similar to: How to Change size of CDR(accountcode) variable?

For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to

I am trying to get cdr via odbc to work on Asterisk 13.7.2 but I keep getting this error: [Feb 9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc: Error in ExecDirect: -1, query is: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequence) VALUES ({ts

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

Hi All I have dozens of these messages on CLI complaining about database connection and error writing CDR to disk. The curious thing is I can find them all inside the database. I "selected" them using uniqueid and manually compared each column with the cdr_adaptive_odbc.c error line. "mysqlcheck -a -e -v DBase" and "mysqlcheck -c -e -v DBase" both returned OK for

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

Trying to do something like this in the sip.conf under my incoming provider profiles: setvar=CDR(accountcode)=${EXTEN} It seems to show up in the CDR but it's showing up exactly like this "${EXTEN}". Is there a way to stuff the DNIS (number dialed) into the accountcode for CDR? I have already accomplished setting on a number by number basis, I just want to do it globally for all

Hi everybody, How can we add new contexts in asterisk realtime module? All I could figure out after googling is that a new context HAS to be declared in extensions.conf with 'switch => Realtime/@<databasetable>' under the context name declaration. This works fine as long as we are adding extensions only to this one context, but doesn't give the freedom to add new contexts for

For a while now I've been looking for a good web based soft phone solution, but so far no luck. A few solutions which I've tried, both Java based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas,

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

On Sat, 25 Apr 2015 17:11:34 +0200 jg <webaccounts173 at jgoettgens.de> wrote: > > > Hi All > > > > I have dozens of these messages on CLI complaining about database > > connection and error writing CDR to disk. > > > > The curious thing is I can find them all inside the database. > > I "selected" them using uniqueid and manually

I've got a 778 DID for vancouver, but don't know if it will be a local call if dialed 604 and vice versa. What are the different area codes in Vancouver and why its easier to get 778 DID than 604? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, Slightly off the list topic, but I hope I'll get some help here. Somebody wants me to implement for his project a Cisco based VoIP system. I told him that I specialize in Asterisk based systems, but he is not even aware of Asterisk. The requirement of project is such that chances are slim that this firm will consider Asterisk based system. So I told him that though not experienced

Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks!

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it

My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part

Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load