similar to: About voip gateway

I want to create a voip service, I do not know much about it, but the first thing I want to know if more than one client can make a call at the same time through internet to the PSTN, and what gateway should I use for this. 2014-12-08 13:07 GMT-08:00 Steve Edwards <asterisk.org at sedwards.com>: > On Mon, 8 Dec 2014, Leonel Florin wrote: > > Hay friends, I want to know how many

On Mon, 8 Dec 2014, Leonel Florin wrote: > Hay friends, I want to know how many simultaneous call can i do > throughout a voip gateway from the internet call to the normal telephony > network, because i want to see what implementation do i have to do > multiple call from internet to differents telephones. Please reply with a few more details of what you are planning on doing. For

I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

On Thu, 6 Aug 2015, Steve Edwards wrote: > Would comparing an INVITE from X-Lite or X-Pro with the INVITE from > Asterisk yield any clues? On Thu, 6 Aug 2015, Murthy Gandikota wrote: > For Asterisk INVITE please view > > http://pastebin.com/v15vMax4 > > For X-Lite INVITE please view > > http://pastebin.com/rmHZKu3N Just a quick glance (because I'm not a SIP

On Thu, 6 Aug 2015, Murthy Gandikota wrote: [trimming cruft nobody cares about anymore] > I use the same password for INBOUND and it works fine! Something amiss > with Asterisk OUTBOUND? because I used the same password with X-Lite and > X-Pro Vonage soft phones with successful calls. Would comparing an INVITE from X-Lite or X-Pro with the INVITE from Asterisk yield any clues? --

http://yro.slashdot.org/article.pl?sid=04/08/04/2212251&tid=158&tid=95&tid=103 Probably some of you already saw this. Now, beyond discussions regarding the legitimacy of such a ruling (whether they have the legal, moral or whatever right to enforce it), there's the technical aspect. Suppose i provide VoIP services using Asterisk, and i fall under the incidence of the FCC ruling

Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. I'm googling now and I recognize that Friday afternoons are the worst time to ask questions, but I'm getting desperate because this is keeping me from rolling a system out to production. (Yup, I know. Who rolls out a system on a Friday

Dears I already read most of post on asterisk group and (http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning) But I could not find a calculator 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding , 3-And what is the number of

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part

I have used the handytone 488 from grandstream in the past.... However I need to be able to send a number to a unit like the 488 and have it dial out. Is there a unit like this available? Basically a 488 unit that can place a call out. Jerry

Hello everyone, I have some questions regarding implementing asterisk and my-sql. I'm no expert at asterisk but I'm going to list up my questions and hopefully someone will be able to help. -The people that setup our server made asterisk write the sound files onto the hard drive, and then somewhat later store these files into my-sql. Is this the proper way to do it? Or would it make

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently

Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the

Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via

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Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

Hi, We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. I've found an issue when Asterisk tries to make a SIP call out using auth, but has the wrong credentials and keeps getting returned a SIP 407, in this example to an OpenSIPs server requiring user auth. Basically this happens: 1. Asterisk sends plain INVITE to OpenSIPs 2. OpenSIPs responds with SIP 407 auth required with a