxx.com:8004 2 . uat.xxx.com:8004 Architecture - Services installed on dev are 1. Database Server 2.Concurrent Processing

Hello Masters Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients

hi! I apologize in advance if this is a newbie dumm question, but I really can't figure it ou. I have lists of sumeric and character data on some URLs, which look like this: <photo id="5876248819" owner="13716719 at N04" secret="faf9bb7f52" server="5264" farm="6" title="our rose garden" ispublic="1"

Hi all, In my Asterisk setup, incoming calls through Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be terminated properly after hangup. However, when calls were forwarded to voicemail, after recording & hangup the PSTN calls and cisco FXO port remained connected unless cisco port was manually shut/no shut. # key used to hang up the call did NOT help. Did anyone

On 15-04-09 12:06 PM, Chad Wallace wrote: >> but don't know where to put those lines. I have BABY defined as >> >channel variable: >> > >> >BABY = SIP/babytel_out >> > >> >but that seems circular, somehow. > You put them in the context for your clients... From what you show > below, I'd say they go in the "local_200"

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

Dear John, I belive your problem has to do with the sequence of startup. I think that .Rprofile is called before the required libraries are attached. You might like to try putting your code into a .First() function and run it that way. Cheers, Andreas Dr Andreas Kiermeier Statistician SARDI FOOD SAFETY PROGRAM 33 Flemington Street Glenside SA 5065 Phone: +61 8 8207 7884 Fax: +61 8

My firewall and asterisk pjsip config only has "permit" options for my ITSP's (SIP trunk) IPs. Here's the script that sets it up. -------------------------------------------------- #!/bin/bash EXIF="eth0" /sbin/iptables --flush /sbin/iptables --policy INPUT DROP /sbin/iptables --policy OUTPUT ACCEPT /sbin/iptables -A INPUT -i lo -j ACCEPT /sbin/iptables -A INPUT -m