Re: [Asterisk-Users] MeetMe Problems

Thu, 23 Jun 2005 02:21:48 -0700

Doing further tests, I discovered that I can successfully do MeetMe on both server B and server C, AS LONG AS all parties are SIP extensions registered on the same server (e.g. server B or server C). However, when I try to bring a call from server A into a MeetMe in server B or server C, that's when the problem shows up. Hope this helps anyone who can help me. 

Thanks,
Waldo

On Jun 22, 2005, at 3:06 PM, Waldo Rubinstein wrote:
I decided to test a similar scenario against another machine (server C). This machine behaves in a similar way as server B. It is also running on Gentoo. When I try to transfer a call into a conference room, it fails. Below is the CLI output of an inbound call coming from server A into server C, ringing extension SIP/ 3211. Once answered, I try transferring to MeetMe room 0211 which fails. 

bacardi init.d # aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
====================================================================== === 
Connected to Asterisk 1.0.7 currently running on bacardi (pid = 10925)
Verbosity was 0 and is now 10
-- Accepting AUTHENTICATED call from 10.0.10.9, requested format = 4, actual format = 4 -- Executing Dial("IAX2/[EMAIL PROTECTED]/16384", "SIP/3211") in new stack 
    -- Called 3211
    -- SIP/3211-1bd8 is ringing
    -- SIP/3211-1bd8 answered IAX2/[EMAIL PROTECTED]/16384
-- Started music on hold, class 'default', on IAX2/ [EMAIL PROTECTED]/16384 
    -- Executing MeetMe("SIP/3211-e3c6", "0211|qM") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '0211'
    -- Started music on hold, class 'default', on SIP/3211-e3c6
    -- Stopped music on hold on SIP/3211-e3c6
    -- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16384
Jun 22 14:59:40 WARNING[10952]: app_meetme.c:667 conf_run: Error getting conference 
    -- Hungup 'Zap/pseudo-1721629866'
== Spawn extension (default, 0211, 1) exited non-zero on 'IAX2/ [EMAIL PROTECTED]/16384' 
    -- Hungup 'IAX2/[EMAIL PROTECTED]/16384'
-- Attempting native bridge of SIP/3211-e3c6<ZOMBIE> and SIP/ 3211-1bd8 Jun 22 14:59:40 WARNING[10951]: rtp.c:1365 ast_rtp_bridge: Can't find native functions for channel 'SIP/3211-e3c6<ZOMBIE>' Jun 22 14:59:40 WARNING[10951]: channel.c:2634 ast_channel_bridge: Private bridge between SIP/3211-e3c6<ZOMBIE> and SIP/3211-1bd8 failed == Spawn extension (default, 3211, 1) exited non-zero on 'SIP/ 3211-e3c6<ZOMBIE>'
I don't know if it has anything to do with the <ZOMBIE> channel. lsmod shows that both zaptel and ztdummy are loaded. Any ideas? 

Thanks,
Waldo

On Jun 22, 2005, at 10:41 AM, Waldo Rubinstein wrote:
Absolutely. Here is the CLI output. I made two attempts. First, I dialed inbound into an extension and then tried using meetme room 0201 from Server B, which didn't work. Then I dialed inbound into the same extension and then tried using meetme room 0215 which resides in Server A. Note that all inbound calls come into Server A, for it has the Digium card. 

SERVER A
=========

gateway0:~# aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25, Copyright (C) 1999-2004 Digium. 
Written by Mark Spencer <[EMAIL PROTECTED]>
===================================================================== ==== Connected to Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 currently running on gateway0 (pid = 2653) 
Verbosity is at least 10
    -- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", "Inbound Call - 3211 - ") in new stack 
    -- Executing Dial("Zap/1-1", "IAX2/corona/3211||r") in new stack
    -- Called corona/3211
    -- Call accepted by 10.0.10.13 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/corona/16386 is ringing
    -- IAX2/corona/16386 answered Zap/1-1
    -- Hungup 'IAX2/corona/16386'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
    -- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", "Inbound Call - 3211 - ") in new stack 
    -- Executing Dial("Zap/1-1", "IAX2/corona/3211||r") in new stack
    -- Called corona/3211
    -- Call accepted by 10.0.10.13 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/corona/16388 is ringing
    -- IAX2/corona/16388 answered Zap/1-1
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '0215'
-- Started music on hold, class 'default', on IAX2/ [EMAIL PROTECTED]/16390 
    -- Hungup 'Zap/31-1'
    -- Hungup 'IAX2/corona/16388'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
    -- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16390
    -- Hungup 'Zap/pseudo-1262753463'
== Spawn extension (meetme, 0215, 1) exited non-zero on 'IAX2/ [EMAIL PROTECTED]/16390' 
    -- Hungup 'Zap/1-1'
    -- Hungup 'IAX2/[EMAIL PROTECTED]/16390'

Here are the relevant sections in the .conf files:

meetme.conf:
[rooms]
conf => 0215

extensions.conf:
[meetme]
exten => 0215,1,MeetMe(0215|qM)
exten => 0215,2,Hangup


SERVER B
=========

corona root # aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
===================================================================== ==== 
Connected to Asterisk 1.0.7 currently running on corona (pid = 5105)
Verbosity is at least 10
    -- Remote UNIX connection
    -- Call accepted by 10.0.10.9 (format ulaw)
    -- Format for call is ulaw
-- Accepting unauthenticated call from 10.0.10.9, requested format = 4, actual format = 4 -- Executing Goto("IAX2/[EMAIL PROTECTED]/16395", "211|1") in new stack 
    -- Goto (client,211,1)
-- Executing Macro("IAX2/[EMAIL PROTECTED]/16395", "stdexten|211| SIP/3211") in new stack -- Executing Dial("IAX2/[EMAIL PROTECTED]/16395", "SIP/3211|20|t") in new stack 
    -- Called 3211
    -- SIP/3211-3c74 is ringing
    -- SIP/3211-3c74 answered IAX2/[EMAIL PROTECTED]/16395
-- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/16395 
    -- Executing MeetMe("SIP/3211-4ed5", "0201|qM") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '0201'
    -- Started music on hold, class 'default', on SIP/3211-4ed5
    -- Stopped music on hold on SIP/3211-4ed5
    -- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16395
Jun 21 18:57:41 WARNING[8254]: app_meetme.c:667 conf_run: Error getting conference 
    -- Hungup 'Zap/pseudo-510190782'
== Spawn extension (client_INT, 0201, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/16395' 
    -- Executing Hangup("IAX2/[EMAIL PROTECTED]/16395", "") in new stack
== Spawn extension (client_INT, h, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/16395' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/3211-4ed5<ZOMBIE>' in macro 'stdexten' == Spawn extension (client, 211, 1) exited non-zero on 'SIP/ 3211-4ed5<ZOMBIE>' 
    -- Executing Hangup("SIP/3211-4ed5<ZOMBIE>", "") in new stack
== Spawn extension (client, h, 1) exited non-zero on 'SIP/ 3211-4ed5<ZOMBIE>' 
    -- Hungup 'IAX2/[EMAIL PROTECTED]/16395'
    -- Hungup 'IAX2/gateway0/16384'
-- Accepting unauthenticated call from 10.0.10.9, requested format = 4, actual format = 4 -- Executing Goto("IAX2/[EMAIL PROTECTED]/16386", "211|1") in new stack 
    -- Goto (client,211,1)
-- Executing Macro("IAX2/[EMAIL PROTECTED]/16386", "stdexten|211| SIP/3211") in new stack -- Executing Dial("IAX2/[EMAIL PROTECTED]/16386", "SIP/3211|20|t") in new stack 
    -- Called 3211
    -- SIP/3211-54f0 is ringing
    -- Format for call is ulaw
    -- SIP/3211-54f0 answered IAX2/[EMAIL PROTECTED]/16386
    -- Hungup 'IAX2/gateway0/16388'
-- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/16386 -- Executing Dial("SIP/3211-937c", "IAX2/gateway0/0215") in new stack 
    -- Called gateway0/0215
    -- Call accepted by 10.0.10.9 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/gateway0/16390 answered SIP/3211-937c
    -- Hungup 'IAX2/gateway0/16391'
    -- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16386
-- Attempting native bridge of IAX2/[EMAIL PROTECTED]/16386 and IAX2/gateway0/16390 -- Attempting native bridge of SIP/3211-937c<ZOMBIE> and SIP/ 3211-54f0 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/3211-937c<ZOMBIE>' in macro 'stdexten' == Spawn extension (client, 211, 1) exited non-zero on 'SIP/ 3211-937c<ZOMBIE>' 
    -- Executing Hangup("SIP/3211-937c<ZOMBIE>", "") in new stack
== Spawn extension (client, h, 1) exited non-zero on 'SIP/ 3211-937c<ZOMBIE>' 
    -- Hungup 'IAX2/gateway0/16390'
== Spawn extension (client_INT, 0215, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/16386' 
    -- Executing Hangup("IAX2/[EMAIL PROTECTED]/16386", "") in new stack
== Spawn extension (client_INT, h, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/16386' 
    -- Hungup 'IAX2/[EMAIL PROTECTED]/16386'

Here are the relevant sections in the .conf files:

meetme.conf:
[rooms]
conf => 0201

extensions.conf:
[meetme]
exten => 0201,1,MeetMe(0201|qM)
exten => 0201,2,Hangup
exten => 0215,1,Dial(${GATEWAY}/${EXTEN})

where ${GATEWAY} is the IAX2 url of Server A

Hope this helps.

Thanks,
Waldo

On Jun 21, 2005, at 4:57 PM, Moises Silva wrote:




it would be very helpfull (IMHO) if you post the output of the
Asterisk console with a high verbosity level. Also, show us how the
important code in your extensions.conf

best regards

On 6/21/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
I have two asterisk machines. One of them has a Digium board (server 
A) and the other is simply using ztdummy (server B). Server A is
running on Debian and Server B is running Gentoo. Server A is running 
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running
Asterisk 1.0.7.

The problem I have is that when I try to transfer a call into a
meetme room in server B, it simply hangs up the call. To be specific, when I press transfer (XFER on the Uniden UIP200) and then the meetme 
room number, the meetme room answers (I hear MOH), but when I hang
up, it drops all calls and not just transfers the call to the meetme 
room.
Now, if I configure the meetme rooms indentically in server A, I can 
transfer the calls from server B to server A's meetme room and
everything works just fine.
I would like for the meetme rooms to work in server B and not having 
to depend on server A for it.

Can anyone shed some light into why this is happening and, more
importantly, how to fix it?

Thanks,
Waldo
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