<pre><br>Hello Russell Bryant,<br><br>I am not able to find out which function and configuration I need to add in extenstion.conf for SMS.<br>The exercise what I am doing is trying to send message for SIP Number 1000 to Sip Number 2000.<br>
Based on this function i want to make a CLI command that will do similar thing as dialplan function.<br>The CLI argumant is like to from "message" where to is destination SIP number and from is sender SIP number.<br>
<br>Please help me for SMS.<br><br><br><br><br> TIA.<br><br>Regards,<br>Vikash<br><br><br><br></pre><br><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
> Hello,<br>
> I tried with asterisk1.6 version for SMS between two softphone but<br>
> understood this version does not support SMS hence I took the asterisk<br>
> branch from below link and used it for asterisk SMS(as this asterisk code<br>
> explained it is working for SMS) but even I am not able to send the message<br>
> from one sip number to another sip number.<br>
> The Link is <a href="http://svn.asterisk.org/svn/asterisk/team/russell/messaging/" target="_blank">http://svn.asterisk.org/svn/asterisk/team/russell/messaging/</a><br>
> <a href="https://reviewboard.asterisk.org/r/1042/" target="_blank">https://reviewboard.asterisk.org/r/1042/</a><br>
><br>
> The log shows "Unable to parse INFO message from" as shown below and In<br>
> packet capture shows 415 unsupported media type.<br>
> I am using Windows messenger softphone.<br>
><br>
><br>
><br>
> *Please help me how I will able to send SMS in asterisk.Please share it if<br>
> any configuration is needed in sip.conf or in other .conf file for SMS.<br>
> *<br>
> I want to send the sms from 1000 to 2000 where 1000 & 2000 is my SIP phone<br>
> number.<br>
><br>
> == Starting Message/ast_msg_queue at default,sip:2000@asterisk,1 failed so<br>
> falling back to exten 's'<br>
> -- Executing [s@default:1] Wait("Message/ast_msg_queue", "1") in new<br>
> stack<br>
> [Jan 1 00:37:49] WARNING[535]: chan_sip.c:18641 handle_request_info:<br>
> Unable to parse INFO message from b27ee315779f40ab<br>
> <a href="mailto:b35c8154bea508d7@172.16.241.42">b35c8154bea508d7@172.16.241.42</a><br>
> . Content<br>
> -- Executing [s@default:2] Answer("Message/ast_msg_queue", "") in new<br>
> stack<br>
> -- Executing [s@default:3] Set("Message/ast_msg_queue",<br>
> "TIMEOUT(digit)=5") in new stack<br>
> [Jan 1 00:37:50] ERROR[514]: pbx.c:3700 ast_func_write: Function TIMEOUT<br>
> not registered<br>
> -- Executing [s@default:4] Set("Message/ast_msg_queue",<br>
> "TIMEOUT(response)=10") in new stack<br>
> [Jan 1 00:37:50] ERROR[514]: pbx.c:3700 ast_func_write: Function TIMEOUT<br>
> not registered<br>
> -- Executing [s@default:5] BackGround("Message/ast_msg_queue",<br>
> "demo-congrats") in new stack<br>
> [Jan 1 00:37:50] WARNING[514]: channel.c:5078 set_format: Unable to find a<br>
> codec translation path from (nothing) to (gs<br>
> m)<br>
> [Jan 1 00:37:50] WARNING[514]: file.c:1004 ast_streamfile: Unable to open<br>
> demo-congrats (format (nothing)): Function no<br>
> t implemented<br>
> [Jan 1 00:37:50] WARNING[514]: pbx.c:9541 pbx_builtin_background:<br>
> ast_streamfile failed on Message/ast_msg_queue for de<br>
> mo-congrats<br>
> -- Executing [s@default:6] BackGround("Message/ast_msg_queue",<br>
> "demo-instruct") in new stack<br>
> [Jan 1 00:37:50] WARNING[514]: channel.c:5078 set_format: Unable to find a<br>
> codec translation path from (nothing) to (gs<br>
> m)<br>
> [Jan 1 00:37:50] WARNING[514]: file.c:1004 ast_streamfile: Unable to open<br>
> demo-instruct (format (nothing)): Function no<br>
> t implemented<br>
> [Jan 1 00:37:50] WARNING[514]: pbx.c:9541 pbx_builtin_background:<br>
> ast_streamfile failed on Message/ast_msg_queue for de<br>
> mo-instruct<br>
> -- Executing [s@default:7] WaitExten("Message/ast_msg_queue", "") in<br>
> new stack<br>
> -- Timeout on Message/ast_msg_queue, going to 't'<br>
> -- Executing [t@default:1] Goto("Message/ast_msg_queue", "#,1") in new<br>
> stack<br>
> -- Goto (default,#,1)<br>
> -- Executing [#@default:1] Playback("Message/ast_msg_queue",<br>
> "demo-thanks") in new stack<br>
> [Jan 1 00:38:00] WARNING[514]: channel.c:5078 set_format: Unable to find a<br>
> codec translation path from (nothing) to (gs<br>
> m)<br>
> [Jan 1 00:38:00] WARNING[514]: file.c:1004 ast_streamfile: Unable to open<br>
> demo-thanks (format (nothing)): Function not<br>
> implemented<br>
> [Jan 1 00:38:00] WARNING[514]: app_playback.c:476 playback_exec:<br>
> ast_streamfile failed on Message/ast_msg_queue for dem<br>
> o-thanks<br>
> -- Executing [#@default:2] Hangup("Message/ast_msg_queue", "") in new<br>
> stack<br>
> == Spawn extension (default, #, 2) exited non-zero on<br>
> 'Message/ast_msg_queue'<br>
><br>
><br>
> *Please help me how can i do sms by using Asterisk.<br>
I just check Mr Russell<br>
Bryant<<a href="https://reviewboard.asterisk.org/users/russell/" target="_blank">https://reviewboard.asterisk.org/users/russell/</a>>has implemented<br>
some simple apps using the pjsua Python module from pjsip<br>
that can send and receive messages sent through Asterisk. Please help me to<br>
know from where i can get this code and procedure.*<br>
<br>
<br>
<br>
<br>
> Regards,<br>
> Vikash<br>
><br>
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Message: 3<br>
Date: Thu, 09 Jun 2011 12:49:17 -0000<br>
From: "Matthew Nicholson" <<a href="mailto:reviewboard@asterisk.org">reviewboard@asterisk.org</a>><br>
Subject: Re: [asterisk-dev] [Code Review] Stop trying to uri_encode<br>
the display name for the caller ID<br>
To: "David Vossel" <<a href="mailto:dvossel@digium.com">dvossel@digium.com</a>>, "Matthew Nicholson"<br>
<<a href="mailto:mnicholson@digium.com">mnicholson@digium.com</a>>, "Russell Bryant" <<a href="mailto:russell@digium.com">russell@digium.com</a>><br>
Cc: , Matthew Nicholson <<a href="mailto:reviewboard@asterisk.org">reviewboard@asterisk.org</a>>, Asterisk<br>
Developers <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:20110609124917.23713.70467@hotblack.digium.com">20110609124917.23713.70467@hotblack.digium.com</a>><br>
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<br>
<br>
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This is an automatically generated e-mail. To reply, visit:<br>
<a href="https://reviewboard.asterisk.org/r/1235/#review3696" target="_blank">https://reviewboard.asterisk.org/r/1235/#review3696</a><br>
-----------------------------------------------------------<br>
<br>
Ship it!<br>
<br>
<br>
If this is just a copy of the ast_escape_quoted function from trunk then this should be fine. You should also copy the unit test from trunk.<br>
<br>
- Matthew<br>
<br>
<br>
On 2011-06-03 10:39:36, jrose wrote:<br>
><br>
> -----------------------------------------------------------<br>
> This is an automatically generated e-mail. To reply, visit:<br>
> <a href="https://reviewboard.asterisk.org/r/1235/" target="_blank">https://reviewboard.asterisk.org/r/1235/</a><br>
> -----------------------------------------------------------<br>
><br>
> (Updated 2011-06-03 10:39:36)<br>
><br>
><br>
> Review request for Asterisk Developers, Russell Bryant, Matthew Nicholson, and David Vossel.<br>
><br>
><br>
> Summary<br>
> -------<br>
><br>
> Under SIP pedantic mode, this function was encoding strings meant for the display name into into a URI safe format when this was not specified in the SIP RFCs. I also took the liberty of adding a little commentary.<br>
><br>
> pedantic mode was the default in Asterisk 1.8+<br>
><br>
><br>
> This addresses bug 18298.<br>
> <a href="https://issues.asterisk.org/jira/browse/18298" target="_blank">https://issues.asterisk.org/jira/browse/18298</a><br>
><br>
><br>
> Diffs<br>
> -----<br>
><br>
> /branches/1.8/channels/chan_sip.c 321727<br>
> /branches/1.8/include/asterisk/utils.h 321727<br>
> /branches/1.8/main/utils.c 321727<br>
><br>
> Diff: <a href="https://reviewboard.asterisk.org/r/1235/diff" target="_blank">https://reviewboard.asterisk.org/r/1235/diff</a><br>
><br>
><br>
> Testing<br>
> -------<br>
><br>
> A number of different calls with user names including more interesting UTF-8 characters from a variety of characters from a variety of sources including trema (like German umlaut characters), Kanji (Japanese pictographic characters), Sanskrit (phonetic characters for one of India's languages), and Cyrillic (Russian). Whether or not they'll display properly depends on the receiving phone... My Grandstream phone doesn't like the higher end UTF-8 characters, but all of my soft phones read them fine.<br>
><br>
> Testing involved a manually set DAHDI channel to have odd caller ID and an incoming call from another Asterisk Box sending SIP from a phone set to also have a display name with high UTF-8 chars.<br>
><br>
><br>
> Thanks,<br>
><br>
> jrose<br>
><br>
><br>
<br>
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Message: 4<br>
Date: Thu, 09 Jun 2011 07:52:28 -0500<br>
From: Russell Bryant <<a href="mailto:russell@digium.com">russell@digium.com</a>><br>
Subject: Re: [asterisk-dev] Help Need Asterisk SMS code and procedure<br>
To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:4DF0C20C.20208@digium.com">4DF0C20C.20208@digium.com</a>><br>
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<br>
On 06/09/2011 01:19 AM, warrior back wrote:<br>
>> == Starting Message/ast_msg_queue at default,sip:2000@asterisk,1 failed so<br>
>> falling back to exten 's'<br>
>> -- Executing [s@default:1] Wait("Message/ast_msg_queue", "1") in new<br>
>> stack<br>
>> [Jan 1 00:37:49] WARNING[535]: chan_sip.c:18641 handle_request_info:<br>
>> Unable to parse INFO message from b27ee315779f40ab<br>
>> <a href="mailto:b35c8154bea508d7@172.16.241.42">b35c8154bea508d7@172.16.241.42</a><br>
>> . Content<br>
>> -- Executing [s@default:2] Answer("Message/ast_msg_queue", "") in new<br>
>> stack<br>
<br>
This log shows the message making it to Asterisk. However, it's going<br>
in to an extension written to handle normal calls. You need to modify<br>
the dialplan to expect to receive messages and route them however you<br>
want to.<br>
<br>
Also, this code is in trunk now. Just use trunk, not my branch.<br>
<br>
--<br>
Russell Bryant<br>
Digium, Inc. | Engineering Manager, Open Source Software<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
<a href="http://www.digium.com" target="_blank">www.digium.com</a> -=- <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a> -=- <a href="http://blogs.asterisk.org" target="_blank">blogs.asterisk.org</a><br>
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</blockquote><br></div><br>