[Asterisk-Users] MeetMe Problems
Waldo Rubinstein
waldo at trianet.net
Wed Jun 22 12:06:37 MST 2005
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 <markster at digium.com>
========================================================================
=
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/bacardi at bacardi/16384", "SIP/3211") in
new stack
-- Called 3211
-- SIP/3211-1bd8 is ringing
-- SIP/3211-1bd8 answered IAX2/bacardi at bacardi/16384
-- Started music on hold, class 'default', on IAX2/
bacardi at bacardi/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/bacardi at bacardi/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/
bacardi at bacardi/16384'
-- Hungup 'IAX2/bacardi at bacardi/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 <markster at digium.com>
> ======================================================================
> ===
> 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/
> server0 at 100000/16390
> -- 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/server0 at 100000/16390
> -- Hungup 'Zap/pseudo-1262753463'
> == Spawn extension (meetme, 0215, 1) exited non-zero on 'IAX2/
> server0 at 100000/16390'
> -- Hungup 'Zap/1-1'
> -- Hungup 'IAX2/server0 at 100000/16390'
>
> 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 <markster at digium.com>
> ======================================================================
> ===
> 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("IAX2/6500 at gateway0/16395", "211|1") in new
> stack
> -- Goto (client,211,1)
> -- Executing Macro("IAX2/6500 at gateway0/16395", "stdexten|211|
> SIP/3211") in new stack
> -- Executing Dial("IAX2/6500 at gateway0/16395", "SIP/3211|20|t")
> in new stack
> -- Called 3211
> -- SIP/3211-3c74 is ringing
> -- SIP/3211-3c74 answered IAX2/6500 at gateway0/16395
> -- Started music on hold, class 'default', on
> IAX2/6500 at gateway0/16395
> -- 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 IAX2/6500 at gateway0/16395
> 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
> 'IAX2/6500 at gateway0/16395'
> -- Executing Hangup("IAX2/6500 at gateway0/16395", "") in new stack
> == Spawn extension (client_INT, h, 1) exited non-zero on
> 'IAX2/6500 at gateway0/16395'
> == 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 'IAX2/6500 at gateway0/16395'
> -- Hungup 'IAX2/gateway0/16384'
> -- Accepting unauthenticated call from 10.0.10.9, requested
> format = 4, actual format = 4
> -- Executing Goto("IAX2/6500 at gateway0/16386", "211|1") in new
> stack
> -- Goto (client,211,1)
> -- Executing Macro("IAX2/6500 at gateway0/16386", "stdexten|211|
> SIP/3211") in new stack
> -- Executing Dial("IAX2/6500 at gateway0/16386", "SIP/3211|20|t")
> in new stack
> -- Called 3211
> -- SIP/3211-54f0 is ringing
> -- Format for call is ulaw
> -- SIP/3211-54f0 answered IAX2/6500 at gateway0/16386
> -- Hungup 'IAX2/gateway0/16388'
> -- Started music on hold, class 'default', on
> IAX2/6500 at gateway0/16386
> -- 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 IAX2/6500 at gateway0/16386
> -- Attempting native bridge of IAX2/6500 at gateway0/16386 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
> 'IAX2/6500 at gateway0/16386'
> -- Executing Hangup("IAX2/6500 at gateway0/16386", "") in new stack
> == Spawn extension (client_INT, h, 1) exited non-zero on
> 'IAX2/6500 at gateway0/16386'
> -- Hungup 'IAX2/6500 at gateway0/16386'
>
> 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 <waldo at trianet.net> 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
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users at lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
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>>>
>>>
>>>
>>
>>
>> --
>> "Su nombre es GNU/Linux, no solamente Linux, mas info en http://
>> www.gnu.org"
>> _______________________________________________
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>>
>
>
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