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Hi Salvatore,<br />
Check your VIOP provider to see what codecs they support, and match then to what you're using in Elastix. I suspect you're using one they don't support or is not enabled on their end. Typically with VOIP providers you can select the codecs you prefer.
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Hi Salvatore,
Check your VIOP provider to see what codecs they support, and match then to what you're using in Elastix. I suspect you're using one they don't support or is not enabled on their end. Typically with VOIP providers you can select the codecs you prefer.

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I do not think it is a codec problem, use this trunk with the same configuration on another voip &nbsp;(with trixbox) and have no problems, also the trunk is registered correctly with the provider sip.<br />
<br />
-<br />
&nbsp;Salvatore.
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I do not think it is a codec problem, use this trunk with the same configuration on another voip  (with trixbox) and have no problems, also the trunk is registered correctly with the provider sip.

-
 Salvatore.

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because when I removed this option I can call.
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because when I removed this option I can call.

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<div class="content wysiwyg-content">
Hi,<br />
in Elastix I can not make a call over a trunk, I configured the outbound route as attached image, however when I try aa call on voip phone Snom I see &quot;Not Acceptable Here&quot; and the asterisk log I do not see anything.<br />
The trunk is sip type and is properly registered.<br />
Thanks.<br />
<br />
-<br />
&nbsp;Salvatore.<br />
<a href="https://filedb.experts-exchange.com/incoming/2013/06_w26/662786/out-route.JPG" target="_blank" class="file-inline" title="outbound route (55 KB)"><svg xmlns="http://www.w3.org/2000/svg" viewBox="0 0 512 512"><path d="M67.508 468.467c-58.005-58.013-58.016-151.92 0-209.943l225.011-225.04c44.643-44.645 117.279-44.645 161.92 0 44.743 44.749 44.753 117.186 0 161.944l-189.465 189.49c-31.41 31.413-82.518 31.412-113.926.001-31.479-31.482-31.49-82.453 0-113.944L311.51 110.491c4.687-4.687 12.286-4.687 16.972 0l16.967 16.971c4.685 4.686 4.685 12.283 0 16.969L184.983 304.917c-12.724 12.724-12.73 33.328 0 46.058 12.696 12.697 33.356 12.699 46.054-.001l189.465-189.489c25.987-25.989 25.994-68.06.001-94.056-25.931-25.934-68.119-25.932-94.049 0l-225.01 225.039c-39.249 39.252-39.258 102.795-.001 142.057 39.285 39.29 102.885 39.287 142.162-.028A739446.174 739446.174 0 0 1 439.497 238.49c4.686-4.687 12.282-4.684 16.969.004l16.967 16.971c4.685 4.686 4.689 12.279.004 16.965a755654.128 755654.128 0 0 0-195.881 195.996c-58.034 58.092-152.004 58.093-210.048.041z"/></svg>out-route.JPG</a>
</div>
</div>


out-route.JPG

Hi,
in Elastix I can not make a call over a trunk, I configured the outbound route as attached image, however when I try aa call on voip phone Snom I see "Not Acceptable Here" and the asterisk log I do not see anything.
The trunk is sip type and is properly registered.
Thanks.

-
 Salvatore.

Outbound Route with Elastix

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

IP Telephony

Linux is a UNIX-like open source operating system with hundreds of distinct distributions, including: Fedora, openSUSE, Ubuntu, Debian, Slackware, Gentoo, CentOS, and Arch Linux. Linux is generally associated with web and database servers, but has become popular in many niche industries and applications.

Linux

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).