<br>Hi klaus,<br><br style="color: rgb(51, 51, 255);"><span style="color: rgb(51, 51, 255);">
1. AMR codec is not installed<br><br><font color="#000000">for the above when I tried to install amr patch from <a href="http://sip.fontventa.com">http://sip.fontventa.com</a>, however because of codec_amr.so module my asterisk does not start in background. only in console it started that too if I uncomment the below lines in "codec.conf" it gives segmentation fault.<br>
;[amr]<br>;octet-aligned=1<br><br>In asterisk Cli when I use commend "core show codec". I find amr codec there.<br><br><br><br><br><br><br> INT BINARY HEX TYPE NAME DESC<br>--------------------------------------------------------------------------------<br>
<br><br>8192 (1 << 13) (0x2000) audio amr (AMR NB)<br><br><br><br></font></span><span style="color: rgb(51, 51, 153);">
2. user information layer 1 is not "h.223 and h.245" (=h324m)</span><br style="color: rgb(51, 51, 153);"><span style="color: rgb(51, 51, 153);"><br><font color="#000000">I don't have knowledge about user informatio layer 1 as in my case it is unknown could you u let me know how to set up.<br>
<br><br><span style="color: rgb(51, 51, 153);">3 </span></font>"no route to destination".<br><br>I need to figure out the above message.<br><br><br>Klaus could you please wht I need to do to fix the first two issue.<br>
<br><br>Thanks,<br><br><br>Jack<br><br><br><br><br></span>
<br><br><br><br><div class="gmail_quote">On Fri, Mar 13, 2009 at 2:17 PM, Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at">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;">
looks like<br>
1. AMR codec is not installed<br>
2. user information layer 1 is not "h.223 and h.245" (=h324m)<br>
3. the switch answer with "no route to destination"<br>
<br>
klaus<br>
<br>
jack nicolson schrieb:<br>
<div><div></div><div class="h5">> Hi Klaus,<br>
><br>
> Below is the response which I am getting while try to make video out<br>
> bound call,<br>
><br>
><br>
> -- Attempting call on Local/9467000603@3G for 1@video:1 (Retry 1)<br>
> -- Executing [9467000603@3G:1] System("Local/9467000603@3G-8c88,2",<br>
> "sh /etc/asterisk/video_callback.sh 9467000603 CALLING") in new stack<br>
> -- Executing [9467000603@3G:2]<br>
> h324m_call("Local/9467000603@3G-8c88,2", "9467000603@3GV") in new stack<br>
> [Mar 13 11:56:14] WARNING[16543]: channel.c:2781 set_format: Unable to<br>
> find a codec translation path from slin to amr<br>
> [Mar 13 11:56:14] WARNING[16543]: app_h324m.c:1143 app_h324m_call:<br>
> app_h324m_call: Unable to set read format to AMR-NB!<br>
> [Mar 13 11:56:14] WARNING[16543]: channel.c:2781 set_format: Unable to<br>
> find a codec translation path from slin to amr<br>
> [Mar 13 11:56:14] WARNING[16543]: app_h324m.c:1145 app_h324m_call:<br>
> app_h324m_call: Unable to set read format to AMR-NB!<br>
> -- Executing [9467000603@3GV:1] Set("Local/9467000603@3GV-daad,2",<br>
> "CHANNEL(transfercapability)=VIDEO") in new stack<br>
> -- Executing [9467000603@3GV:2] Dial("Local/9467000603@3GV-daad,2",<br>
> "Zap/g0/9467000603") in new stack<br>
> -- Making new call for cr 32785<br>
> -- Requested transfer capability: 0x18 - VIDEO<br>
> > Protocol Discriminator: Q.931 (8) len=44<br>
> > Call Ref: len= 2 (reference 17/0x11) (Originator)<br>
> > Message type: SETUP (5)<br>
> > [04 03 88 90 bf]<br>
> > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer<br>
> capability: Unrestricted digital information (8)<br>
> > Ext: 1 Trans mode/rate: 64kbps,<br>
> circuit-mode (16)<br>
> > User information layer 1: Unknown (63)<br>
> > [18 03 a9 83 81]<br>
> > Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0<br>
> Exclusive Dchan: 0<br>
> > ChanSel: As indicated in following octets<br>
> > Ext: 1 Coding: 0 Number Specified Channel<br>
> Type: 3<br>
> > Ext: 1 Channel: 1 ]<br>
> > [6c 0d 21 80 30 34 30 34 34 33 31 33 30 30 30]<br>
> > Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI:<br>
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)<br>
> > Presentation: Presentation permitted, user<br>
> number not screened (0) '04044313000' ]<br>
> > [70 0b c1 39 34 36 37 30 30 30 36 30 33]<br>
> > Called Number (len=13) [ Ext: 1 TON: Subscriber Number (4) NPI:<br>
> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9467000603' ]<br>
> > [a1]<br>
> > Sending Complete (len= 1)<br>
> q931.c:3092 q931_setup: call 32785 on channel 1 enters state 1 (Call<br>
> Initiated)<br>
> -- Called g0/9467000603<br>
> < Protocol Discriminator: Q.931 (8) len=10<br>
> < Call Ref: len= 2 (reference 17/0x11) (Terminator)<br>
> < Message type: CALL PROCEEDING (2)<br>
> < [18 03 a9 83 81]<br>
> < Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0<br>
> Exclusive Dchan: 0<br>
> < ChanSel: As indicated in following octets<br>
> < Ext: 1 Coding: 0 Number Specified Channel Type: 3<br>
> < Ext: 1 Channel: 1 ]<br>
> -- Processing IE 24 (cs0, Channel Identification)<br>
> q931.c:3641 q931_receive: call 32785 on channel 1 enters state 3<br>
> (Outgoing call Proceeding)<br>
> -- Zap/1-1 is proceeding passing it to Local/9467000603@3GV-daad,2<br>
> < Protocol Discriminator: Q.931 (8) len=9<br>
> < Call Ref: len= 2 (reference 17/0x11) (Terminator)<br>
> < Message type: DISCONNECT (69)<br>
> < [08 02 82 83]<br>
> < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0<br>
> Location: Public network serving the local user (2)<br>
> < Ext: 1 Cause: No route to destination (3), class =<br>
> Normal Event (0) ]<br>
> -- Processing IE 8 (cs0, Cause)<br>
> q931.c:3784 q931_receive: call 32785 on channel 1 enters state 12<br>
> (Disconnect Indication)<br>
> -- Channel 0/1, span 1 got hangup request, cause 3<br>
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,<br>
> peerstate Disconnect Request<br>
> q931.c:2925 q931_release: call 32785 on channel 1 enters state 19<br>
> (Release Request)<br>
> > Protocol Discriminator: Q.931 (8) len=9<br>
> > Call Ref: len= 2 (reference 17/0x11) (Originator)<br>
> > Message type: RELEASE (77)<br>
> > [08 02 81 83]<br>
> > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0<br>
> Location: Private network serving the local user (1)<br>
> > Ext: 1 Cause: No route to destination (3), class =<br>
> Normal Event (0) ]<br>
> -- Hungup 'Zap/1-1'<br>
> == Everyone is busy/congested at this time (1:0/0/1)<br>
> == Auto fallthrough, channel 'Local/9467000603@3GV-daad,2' status is<br>
> 'CHANUNAVAIL'<br>
> == Auto fallthrough, channel 'Local/9467000603@3G-8c88,2' status is<br>
> 'UNKNOWN'<br>
> [Mar 13 11:56:14] NOTICE[16542]: pbx_spool.c:341 attempt_thread: Call<br>
> failed to go through, reason (1) Hangup<br>
> < Protocol Discriminator: Q.931 (8) len=5<br>
> < Call Ref: len= 2 (reference 17/0x11) (Terminator)<br>
> < Message type: RELEASE COMPLETE (90)<br>
> q931.c:3724 q931_receive: call 32785 on channel 1 enters state 0 (Null)<br>
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null<br>
> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null<br>
><br>
><br>
><br>
><br>
> Could you help me to fix this issue.<br>
><br>
><br>
> Thanks,<br>
><br>
> Jack<br>
><br>
><br>
> On Fri, Mar 13, 2009 at 9:44 AM, jack nicolson<br>
</div></div><div class="im">> <<a href="mailto:jack.nicolson123@gmail.com">jack.nicolson123@gmail.com</a> <mailto:<a href="mailto:jack.nicolson123@gmail.com">jack.nicolson123@gmail.com</a>>> wrote:<br>
><br>
> Please ignore the previous message you are right Klaus. There is no<br>
> zap channel i need to start them.<br>
><br>
><br>
> Thanks<br>
><br>
> Jack<br>
><br>
><br>
> On Fri, Mar 13, 2009 at 9:20 AM, jack nicolson<br>
</div><div class="im">> <<a href="mailto:jack.nicolson123@gmail.com">jack.nicolson123@gmail.com</a> <mailto:<a href="mailto:jack.nicolson123@gmail.com">jack.nicolson123@gmail.com</a>>> wrote:<br>
><br>
> Hi Klaus,<br>
><br>
> My normal audio outbound call works fine.only problem with video<br>
> outbound call.<br>
><br>
> My asterisk box is connected to E1 line through Digium card.<br>
><br>
><br>
> Thanks,<br>
><br>
> Jack<br>
><br>
><br>
> On Thu, Mar 12, 2009 at 8:30 PM, Klaus Darilion<br>
> <<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a><br>
</div><div class="im">> <mailto:<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>>> wrote:<br>
><br>
><br>
><br>
> jack nicolson schrieb:<br>
> > -- Executing [9467000603@3GV:2]<br>
> Dial("Local/9467000603@3GV-0624,2",<br>
> > "Zap/g0/9467000603") in new stack<br>
> > [Mar 12 19:15:36] WARNING[9206]: channel.c:3027<br>
> ast_request: No channel<br>
> > type registered for 'Zap'<br>
> > [Mar 12 19:15:36] WARNING[9206]: app_dial.c:1183<br>
> dial_exec_full: Unable<br>
> > to create channel of type 'Zap' (cause 66 - Channel not<br>
> implemented)<br>
> > == Everyone is busy/congested at this time (1:0/0/1)<br>
><br>
><br>
> There is no Zap channel. How are you connected to the PSTN?<br>
> Zap? Dahdi?<br>
> Are normal audio calls work?<br>
><br>
> klaus<br>
><br>
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><br>
><br>
><br>
><br>
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