# IAX, SIP, Asterisk? **URL:** \ **Category:** Asterisk General\ **Created:** [February 19, 2009, 7:54am UTC](https://community.asterisk.org/t/iax-sip-asterisk/25633 "2009-02-19T07:54:21Z")\ **Posts on this page:** 7\ **Page:** 1
**Author:** ![knowledgestudent](https://avatars.discourse-cdn.com/v4/letter/k/c4cdca/32.png) [@knowledgestudent](https://community.asterisk.org/u/knowledgestudent)\ **Post date:** [February 19, 2009, 7:54am UTC](https://community.asterisk.org/t/iax-sip-asterisk/25633/1 "2009-02-19T07:54:21Z")
if I am using the IAX API in my application that makes calls , and have an extension to be connected to all other , other extension are using sip phones , we are all connected to (trixbox) asterisk, we would be able to call each other, wouldn’t we? I mean would it a problem to be different in our account, in other word, is there any problem if users connected to the Asterisk either via sip or IAX?, what are main differences beside that IAX use one UDP port, …! which one is better for calls? thanks in advance ---
**Author:** ![mbruni](https://avatars.discourse-cdn.com/v4/letter/m/ecccb3/32.png) [@mbruni](https://community.asterisk.org/u/mbruni)\ **Post date:** [February 19, 2009, 8:22am UTC](https://community.asterisk.org/t/iax-sip-asterisk/25633/2 "2009-02-19T08:22:25Z")
> [@](#): > > we would be able to call each other, wouldn’t we? Yes sure. Cheers. Marco Bruni [www.marcobruni.net](http://www.marcobruni.net) ---
**Author:** ![yuxincarol](https://avatars.discourse-cdn.com/v4/letter/y/7ba0ec/32.png) [@yuxincarol](https://community.asterisk.org/u/yuxincarol)\ **Post date:** [March 2, 2009, 2:53am UTC](https://community.asterisk.org/t/iax-sip-asterisk/25633/3 "2009-03-02T02:53:57Z")
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**Author:** ![riddlebox](https://avatars.discourse-cdn.com/v4/letter/r/dbc845/32.png) [@riddlebox](https://community.asterisk.org/u/riddlebox)\ **Post date:** [March 4, 2009, 3:44pm UTC](https://community.asterisk.org/t/iax-sip-asterisk/25633/4 "2009-03-04T15:44:38Z")
yuxincarol, If I wanted ads everywhere I would just sit down and watch tv! ---
**Author:** ![knowledgestudent](https://avatars.discourse-cdn.com/v4/letter/k/c4cdca/32.png) [@knowledgestudent](https://community.asterisk.org/u/knowledgestudent)\ **Post date:** [March 11, 2009, 9:05am UTC](https://community.asterisk.org/t/iax-sip-asterisk/25633/5 "2009-03-11T09:05:34Z")
after initiating a call from an application using the manager API, and made sure that a channel was initiated successfully but there is no voice . how possibly should i handle the voice so that the user really has a call not just ringing and answering and then nothing ?? I am sorry if the Question is silly some how 😒😒 I have read about the RTP , I am using java , if it is really the best answer to talk via the asterisk , where to start thanks in advance Regards ---
**Author:** ![mbruni](https://avatars.discourse-cdn.com/v4/letter/m/ecccb3/32.png) [@mbruni](https://community.asterisk.org/u/mbruni)\ **Post date:** [March 11, 2009, 9:16am UTC](https://community.asterisk.org/t/iax-sip-asterisk/25633/6 "2009-03-11T09:16:35Z")
Check [asterisk-java.org/0.2/tutorial.html](http://asterisk-java.org/0.2/tutorial.html) , there is an example about Originate. Cheers. Marco Bruni [www.marcobruni.net](http://www.marcobruni.net) ---
**Author:** ![knowledgestudent](https://avatars.discourse-cdn.com/v4/letter/k/c4cdca/32.png) [@knowledgestudent](https://community.asterisk.org/u/knowledgestudent)\ **Post date:** [March 11, 2009, 9:53am UTC](https://community.asterisk.org/t/iax-sip-asterisk/25633/7 "2009-03-11T09:53:03Z")
Hi, thanks mbruni for replying I have used that example before, it initiates the call but then I hear no voice, I mean that I can’t hear the other party (the called) after she answers as there is no other party !!! so the Question is what should I do after the initiating of the call to handle the call, regards,