similar to: sendmail or postfix?

Hi, I have a task to load test a few VoIP servers, and also test our trunks on regular intervals to see how reliable they are, i.e. how often they go down, if at all. I did some search and found a lot of VoIP testing tools. I selected some of them and testing them one by one. But so far haven't found any which could do good load testing and test the trunks. So far SIPp is the only one which

I am wondering if there is any such thermostat available which can be controlled from Asterisk. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be

Hi, Is it possible to create a clear zaptel channel which doesn't require to be picked up? The requirement of my client is to open a clear channel to a recorder which starts recording certain message. Currently the channel which is created by zaptel requires the other end to answer the call, and the recorded can't answer, so the channel get hung up after a certain number of rings. Zaptel

Hi everybody, I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work with new versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should work fine. I have sound breaking many times a day over the trunks. Server is AMD Athlon 2.4 GHz with

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

Hi, After upgrading to new version of Asterisk and Zaptel, my Tormenta2 card has stopped working. Zaptel doesn't detect it. What should I do to make it work again. It is enabled in /etc/sysconfig/zaptel. Where else I have to enable it so zaptel can detect it and make it work. zaptel.conf and zapata.conf are the same as I had them before. I tried googling but couldn't find any help. --

Hi, I've tried a few times wiki instructions for setting up a streaming MoH, by creating stream.mp3 file in ../moh/stream folder. It never worked. I need to setup streaming MoH and please if someone has a working example, please help me to set it up. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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,

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

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

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

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

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 >

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

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:

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

Greetings everyone, I've been trying to make this java based webphone work for everybody visiting my website, but seems like for many users it doesn't work. In order to get a better idea what is the success rate of this webphone, I would appreciate help from anybody who could make a few calls from it within North America and if it doesn't work, send me what error you get, or if it

Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com

Hi, Everything was working fine on this 10 phone office, but for last few weeks they are complaining that audio goes blank for a few seconds during the conversation, and then comes back on. It goes blank for both parties. What are the possible causes for this to happen? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: