[asterisk-users] No remote address on RTP instance - On Ringing
Hello Everyone, I have a new problem where when placing the call, asterisk will automatically go into music on hold until the call is connected (ie, no ringing). It was kind of confusing because sometimes `SESSION PROGRESS` takes longer than others, during this time we are in MOH. The call does eventually connect and the MOH stops. When debugging I saw the following debug message: [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame This is a straight SIP channel. No DAHDI. Your Help is Greatly Appreciated. This is a new one for me :) Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Nick Cameo wrote: There is two way audio, it's just during ringing that this happens. If you can put the SIP signaling and Asterisk console output up somewhere then we can have a better idea of what Asterisk is being told, and what it is doing. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Does your Dial() command include the m option? jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
There is two way audio, it's just during ringing that this happens. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Hello Jg, Thank your for your response. No m option on dial. I think it's a RTP relay issue however, do not know how to diagnose the SDP payload. Any help would be appreciated. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Nick Cameo wrote: I ran a test call with trace can be found here: http://pastebin.com/f8MuxaFV I also wanted to mention that yes we have * setup with disallow=all and allow=g729 for testing, maybe permanently if we can successfully setup G729 pass through. That being said, the same problem is still there using allow=ulaw,alaw,g729. As someone else mentioned the 'm' option is definitely in place. Asterisk is doing exactly as it is told. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Maybe the ringtone from downstream is not reaching asterisk, and thus a2billing is appending the `m` to the dial command? With digital systems (starting with ISDN, or so), ringing is signaled, or indicated. The ringtone is produced locally, either by the PBX or by the SIP phone itself. Since you do get the invitation, everything is fine. If you really can't remove the m, you could still use an audio file with a funny ringtone and stuff this into an moh class. Dirty, but it will work. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Yeah of course. Still digging into it :). Will post the solution if I find it. a2billing forum takes for ever to answer... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
I have no idea where the `m` is coming from. I even looked into the A2Billing script. Still digging -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Yes of course, I just did not want to overwhelm you guys with SIP trace. Before that though, I realized something: [Sep 10 12:03:30] WARNING[8178]: res_musiconhold.c:802 set_moh_exec: SetMusicOnHold application is deprecated and will be removed. Use Set(CHANNEL(musicclass)=...) instead -- AGI Script Executing Application: (DIAL) Options: (SIP/VTrunk/19042572451,60,HRrL(24:61000:3)m) There is that `m` option that jg was referring to. However, in a2billing, I have made sure there is no `m` option in the `Dial Command Params`: ,60,HRrL(%timeout%:61000:3). The extension for the entry does not include the option either: exten = 1000,1,Answer exten = 1000,n,Wait(1) exten = 1000,n,AGI(a2billing.php) exten = 1000,n,Wait(1) exten = 1000,n,Hangup Will run a test call with trace right now. Kind Regards, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Hope this helps someone save a day of running around.
So my issue was with a2billing. The warning `No remote address on RTP
instance '0xb6d16a28' so dropping frame`
was not related to the music on hold coming on during ringing.
The Problem:
We have a script that loads rates into `a2billing.cc_ratecard` table. The
problem field was `musiconhold`.
Loading the field with `null`, causes this conditional statement in
`lib/asterisk/agi-bin/lib/Class.RateEngine.php` to fire:
if (strlen($musiconhold) 0 $musiconhold != selected) {
$dialparams .= m;
This added the m to the DIAL command.
The Solution:
Make sure cc_ratecard.musiconhold = '';
Thank you all for your help. I can rest now :).
Nick from Toronto.
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Re: [asterisk-users] No remote address on RTP instance - On Ringing
Did you the a2billing settings for a music on hold setting I remember seeing some setting -Original Message- From: Nick Cameo sym...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 10 Sep 2013 12:46:54 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No remote address on RTP instance - On Ringing I have no idea where the `m` is coming from. I even looked into the A2Billing script. Still digging -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Nick Cameo wrote: Yeah!!! The Dial command setting: http://forum.asterisk2billing.org/viewtopic.php?f=2t=1704 I know this is not an a2billing mailing list, and I am sorry however, I do think that the No remote address on RTP instance may have something to do with it. Maybe the ringtone from downstream is not reaching asterisk, and thus a2billing is appending the `m` to the dial command? I'm sorry if this sounds crazy... :) The No remote address on RTP instance occurs when the RTP stack has been told to send media but it does not yet have address information to know where to send it. As for your theory... not possible. Stuff can't react in that fashion and change dial options mid-flow. Your output in a previous email also showed the 'm' being specified to Dial, from an AGI. That's where you should focus. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Oh scandalous Instead of playing the MOH, I would like to play the ringtone that is on the machine. Ummm, where is it? :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
I ran a test call with trace can be found here: http://pastebin.com/f8MuxaFV I also wanted to mention that yes we have * setup with disallow=all and allow=g729 for testing, maybe permanently if we can successfully setup G729 pass through. That being said, the same problem is still there using allow=ulaw,alaw,g729. Thanks in Advance, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
i have used a2billing some time ago maybe there is somthing new . you can try shoot up loglevel to 4 and see the verbose of agi that may give you some hint. On Tue, Sep 10, 2013 at 7:34 PM, jg webaccou...@jgoettgens.de wrote: Maybe the ringtone from downstream is not reaching asterisk, and thus a2billing is appending the `m` to the dial command? With digital systems (starting with ISDN, or so), ringing is signaled, or indicated. The ringtone is produced locally, either by the PBX or by the SIP phone itself. Since you do get the invitation, everything is fine. If you really can't remove the m, you could still use an audio file with a funny ringtone and stuff this into an moh class. Dirty, but it will work. jg -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
Yeah!!! The Dial command setting: http://forum.asterisk2billing.org/viewtopic.php?f=2t=1704 I know this is not an a2billing mailing list, and I am sorry however, I do think that the No remote address on RTP instance may have something to do with it. Maybe the ringtone from downstream is not reaching asterisk, and thus a2billing is appending the `m` to the dial command? I'm sorry if this sounds crazy... :) N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
