<div dir="ltr">That's right, I used a 'l' instead of '1'! Thank you.<br>I've made the modification on extension.conf (there's nothing to change on the sip.conf) but the call can not go through...is there another file to modify?<br>
<br>The new outcome is:<br><br> -- Executing Dial("SIP/105-6298", "SIP/l03") in new stack<br>Jan 1 00:54:38 WARNING[606]: chan_sip.c:1407 create_addr: No such host: l03<br>Jan 1 00:54:38 NOTICE[606]: app_dial.c:764 dial_exec: Unable to create channel<br>
of type 'SIP'<br> == Everyone is busy/congested at this time<br> -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 103<br> -- Timeout on SIP/105-6298<br> == CDR updated on SIP/105-6298<br>
-- Executing Goto("SIP/105-6298", "#|1") in new stack<br> -- Goto (default,#,1)<br> -- Executing Playback("SIP/105-6298", "demo-thanks") in new stack<br>Jan 1 00:54:48 WARNING[606]: file.c:475 ast_openstream: File demo-thanks does n<br>
ot exist in any format<br>Jan 1 00:54:48 WARNING[606]: file.c:787 ast_streamfile: Unable to open demo-tha<br>nks (format ulaw): No such file or directory<br>Jan 1 00:54:48 WARNING[606]: app_playback.c:83 playback_exec: ast_streamfile fa<br>
iled on SIP/105-6298 for demo-thanks<br> -- Executing Hangup("SIP/105-6298", "") in new stack<br> == Spawn extension (default, #, 2) exited non-zero on 'SIP/105-6298'<br> -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 105<br>
<br><br>extension.conf<br><br>[default] <br><br>exten => 103,1,Dial(SIP/103)<br>exten => 105,1,Dial(SIP/105)<br><br>Thank you all!<br>Khaldon<br><br><br><div class="gmail_quote">2008/8/25 Pavel Jezek <span dir="ltr"><<a href="mailto:pavel.jezek@i.cz">pavel.jezek@i.cz</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">you should issue 'sip show peers' command to see, if your phones are<br>
available,<br>
put 'qualify=yes' in your sip.conf<br>
'sip show registry' command is usefull to see if your _asterisk_ is<br>
registered to some another sip server, eg. voip provider..<br>
<font color="#888888">PJ<br>
</font><div><div></div><div class="Wj3C7c"><br>
<br>
<br>
<br>
David Boyd wrote:<br>
> -----Original Message-----<br>
> From: ims.asuser ims.asuser <<a href="mailto:ims.asuser@gmail.com">ims.asuser@gmail.com</a>><br>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
> To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
> Subject: [asterisk-users] Really WEIRD: can register but can not call!<br>
> Date: Mon, 25 Aug 2008 12:26:45 +0200<br>
><br>
> Hi all,<br>
><br>
> I have a very weird problem.<br>
><br>
> I have 2 users (103 and 105). They are able to register in Asterisk, but<br>
> they can not call each other.<br>
><br>
> Hereunder is the outcome:<br>
><br>
> openwrt3*CLI><br>
> -- Registered SIP '103' at <a href="http://192.168.3.9" target="_blank">192.168.3.9</a> port 6127 expires 3600<br>
> -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 103<br>
> openwrt3*CLI><br>
> openwrt3*CLI><br>
> -- Registered SIP '105' at <a href="http://192.168.3.6" target="_blank">192.168.3.6</a> port 8377 expires 3600<br>
> -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 105<br>
> openwrt3*CLI><br>
> openwrt3*CLI><br>
> -- Executing Dial("SIP/105-0ead", "SIP/l03") in new stack<br>
> Jan 1 00:19:26 WARNING[498]: chan_sip.c:1407 create_addr: No such host:<br>
> l03<br>
> Jan 1 00:19:26 NOTICE[498]: app_dial.c:764 dial_exec: Unable to create<br>
> channel<br>
> of type 'SIP'<br>
> == Everyone is busy/congested at this time<br>
> openwrt3*CLI><br>
> openwrt3*CLI><br>
> -- Timeout on SIP/105-0ead<br>
> == CDR updated on SIP/105-0ead<br>
> -- Executing Goto("SIP/105-0ead", "#|1") in new stack<br>
> -- Goto (default,#,1)<br>
> -- Executing Playback("SIP/105-0ead", "demo-thanks") in new stack<br>
> Jan 1 00:19:36 WARNING[498]: file.c:475 ast_openstream: File<br>
> demo-thanks does n<br>
> ot exist in any format<br>
> Jan 1 00:19:36 WARNING[498]: file.c:787 ast_streamfile: Unable to open<br>
> demo-tha<br>
> nks (format ulaw): No such file or directory<br>
> Jan 1 00:19:36 WARNING[498]: app_playback.c:83 playback_exec:<br>
> ast_streamfile fa<br>
> iled on SIP/105-0ead for demo-thanks<br>
> -- Executing Hangup("SIP/105-0ead", "") in new stack<br>
> == Spawn extension (default, #, 2) exited non-zero on 'SIP/105-0ead'<br>
><br>
><br>
> The "show sip registry" command shows that no users are registered.<br>
> That's really WEIRD!<br>
><br>
><br>
> Please see the sip.conf and extension.conf files.<br>
><br>
> sip.conf:<br>
><br>
> [general]<br>
> context=default ; Default context for incoming calls<br>
> ;recordhistory=yes ; Record SIP history by default<br>
> ; (see sip history / sip no history)<br>
> ;realm=mydomain.tld ; Realm for digest authentication<br>
> ; defaults to "asterisk"<br>
> ; Realms MUST be globally unique<br>
> according to RF<br>
> ; Set this to your host name or domain<br>
> name<br>
> port=5060 ; UDP Port to bind to (SIP standard port<br>
> is 5060<br>
> bindaddr=x.x.x.x ; IP address to bind to (<a href="http://0.0.0.0" target="_blank">0.0.0.0</a> binds to all)<br>
> srvlookup=yes ; Enable DNS SRV lookups on outbound<br>
> calls<br>
> ; Note: Asterisk only uses the first<br>
> host<br>
> ; in SRV records<br>
> ; Disabling DNS SRV lookups disables the<br>
> ; ability to place SIP calls based on<br>
> domain<br>
> ; names to some other SIP users on the<br>
> Internet<br>
><br>
> [103] ;<br>
> ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!<br>
> type=friend<br>
> username=103 ; Authorization User dans X-Lite<br>
> secret=1234<br>
> callerid="Philippe" <103> ; nom et numéro affichés dans le X-Lite<br>
> appelé l<br>
> context=default<br>
> host=dynamic<br>
> nat=no ; X-Lite is behind a NAT router<br>
> canreinvite=no ; Typically set to NO if behind NAT<br>
> disallow=all ; désactive tous les codages sauf ceux spécifiés<br>
> ci-aprè<br>
> allow=gsm ; GSM consumes far less bandwidth than<br>
> ulaw<br>
> allow=ulaw<br>
> allow=alaw<br>
><br>
> [105] ;<br>
> ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!<br>
> type=friend<br>
> username=105 ; Authorization User dans X-Lite<br>
> secret=1234<br>
> callerid="Khalid" <105> ; nom et numéro affichés dans le X-Lite<br>
> appelé lor<br>
> context=default<br>
> host=dynamic<br>
> nat=no ; X-Lite is behind a NAT router<br>
> canreinvite=no ; Typically set to NO if behind NAT<br>
> disallow=all ; désactive tous les codages sauf ceux spécifiés<br>
> ci-aprè<br>
> allow=ulaw<br>
> allow=alaw<br>
><br>
><br>
> extension.conf:<br>
><br>
> [default] ; context par défaut des utilisateurs SIP répertoriés<br>
> dans sip.c<br>
><br>
><br>
> exten => 103,1,Dial(SIP/l03)<br>
> exten => 105,1,Dial(SIP/l05)<br>
><br>
><br>
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><br>
><br>
><br>
> Your extensions are listed as SIP/l03 and SIP/l05 and should be SIP/103 and SIP/105. Plus a problem with some recorded files.<br>
><br>
><br>
> Regards,<br>
> Dave<br>
><br>
><br>
><br>
><br>
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