<div>
This type of problem occurs because, in SIP, the media path tries to go direct from one end point to the other even where the SIP messages are taking a longer path. When one of the end-points is behind a NAT router or firewall then it can be impossible for the media path to connect directly. Even in situations where both Asterisk servers are directly connected to the Internet, you can get the problem if the phone you are using is a SIP phone on the LAN. Try adding option w to the Dial command for you local extension as this forces Asterisk to stay in the media path. (It is also supposed to be possible using the SIP option &quot;canreinvite=no&quot;, but I find option w is more reliable):<br />
exten =&gt; 1136,1,Dial(SIP/USER|30|w)<wbr /> 
</div>


<div>
okie thankzz for those who reply ...<br />
<br />
i got success by using IAX trunk instead of using SIP .... <br />
<br />
canda:<br />
<br />
[infospanca]<br />
auth=md5<br />
type=friend<br />
username=infospanca<br />
secret=infospanca<br />
host=dynamic<br />
disallow=all<br />
allow=ulaw<br />
allow=alaw<br />
allow=ilbc<br />
allow=gsm<br />
<br />
pakistan<br />
exten =&gt; _1XXX,1,Dial(IAX2/infospan<wbr />ca:infospa<wbr />nca@203.21<wbr />5.190.50/$<wbr />{EXTEN})<br />
hope this will help others two . :)<br />
thankzz <br />

</div>


<div>
<div class="content wysiwyg-content">
Hi ,<br />
i want to make call between two asterisk server. Both the servers are on different locations. when i tried to call asterisk to asterisk server on the same network it goes fine But when i try the same on the server on the other end the call established but having either one way voice or no voice . my first asterisk server is in pakistan and other is in canada . i have exten =&gt; 1136,1,Dial(SIP/${EXTEN}@c<wbr />anada) in one server and on the other server i have exten =&gt; 1136,1,Dial(SIP/USER) . now i got either one way voice or no voice when i dial 1136. so need help on that why we got such kind of issue.<br />
looking forward for good reply<br />
thankzz
</div>
</div>


Hi ,
i want to make call between two asterisk server. Both the servers are on different locations. when i tried to call asterisk to asterisk server on the same network it goes fine But when i try the same on the server on the other end the call established but having either one way voice or no voice . my first asterisk server is in pakistan and other is in canada . i have exten => 1136,1,Dial(SIP/${EXTEN}@canada) in one server and on the other server i have exten => 1136,1,Dial(SIP/USER) . now i got either one way voice or no voice when i dial 1136. so need help on that why we got such kind of issue.
looking forward for good reply
thankzz

One way voice in asterisk to asterisk server

Networking is the process of connecting computing devices, peripherals and terminals together through a system that uses wiring, cabling or radio waves that enable their users to communicate, share information and interact over distances. Often associated are issues regarding operating systems, hardware and equipment, cloud and virtual networking, protocols, architecture, storage and management.

Networking

The variety of Linux distributions creates myriad issues relating to configuration and operations when computers are networked, not the least of which is the use of various network management applications, some of which are included with specific distributions, while others are standalone applications.

Linux Networking

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).