Summary:ASTERISK-16601: chan_multicast_rtp.so MulticastRTP no audio when using Page() App
Reporter:Stacy Vinson (svinson)Labels:
Date Opened:2010-08-20 16:38:37Date Closed:2012-03-28 12:13:21
Versions:Frequency of
is related toASTERISK-19598 Garbled audio using Page app and MulticastRTP channel
Environment:Attachments:( 0) issue_0017896_log
( 1) local.cfg
( 2) mrtp-dial-cmd.pcap
( 3) mrtp-page-cmd.pcap
( 4) MulticastRTP-CiscoSpanPort.pcap
( 5) MulticastRTP-Debug.txt
( 6) MulticastRTP-Page.wav
( 7) server.cfg
Description:When I use the Page() app with the MulticastRTP channel the phone answers but i don't get any audio. when I use the Dial() command the audio works fine.
the Page() command works fine with the SIP channel. just not the MulticastRTP channel. let me know what i can do to help debug this issue. Thanks,
Comments:By: Paul Belanger (pabelanger) 2010-08-20 17:27:05

We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:


By: Leif Madsen (lmadsen) 2010-08-24 13:06:47

We're probably going to need to see your dialplan you're using, along with the configuration of the sip.conf for the peer and the general section, the device you're using, and a PCAP capture (from tshark or wireshark).

I looked at the debug and I'm not sure it's giving anything particularly useful.

By: Stacy Vinson (svinson) 2010-08-24 21:40:56

my sip.conf is defaults other than:


(and the phone config)

callerid="test" <4005>


(not working)
exten => 4000,1,Answer
exten => 4000,n,Set(CALLERID(name)={Page:${CALLERID(name) })
exten => 4000,n,Set(TIMEOUT(digit)=5)
exten => 4000,n,SIPAddHeader(Call-Info: Answer-After=0)  ;  Grandstream, Snoms
exten => 4000,n,SIPAddHeader(Alert-Info: info=alert-autoanswer) ;AASTRA/poly
exten => 4000,n,Page(MulticastRTP/basic/
exten => 4000,n,Hangup

exten => 4200,1,Answer
exten => 4200,n,Set(CALLERID(name)={Page:${CALLERID(name) })
exten => 4200,n,Set(TIMEOUT(digit)=5)
exten => 4200,n,SIPAddHeader(Call-Info: Answer-After=0)  ;  Grandstream, Snoms
exten => 4200,n,SIPAddHeader(Alert-Info: info=alert-autoanswer) ;AASTRA/poly
exten => 4200,n,Page(SIP/4005)
exten => 4200,n,Hangup

exten => 4300,1,Dial(MulticastRTP/basic/
exten => 4300,n,Hangup

also take a look at the pcap files, in the file mrtp-dial-cmd.pcap you will see data in the data field for the multicast address. but in the file mrtp-page-cmd.pcap using the page command the data filed is all ff's

Thanks, for your help.

By: Stacy Vinson (svinson) 2010-08-24 21:53:26

also the phone i'm testing the multicast on is a AASTRA 57i with firmware, and i get the same result when makeing the page call from a SIP, skinny (7912) or MGCP phone.

By: Russell Bryant (russell) 2010-09-09 16:37:29

Can you tell us what options you have set on the phone for Multicast RTP support?  I was just testing this recently and it was working fine for me.  The tricky part was getting the phones configured properly.

By: Stacy Vinson (svinson) 2010-09-09 19:06:18

I have attached two files that are the aastra config files.
the only options on the phone are Multicast-ip and port. i dont't see how a phone configuration would stop the server from sending multicast audio. Take a look at the attached pcap files. also the phone works fine with multicast when using the dial command. I would not think there should be a difference when using the page() or dial() command.

exten => 4300,1,Dial(MulticastRTP/basic/  works fine
exten => 4000,1,Page(MulticastRTP/basic/  phone answers but no audio

Thanks for your help.

By: Russell Bryant (russell) 2010-09-13 06:35:04

Thanks for the clarification.  I read through this too fast at first.

If Dial() works but Page() doesn't, it's likely that MeetMe() isn't working properly (since Page() uses MeetMe() internally).  Try MeetMe() directly and see if it works for you.

By: Stacy Vinson (svinson) 2010-09-13 13:12:55

MeetMe() app work fine for me. jsut as a note i'm using a Wildcard TDM400P REV H Board 5 for timeing, not sure if that is still needed. my guess is that their is a problem with meetme passing the audio to chan_multicast_rtp.so, but i'm not sure of a good way to debug this. also i can say that I have the same problem with asterisk-1.8.0-beta5.


By: Jean Francois (jeenux) 2010-11-19 20:34:04.000-0600

I'm having the same issue and thought I could add some information on the tests I did.

I am using Aastra 9143i and Aastra 6753i phones with the latest Asterisk release (1.8.0).

These are the commands I tried:

exten => 8996,1,MeetMe(8996,d)
exten => 8997,1,Dial(MulticastRTP/basic/
exten => 8998,1,Page(MulticastRTP/basic/

MeetMe works fine as we can dial 8996 with two phones and talk to each other. It also works using Dial command but not Page. I then tried using the MulticastRTP channel with a Cisco 7941 phone listening on the specified multicast address but it doesn't work either (only Dial works).

One last thing I tried was to dial the multicast address with X-Lite and then start music on my computer. I can hear a chirp of the music about every two seconds. So sound is partly going to the phones. I changed the network switch and also tried on another network (tried it at home) with the same results.

I attached the debug log (MulticastRTP-Debug.txt), a pcap file from the PC port in "spanned" mode behind the Cisco phone (MulticastRTP-CiscoSpanPort.pcap) and also a WAV file while playing the music from X-Lite (MulticastRTP-Page.wav).

Let me know if you need more info,

Thanks for your time

By: Alex Hieronymi (stpaulalex) 2011-01-17 00:50:07.000-0600

Having a similar (possibly same) issue with 1.8.2 using a Cisco SPA-504G and 508G on SIP.  Calling with Dial(MulticastRTP...) works fine, but Page(MulticastRTP...&[any other channel]) causes the phones to receive garbled audio.

I edited the Page() application to use ConfBridge instead of MeetMe to rule out the suggested cause above.  The debug logs show the same codec for native, read, write formats when using Page(MulticastRTP...) - just one channel.  When adding another channel to the page app (in addition to the MulticastRTP channel), the audio becomes garbled on the MulticastRTP channel.  Viewing the MulticastRTP channel info shows:

 NativeFormats: 0x1000 (g722)
   WriteFormat: 0x40 (slin)
    ReadFormat: 0x40 (slin)
WriteTranscode: Yes slin->g722
 ReadTranscode: Yes g722->slin
   Application: ConfBridge
          Data: 1375967666d,m1qw(5)

Since I'm using Cisco/Linksys phones, I'm using the control port parameter in MulticastRTP.

exten => 509,1,Answer()
exten => 509,n,Page(console/dsp&MulticastRTP/linksys/
exten => 509,n,Hangup()

When the above extension is dialed, I console audio sounds fine, but audio from the phones (via Multicast speakerphone paging) is garbled.  I've run the following diagnostic tests to try to determine which part is causing the issue:

; Dial console - works fine
exten => 501,1,Dial(console/dsp)
exten => 501,n,Hangup

; Dial MulticastRTP for phone paging - works fine
exten => 503,1,Dial(MulticastRTP/linksys/
exten => 503,n,Hangup()

; Page the console - works fine
exten => 507,1,Answer()
exten => 507,n,Page(console/dsp)
exten => 507,n,Hangup()

; Page the Multicast channel - works fine
exten => 508,1,Answer()
exten => 508,n,Page(MulticastRTP/linksys/
exten => 508,n,Hangup()

; Page console and multicast channels - garbled sound on multicast
exten => 509,1,Answer()
exten => 509,n,Page(console/dsp&MulticastRTP/linksys/
exten => 509,n,Hangup()

  -- Executing [511@phones:1] Answer("SIP/323-0000001c", "") in new stack
  -- Executing [511@phones:2] Page("SIP/323-0000001c", "MulticastRTP/linksys/") in new stack
[Jan 31 23:57:23] NOTICE[27158]: app_page.c:197 page_exec: ConfBridge,756221750d,m1qw(5)
[Jan 31 23:57:23] NOTICE[27158]: chan_multicast_rtp.c:119 multicast_rtp_request: Format = g722  
 -- Called linksys/
<< Call placed to 'dsp' on console >>
<< Auto-answered >>
   -- Called dsp
   -- MulticastRTP/0xd4c0df0 answered
   -- ALSA/default answered
   -- <SIP/323-0000001c> Playing 'beep.slin' (language 'en')
<< Hangup on console >>
 == Spawn extension (phones, 511, 2) exited non-zero on 'SIP/323-0000001c'

; Page Multicast and a SIP phone - multicast works fine until other SIP phone (202) picks up, then
  the same garbled effect starts
exten => 508,1,Answer()
exten => 508,n,Page(MulticastRTP/linksys/
exten => 508,n,Hangup()


My thought is that the codec conversion between slin and g722/ulaw/alaw/gsm is failing inside either the MulticastRTP driver or the bridging application.  This may include the issue of sampling rate conversion (8KHz to 16KHz) based upon the audio generated.

By: Terry Wilson (twilson) 2012-03-20 20:07:39.822-0500

I have been unable to reproduce either problem mentioned below. As this is a very old issue, I assume that it has since been fixed. I will leave the issue open until next week in case anyone wants to comment.

By: Terry Wilson (twilson) 2012-03-26 15:38:41.683-0500

I was able to make multicast rtp calls with app_page with no problem. I assume that this has been fixed.

By: Remi Quezada (remiq) 2012-03-28 09:40:49.989-0500

I still get choppy audio with latest 1.8 SVN branch r360625.

By: Mark Murawski (kobaz) 2012-03-28 10:41:01.424-0500

Reporter still has a problem

By: Mark Murawski (kobaz) 2012-03-28 10:43:55.465-0500

Reporter, please provide more reproduction information.

By: Mark Murawski (kobaz) 2012-03-28 10:52:42.383-0500

Closing ticket for a new, related ticket.

By: Shamus Rask (juggler00) 2013-03-28 12:46:48.902-0500

I am experiencing the exact same problem as the original Bug report. I've got an Aastra 9143i running the latest f/w, Asterisk

When I try to Page a multicast address, I get no audio. When I Dial the same multicast address, the audio works.

Is there a way I should "re-open" this Bug?