Hi Klaus,<br><br>Its working now I can see the video but audio is not coming. In asterisk log its saying. <br><br>"NOTICE[5485]: channel.c:2272 __ast_read: Dropping incompatible voice frame on Local/1@vxml-d45b,1 of format ulaw since our native format has changed to unknown"<br>
<br>my video file codec are as below.<br>mp4info version 1.5.0.1<br>xxxx.3gp:<br>Track Type Info<br>1 audio G.711 uLaw, 3.640 secs, 64 kbps, 8000 Hz<br>2 hint Payload PCMU for track 1<br>3 video H.263, 1.234 secs, 200 kbps, 176x144 @ 29.983793 fps<br>
4 hint Payload H263-2000 for track 3<br> Metadata Tool: mp4creator 1.5.0.1<br><br><br><br>Another issue is I am trying to connect my asterisk server locally with my nokia N81 (version:v 10.0.0.35 RM-223) using wlan but I am getting registration failure error.<br>
<br>Below are my sip setting for the mobile as many mentioned in different sites that it works.<br>profilename:SIPtest.<br>serviceprofile:IETF<br>Default access point: <mywifipoint><br>public user name: xxxx@<asteriskserverIP><br>
use compression: No<br>Registration: Always on<br>Use Security: No<br> Proxy server<br>proxy server address:192.168.x.x<asterisk server IP><br>Realm: asterisk<br>UserName:xxxx<br>Password:xxxx123<br>Allow loose routing: yes<br>
Transport type: UDP<br>Port:5060<br><br>Registrar server address<br><br>registrar server address:192.168.x.x<asterisk serverIP><br>
Realm: asterisk<br>
UserName:xxxx<br>
Password:xxxx123<br>
Allow loose routing: yes<br>
Transport type: UDP<br>
Port:5060<br><br>then I created profile in internet telephone with the above sip setting.<br><br>Do I need to install updates for the mobile and software like <u>SIP</u>VOIP setting software.<br><br>I am sorry if the above issue not related to the forum. please discard it.<a href="http://sw.nokia.com/id/d61bd0ec-1304-45dd-9283-63d631cb86b1/SIP_VoIP_Settings_v1_2_en.sis"><span style="font-size: 1.1em;"></span></a><br>
<br>Thanks,<br><br>Jack<br><br><br><br><br><div class="gmail_quote">On Thu, Mar 5, 2009 at 3:49 PM, Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">post the asterisk console during an incoming call with<br>
> pri debug span 1 (or whatever span you use)<br>
> set verbose 9<br>
> set debug 9<br>
<br>
<br>
regards<br>
klaus<br>
<br>
<br>
jack nicolson schrieb:<br>
<div><div></div><div>> Hi,<br>
><br>
> Just now in my city 3g service is implemented so I started looking<br>
> for IVVR stuff. Anyway my asterisk video issue is.<br>
><br>
><br>
> I followed your instruction for video call as given below.<br>
><br>
> <a href="http://www.allasterisk.com/lists/asterisk-video@lists.digium.com/2007-09/msg00125.html" target="_blank">http://www.allasterisk.com/lists/asterisk-video@lists.digium.com/2007-09/msg00125.html</a><br>
><br>
><br>
> However I am able to connect to my server the command<br>
> h324m_gw_answer when executed. it remain in that state only and not<br>
> proceeding to next step. after few minutes I need to hang the call.<br>
><br>
><br>
> below is my asterisk log<br>
><br>
> Going to extension s|1 because of immediate=yes<br>
> -- Accepting call from '9xxxxxxxxx' to 's' on channel 0/11, span 1<br>
> -- Executing [s@incoming:1] Answer("Zap/11-1", "") in new stack<br>
> -- Executing [s@incoming:2] h324m_gw("Zap/11-1", "1@vxml") in new<br>
> stack<br>
> -- Executing [1@vxml:1] h324m_gw_answer("Local/1@vxml-768d,2",<br>
> "") in new stack<br>
><br>
><br>
><br>
> I want to know how long it will take for 3G negotiation or is there<br>
> any other issue.<br>
><br>
><br>
> I am connected to PRI E1 line<br>
><br>
> Thanks,<br>
><br>
> Jack<br>
><br>
><br>
</div></div>> ------------------------------------------------------------------------<br>
><br>
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</blockquote></div><br>