[Asterisk-Users] MeetMe Problems
Waldo Rubinstein
waldo at trianet.net
Thu Jun 23 02:19:32 MST 2005
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 <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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>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>> --
>>> "Su nombre es GNU/Linux, no solamente Linux, mas info en http://
>>> www.gnu.org"
>>> _______________________________________________
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>>>
>>>
>>
>>
>>
>
>
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