From msc at freeswitch.org  Tue Dec  1 16:56:25 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 1 Dec 2009 16:56:25 -0800
Subject: [Freeswitch-dev] FreeSWITCH Survey: What Environment Do You
	Normally Use?
Message-ID: <87f2f3b90912011656m5d08d466n641e341ec9889373@mail.gmail.com>

Hi folks,

I'm doing a little survey to get an idea of what everyone prefers to use for
their operating environment, like 32 vs. 64 bit, Linux vs. Windows, etc.
Please log in to the main page and check out this node:

http://www.freeswitch.org/node/206

Select the environment that you use the most or prefer to use. In a week or
so I will send out the final tally. I'm sure the only question is who will
come in second place after 64-bit CentOS/Red Hat. :)

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091201/1d71d6b5/attachment.html 

From shiyanov at gmail.com  Thu Dec  3 10:16:20 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Thu, 3 Dec 2009 21:16:20 +0300
Subject: [Freeswitch-dev] uuid_bridge kills both channels if they are
	executing java app
In-Reply-To: <191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com>
References: <d211f54a0911191146m6cc8fec7tebb26fffb1f639a2@mail.gmail.com>
	<191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com>
	<d211f54a0911210158x74005ea6mccbdf58948e7deba@mail.gmail.com>
	<d211f54a0912021108v52e8eebeu696ea40df358dd9d@mail.gmail.com>
	<191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com>
Message-ID: <d211f54a0912031016u562c4e94n638f40620bce166@mail.gmail.com>

Hi there!

This message is a forward from user-mail-list.
I'm trying to fix such a problem:
FreSwithch compiled from SVN-trunk, date = 11/02/2009.

What is need: connect two users, initially one is on the home-grown
java-based IVR and other party is off hook.

What is done/got:
User1 is on the java application, it represents simple IVR system, and the
most used FS API operation is "streamFile".
User2 is off hook.
next:
(mod_socket) create_uuid
<get uuid_x>
bgapi originate
{origination_caller_id_name=User1}[origination_uuid=uuid_x]User1 &park()
<User1 answers incomming call>
<get event channel_User1 answered>
<get event channel_User1 parked>
uuid_bridge uuid_User1 uuid_User2
<get event channel_User1 hangup, cause=NORMAL_CLEARING>
<get event channel_User2 hangup, cause=*DESTINATION_OUT_OF_ORDER*>
<both channel_User1 and channel_User2 are down>


FS log is here: http://pastebin.freeswitch.org/11380


Thank you much for any help,
Artem





---------- Forwarded message ----------
From: Anthony Minessale <anthony.minessale at gmail.com>
Date: Wed, Dec 2, 2009 at 10:24 PM
Subject: Re: [Freeswitch-users] uuid_bridge kills both channels if they are
executing java app
To: freeswitch-users at lists.freeswitch.org


you should be working on SVN trunk if you are doing development, we are so
far forward from 1.0.4 we can't do debugging very easily.

I don't know all of the details of what you are trying to do but you are
hitting some race conditions because of the async nature of the socket
connection and the way you are using it.




On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> I'm back again with the same issue.
> Now it is became worse: it reproduces occasionally.
> [FS version is 1.04, test_load = 2 active calls]
>
> I've got 2 logs: successful and not.
> Here is a bad_case:
>
> 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute
> java(/usr/local/freeswitch/scripts/fs2agi.jar
> org.starpound.fs2agi.Translator
> ${agi_url})
> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
> INFO: ***************************************************
> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for
> session
> 2898ad41-4ec1-4628-89fd-651a93a7221d
> 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI application
> agi://localhost:4573/hello.agi?callId=929
> 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream
> handle!
>
> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
> 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17]
> 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready
> sofia/internal/2001!
> 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel
> [sofia/internal/2001] has
> been answered
> Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session
> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
> java.lang.Exception: Internal FreeSwitch failure while streamming file, see
> FreeSwitch logs for details
>         at
>
> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>         at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>         at org.starpound.fs2agi.Translator.run(Translator.java:56)
>         at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>         at
>
> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>         at
>
> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>         at java.lang.reflect.Method.invoke(Method.java:597)
>         at org.freeswitch.Launcher.launch(Launcher.java:80)
> 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup
> sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application
> agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for
> details.
> 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup
> sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE]
> [DESTINATION_OUT_OF_ORDER]
> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17
> (sofia/external/6786081291 at 66.19.38.143) Ended
> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
> Channel
> sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY]
> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18
> (sofia/internal/2001) Ended
> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
> Channel
> sofia/internal/2001 [CS_DESTROY]
>
>
>
> Message
> "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session
> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
> ..."
> is sent from my app upon the onHangup().`
>
> And here is a good_case:
>
> 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute
> java(/usr/local/freeswitch/scripts/fs2agi.jar
> org.starpound.fs2agi.Translator
> ${agi_url})
> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run
> INFO: ***************************************************
> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run
> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for
> session
> 7c37369b-ffb2-4436-9288-a640047d0e5e
> 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI application
> agi://localhost:4573/hello.agi?callId=932
> 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream
> handle!
>
> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
> 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488]
> 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready
> sofia/internal/2001!
> 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel
> [sofia/internal/2001] has
> been answered
> Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed
> INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for session
> 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is:
> java.lang.Exception: Internal FreeSwitch failure while streamming file, see
> FreeSwitch logs for details
>         at
>
> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>         at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>         at org.starpound.fs2agi.Translator.run(Translator.java:56)
>         at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>         at
>
> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>         at
>
> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>         at java.lang.reflect.Method.invoke(Method.java:597)
>         at org.freeswitch.Launcher.launch(Launcher.java:80)
> 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application
> agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for
> details.
> 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port from
> 172.26.10.39:26402 to 91.190.120.190:26402
>
>
>
> Suggestions?
>
>
>
>
>
>
>
>
>
>
>
> On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>
>> Anthony,
>>
>> >>As soon as you call uuid_bridge you are transferring both legs of the
>> call to bridge to each other.
>> >>This means your java app must exit so the channels can connect to each
>> other.
>>
>> I didn't know that. Now my java app is exiting upon the onHangup() call so
>> everything has become "ok". Thank you much.
>> I'll add note to the wiki about this issue.
>>
>> Artem
>>
>>
>>
>>
>> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Your "annoying behaviour" is the exact behavior you should be getting
>>> considering what you told FS to do.
>>>
>>> As soon as you call uuid_bridge you are transferring both legs of the
>>> call to bridge to each other.
>>> This means your java app must exit so the channels can connect to each
>>> other.
>>>
>>> remember that you hangup hook can be called when the channel is
>>> transferred not only when it hangs up.
>>> you have to test which is happening based on the input to your callback.
>>>
>>>
>>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>>
>>>> Hi there!
>>>>
>>>> I've got annoying FS behavior:
>>>> There are 2 channels executing the same Java application (application
>>>> itself is an IVR). If I try to bridge them with uuid_bridged then both
>>>> channels are killed. Here is a log from FS console:
>>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165
>>>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE ->
>>>> CS_HIBERNATE
>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>> called
>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done
>>>> playing file
>>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done
>>>> playing file
>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal
>>>> sofia/internal/1005 at 192.168.147.130 [BREAK]
>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167
>>>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change
>>>> CS_EXECUTE -> CS_HIBERNATE
>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>> called
>>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output:
>>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>>
>>>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG]
>>>> switch_core_session.c:933 Send signal
>>>> sofia/internal/1001 at master.agent.starpoundtec
>>>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal
>>>> sofia/internal/1005 at 192.168.147.130 [BREAK]
>>>>
>>>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream
>>>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>>>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal
>>>> sofia/internal/1001 at master.agent.starpoundtech.net [BREAK]
>>>>
>>>> (FS version is 1.0.4)
>>>>
>>>> Any thoughts?
>>>>
>>>>
>>>> Artem
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091203/c3e1a19a/attachment-0001.html 

From anthony.minessale at gmail.com  Thu Dec  3 10:41:50 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Dec 2009 12:41:50 -0600
Subject: [Freeswitch-dev] uuid_bridge kills both channels if they are
	executing java app
In-Reply-To: <d211f54a0912031016u562c4e94n638f40620bce166@mail.gmail.com>
References: <d211f54a0911191146m6cc8fec7tebb26fffb1f639a2@mail.gmail.com>
	<191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com>
	<d211f54a0911210158x74005ea6mccbdf58948e7deba@mail.gmail.com>
	<d211f54a0912021108v52e8eebeu696ea40df358dd9d@mail.gmail.com>
	<191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com>
	<d211f54a0912031016u562c4e94n638f40620bce166@mail.gmail.com>
Message-ID: <191c3a030912031041i1a941f6ev970c687d124b34d6@mail.gmail.com>

The case of the channel hangup with DESTINATION_OUT_OF_ORDER is not depicted
in your log.
can you capture the log of the entire procedure top to bottom and file a
jira ticket on http://jira.freeswitch.org and attach the trace as a .txt
file.



On Thu, Dec 3, 2009 at 12:16 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Hi there!
>
> This message is a forward from user-mail-list.
> I'm trying to fix such a problem:
> FreSwithch compiled from SVN-trunk, date = 11/02/2009.
>
> What is need: connect two users, initially one is on the home-grown
> java-based IVR and other party is off hook.
>
> What is done/got:
> User1 is on the java application, it represents simple IVR system, and the
> most used FS API operation is "streamFile".
> User2 is off hook.
> next:
> (mod_socket) create_uuid
> <get uuid_x>
> bgapi originate
> {origination_caller_id_name=User1}[origination_uuid=uuid_x]User1 &park()
> <User1 answers incomming call>
> <get event channel_User1 answered>
> <get event channel_User1 parked>
> uuid_bridge uuid_User1 uuid_User2
> <get event channel_User1 hangup, cause=NORMAL_CLEARING>
> <get event channel_User2 hangup, cause=*DESTINATION_OUT_OF_ORDER*>
> <both channel_User1 and channel_User2 are down>
>
>
> FS log is here: http://pastebin.freeswitch.org/11380
>
>
> Thank you much for any help,
> Artem
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Date: Wed, Dec 2, 2009 at 10:24 PM
> Subject: Re: [Freeswitch-users] uuid_bridge kills both channels if they are
> executing java app
> To: freeswitch-users at lists.freeswitch.org
>
>
> you should be working on SVN trunk if you are doing development, we are so
> far forward from 1.0.4 we can't do debugging very easily.
>
> I don't know all of the details of what you are trying to do but you are
> hitting some race conditions because of the async nature of the socket
> connection and the way you are using it.
>
>
>
>
> On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:
>
>> I'm back again with the same issue.
>> Now it is became worse: it reproduces occasionally.
>> [FS version is 1.04, test_load = 2 active calls]
>>
>> I've got 2 logs: successful and not.
>> Here is a bad_case:
>>
>> 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute
>> java(/usr/local/freeswitch/scripts/fs2agi.jar
>> org.starpound.fs2agi.Translator
>> ${agi_url})
>> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
>> INFO: ***************************************************
>> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
>> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for
>> session
>> 2898ad41-4ec1-4628-89fd-651a93a7221d
>> 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI
>> application
>> agi://localhost:4573/hello.agi?callId=929
>> 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream
>> handle!
>>
>> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>> 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel
>> sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17]
>> 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready
>> sofia/internal/2001!
>> 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel
>> [sofia/internal/2001] has
>> been answered
>> Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
>> session
>> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
>> java.lang.Exception: Internal FreeSwitch failure while streamming file,
>> see
>> FreeSwitch logs for details
>>         at
>>
>> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>>         at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>>         at org.starpound.fs2agi.Translator.run(Translator.java:56)
>>         at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>>         at
>>
>> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>>         at
>>
>> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>>         at java.lang.reflect.Method.invoke(Method.java:597)
>>         at org.freeswitch.Launcher.launch(Launcher.java:80)
>> 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup
>> sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application
>> agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for
>> details.
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup
>> sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE]
>> [DESTINATION_OUT_OF_ORDER]
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17
>> (sofia/external/6786081291 at 66.19.38.143) Ended
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
>> Channel
>> sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY]
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18
>> (sofia/internal/2001) Ended
>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
>> Channel
>> sofia/internal/2001 [CS_DESTROY]
>>
>>
>>
>> Message
>> "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
>> session
>> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
>> ..."
>> is sent from my app upon the onHangup().`
>>
>> And here is a good_case:
>>
>> 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute
>> java(/usr/local/freeswitch/scripts/fs2agi.jar
>> org.starpound.fs2agi.Translator
>> ${agi_url})
>> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run
>> INFO: ***************************************************
>> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run
>> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for
>> session
>> 7c37369b-ffb2-4436-9288-a640047d0e5e
>> 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI
>> application
>> agi://localhost:4573/hello.agi?callId=932
>> 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream
>> handle!
>>
>> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>> 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel
>> sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488]
>> 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready
>> sofia/internal/2001!
>> 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel
>> [sofia/internal/2001] has
>> been answered
>> Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>> INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for
>> session
>> 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is:
>> java.lang.Exception: Internal FreeSwitch failure while streamming file,
>> see
>> FreeSwitch logs for details
>>         at
>>
>> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>>         at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>>         at org.starpound.fs2agi.Translator.run(Translator.java:56)
>>         at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>>         at
>>
>> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>>         at
>>
>> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>>         at java.lang.reflect.Method.invoke(Method.java:597)
>>         at org.freeswitch.Launcher.launch(Launcher.java:80)
>> 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application
>> agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for
>> details.
>> 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port
>> from
>> 172.26.10.39:26402 to 91.190.120.190:26402
>>
>>
>>
>> Suggestions?
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>
>>> Anthony,
>>>
>>> >>As soon as you call uuid_bridge you are transferring both legs of the
>>> call to bridge to each other.
>>> >>This means your java app must exit so the channels can connect to each
>>> other.
>>>
>>> I didn't know that. Now my java app is exiting upon the onHangup() call
>>> so everything has become "ok". Thank you much.
>>> I'll add note to the wiki about this issue.
>>>
>>> Artem
>>>
>>>
>>>
>>>
>>> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Your "annoying behaviour" is the exact behavior you should be getting
>>>> considering what you told FS to do.
>>>>
>>>> As soon as you call uuid_bridge you are transferring both legs of the
>>>> call to bridge to each other.
>>>> This means your java app must exit so the channels can connect to each
>>>> other.
>>>>
>>>> remember that you hangup hook can be called when the channel is
>>>> transferred not only when it hangs up.
>>>> you have to test which is happening based on the input to your callback.
>>>>
>>>>
>>>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>>>
>>>>> Hi there!
>>>>>
>>>>> I've got annoying FS behavior:
>>>>> There are 2 channels executing the same Java application (application
>>>>> itself is an IVR). If I try to bridge them with uuid_bridged then both
>>>>> channels are killed. Here is a log from FS console:
>>>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165
>>>>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE ->
>>>>> CS_HIBERNATE
>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>>> called
>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done
>>>>> playing file
>>>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done
>>>>> playing file
>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send
>>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK]
>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167
>>>>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change
>>>>> CS_EXECUTE -> CS_HIBERNATE
>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>>> called
>>>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output:
>>>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>>>
>>>>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG]
>>>>> switch_core_session.c:933 Send signal
>>>>> sofia/internal/1001 at master.agent.starpoundtec
>>>>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send
>>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK]
>>>>>
>>>>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream
>>>>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>>>>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send
>>>>> signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK]
>>>>>
>>>>> (FS version is 1.0.4)
>>>>>
>>>>> Any thoughts?
>>>>>
>>>>>
>>>>> Artem
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091203/41018bd3/attachment-0001.html 

From mbrancaleoni at voismart.it  Thu Dec  3 10:47:56 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Thu, 3 Dec 2009 19:47:56 +0100 (CET)
Subject: [Freeswitch-dev] mod event multicast drops some events on outbound
	call legs ?
In-Reply-To: <59221839.4591.1259861292832.JavaMail.root@mx.voismart.com>
Message-ID: <1445390143.4611.1259866076238.JavaMail.root@mx.voismart.com>

Hi,

I was adding "multicast presence" to FS using mod_event_multicast.
Useful to have BLF over some machines :)

First implementation works almost ok, but I'm facing a strange issue.

Seems that mod_event_multicast drops some events, which can seen correctly
over mod_event_socket.

For example I can see only PRESENCE_IN events for the caller, but not
for the called, when the state is early.

For example: if caller is 1001 and called is 1000, over event_socket I see
PRESENCE_IN updates for both during all the transaction, but over mcast 
I see only event initial events for 1001 (which is confirmed), and
I get events for 1000 only when the call is cancelled (1000 terminated)
or answere (1000 is confirmed also).

Over event socket is all ok, I cann see everything.

Debugging presence I can get over console all the PRESENCE_IN messages.

Seems for some reason that event multicast is not able to
get events for the outbound leg when is in early state?

Any hints?

I'll be glad to release the patch to sofia to add multicast BLF when all works :)

regards,
matteo.


From anthony.minessale at gmail.com  Thu Dec  3 11:03:29 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 3 Dec 2009 13:03:29 -0600
Subject: [Freeswitch-dev] mod event multicast drops some events on
	outbound call legs ?
In-Reply-To: <1445390143.4611.1259866076238.JavaMail.root@mx.voismart.com>
References: <59221839.4591.1259861292832.JavaMail.root@mx.voismart.com>
	<1445390143.4611.1259866076238.JavaMail.root@mx.voismart.com>
Message-ID: <191c3a030912031103i2e630c96g4187c4a1957c4272@mail.gmail.com>

where are you seeing it drop?
if you log them in the module is it sending them but then they are not being
recv?
can you explain a bit more?


On Thu, Dec 3, 2009 at 12:47 PM, Matteo <mbrancaleoni at voismart.it> wrote:

> Hi,
>
> I was adding "multicast presence" to FS using mod_event_multicast.
> Useful to have BLF over some machines :)
>
> First implementation works almost ok, but I'm facing a strange issue.
>
> Seems that mod_event_multicast drops some events, which can seen correctly
> over mod_event_socket.
>
> For example I can see only PRESENCE_IN events for the caller, but not
> for the called, when the state is early.
>
> For example: if caller is 1001 and called is 1000, over event_socket I see
> PRESENCE_IN updates for both during all the transaction, but over mcast
> I see only event initial events for 1001 (which is confirmed), and
> I get events for 1000 only when the call is cancelled (1000 terminated)
> or answere (1000 is confirmed also).
>
> Over event socket is all ok, I cann see everything.
>
> Debugging presence I can get over console all the PRESENCE_IN messages.
>
> Seems for some reason that event multicast is not able to
> get events for the outbound leg when is in early state?
>
> Any hints?
>
> I'll be glad to release the patch to sofia to add multicast BLF when all
> works :)
>
> regards,
> matteo.
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091203/d865e382/attachment.html 

From mbrancaleoni at voismart.it  Thu Dec  3 11:26:02 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Thu, 3 Dec 2009 20:26:02 +0100 (CET)
Subject: [Freeswitch-dev] mod event multicast drops some events
	on	outbound call legs ?
In-Reply-To: <1643190246.4619.1259868183555.JavaMail.root@mx.voismart.com>
Message-ID: <1401174787.4621.1259868362534.JavaMail.root@mx.voismart.com>

Hi,

----- "Anthony Minessale" <anthony.minessale at gmail.com> ha scritto:
> where are you seeing it drop?

on both fs istances.

> if you log them in the module is it sending them but then they are not
> being recv?

No, they're not even enterin the event_handler callback on mod_event_multicast

> can you explain a bit more?

sure.
The scenario is simple: two fs on 2 different machines, on same lan.

Fs compiled from trunk, standard config with mod_event_multicast
compiled and enabled.

Sip registrations are shared ok via multicast, so module is working.
Sip domain is the same on both fs istances, to be able to register
to any server and call any client. This is also working ok.

Then I started looking into events sent & received via multicast.

I uncommented the debug print on the mcast event receiver, and added
a simple debug line on the event handler callback to print out the event type
ready to be sent.

Every event that's sent, is received ok. No problem here.

But I noticed that the events of type PRESENCE_IN, regarding the outbound
leg of a call are not sent to the multicast event handler to be propagated 
(no debug print at all) ONLY when the outbound channel is in early state.

Previous events like EXECUTE, EXECUTE_COMPLETE, CODEC, CALL_UPDATE, etc, which
happens before dialing the outboung leg are sent correctly.

The "missing" events are present on the core because:
* enabling presence debug, I can see them on cli
* I can get them on the event socket

But I don't see them entering the mcast handler to be transmitted, 
so they cannot be received :)

As soon as the outbound leg is answered, the corresponding event
is sent ok via multicast. Same when the call is closed.
Only the early state does not hit the handler.

regards,
Matteo.


the call


From andrew at hijacked.us  Thu Dec  3 11:40:12 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 3 Dec 2009 14:40:12 -0500
Subject: [Freeswitch-dev] mod event multicast drops some events
	on	outbound call legs ?
In-Reply-To: <1401174787.4621.1259868362534.JavaMail.root@mx.voismart.com>
References: <1643190246.4619.1259868183555.JavaMail.root@mx.voismart.com>
	<1401174787.4621.1259868362534.JavaMail.root@mx.voismart.com>
Message-ID: <20091203194012.GA22725@hijacked.us>

And what do your event bindings look like? just 'all'?

Andrew


From mbrancaleoni at voismart.it  Thu Dec  3 11:47:45 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Thu, 3 Dec 2009 20:47:45 +0100 (CET)
Subject: [Freeswitch-dev] mod event multicast drops some
	events	on	outbound call legs ?
In-Reply-To: <20091203194012.GA22725@hijacked.us>
Message-ID: <181757844.4624.1259869665939.JavaMail.root@mx.voismart.com>

Hi,

----- "Andrew Thompson" <andrew at hijacked.us> ha scritto:
> And what do your event bindings look like? just 'all'?

Yes, default config.

regards,
matteo


From mbrancaleoni at voismart.it  Fri Dec  4 01:21:21 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Fri, 4 Dec 2009 10:21:21 +0100 (CET)
Subject: [Freeswitch-dev] mod event multicast drops some events on
 outbound call legs ?
In-Reply-To: <665901512.4705.1259916997041.JavaMail.root@mx.voismart.com>
Message-ID: <672860228.4720.1259918481724.JavaMail.root@mx.voismart.com>

Hi,

----- "Anthony Minessale" <anthony.minessale at gmail.com> ha scritto:

> where are you seeing it drop?
> if you log them in the module is it sending them but then they are not
> being recv?

I've done some more detailed tests.
Seems that the event is sent correctly over multicast (I can get it
with wireshark), but is not received on the listener in
SWITCH_MODULE_RUNTIME_FUNCTION(mod_event_multicast_runtime) function.

I've attached two logs:
mcast_dump.txt is the udp stream got with wireshark. All events
are present here, so they're on wire.

mcast_on_receiver.txt is what mod_event_multicast gets in input.

Seems that the issue happens frequently, but sometimes works.
Maybe some race condition in the receiving routine ?

I'll try to investigate further.

regards,
matteo
-------------- next part --------------
A non-text attachment was scrubbed...
Name: mcast_dump.txt.gz
Type: application/x-gzip
Size: 7357 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091204/c429d399/attachment-0002.gz 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: mcast_on_receiver.txt.gz
Type: application/x-gzip
Size: 8924 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091204/c429d399/attachment-0003.gz 

From mbrancaleoni at voismart.it  Fri Dec  4 02:19:27 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Fri, 4 Dec 2009 11:19:27 +0100 (CET)
Subject: [Freeswitch-dev] mod event multicast drops some events on
 outbound call legs ?
In-Reply-To: <37787900.4751.1259921827159.JavaMail.root@mx.voismart.com>
Message-ID: <2134690418.4753.1259921967638.JavaMail.root@mx.voismart.com>

Hi again,

I've found a fix, but maybe the error is hidded somewhere else.

If in the function SWITCH_MODULE_RUNTIME_FUNCTION(mod_event_multicast_runtime),
line 515 in mod_event_multicast.c I change switch_yield(100000);  to a lower
time, for example switch_yield(1000); it works ok.

so maybe a loo long sleep time makes the multicast receiver loose
packets if we have many events (I'm sending all the events for now,
maybe with filters this will not happen).

So, with switch_yield(1000); works ok.

But this's the right fix?

regards,
matteo


From shiyanov at gmail.com  Fri Dec  4 02:20:01 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Fri, 4 Dec 2009 13:20:01 +0300
Subject: [Freeswitch-dev] uuid_bridge kills both channels if they are
	executing java app
In-Reply-To: <191c3a030912031041i1a941f6ev970c687d124b34d6@mail.gmail.com>
References: <d211f54a0911191146m6cc8fec7tebb26fffb1f639a2@mail.gmail.com>
	<191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com>
	<d211f54a0911210158x74005ea6mccbdf58948e7deba@mail.gmail.com>
	<d211f54a0912021108v52e8eebeu696ea40df358dd9d@mail.gmail.com>
	<191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com>
	<d211f54a0912031016u562c4e94n638f40620bce166@mail.gmail.com>
	<191c3a030912031041i1a941f6ev970c687d124b34d6@mail.gmail.com>
Message-ID: <d211f54a0912040220u72c3d25fqe559fe93d1c759e3@mail.gmail.com>

Anthony,

in pastebined log you could search "DESTINATION_OUT_OF_ORDER" and you'll
find it.
Anyway, I'll fire a bug in Jira.

Artem



On Thu, Dec 3, 2009 at 9:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The case of the channel hangup with DESTINATION_OUT_OF_ORDER is not
> depicted in your log.
> can you capture the log of the entire procedure top to bottom and file a
> jira ticket on http://jira.freeswitch.org and attach the trace as a .txt
> file.
>
>
>
> On Thu, Dec 3, 2009 at 12:16 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>
>> Hi there!
>>
>> This message is a forward from user-mail-list.
>> I'm trying to fix such a problem:
>> FreSwithch compiled from SVN-trunk, date = 11/02/2009.
>>
>> What is need: connect two users, initially one is on the home-grown
>> java-based IVR and other party is off hook.
>>
>> What is done/got:
>> User1 is on the java application, it represents simple IVR system, and the
>> most used FS API operation is "streamFile".
>> User2 is off hook.
>> next:
>> (mod_socket) create_uuid
>> <get uuid_x>
>> bgapi originate
>> {origination_caller_id_name=User1}[origination_uuid=uuid_x]User1 &park()
>> <User1 answers incomming call>
>> <get event channel_User1 answered>
>> <get event channel_User1 parked>
>> uuid_bridge uuid_User1 uuid_User2
>> <get event channel_User1 hangup, cause=NORMAL_CLEARING>
>> <get event channel_User2 hangup, cause=*DESTINATION_OUT_OF_ORDER*>
>> <both channel_User1 and channel_User2 are down>
>>
>>
>> FS log is here: http://pastebin.freeswitch.org/11380
>>
>>
>> Thank you much for any help,
>> Artem
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Date: Wed, Dec 2, 2009 at 10:24 PM
>> Subject: Re: [Freeswitch-users] uuid_bridge kills both channels if they
>> are executing java app
>> To: freeswitch-users at lists.freeswitch.org
>>
>>
>> you should be working on SVN trunk if you are doing development, we are so
>> far forward from 1.0.4 we can't do debugging very easily.
>>
>> I don't know all of the details of what you are trying to do but you are
>> hitting some race conditions because of the async nature of the socket
>> connection and the way you are using it.
>>
>>
>>
>>
>> On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>
>>> I'm back again with the same issue.
>>> Now it is became worse: it reproduces occasionally.
>>> [FS version is 1.04, test_load = 2 active calls]
>>>
>>> I've got 2 logs: successful and not.
>>> Here is a bad_case:
>>>
>>> 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute
>>> java(/usr/local/freeswitch/scripts/fs2agi.jar
>>> org.starpound.fs2agi.Translator
>>> ${agi_url})
>>> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
>>> INFO: ***************************************************
>>> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
>>> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for
>>> session
>>> 2898ad41-4ec1-4628-89fd-651a93a7221d
>>> 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI
>>> application
>>> agi://localhost:4573/hello.agi?callId=929
>>> 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream
>>> handle!
>>>
>>> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>>> 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel
>>> sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17]
>>> 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready
>>> sofia/internal/2001!
>>> 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel
>>> [sofia/internal/2001] has
>>> been answered
>>> Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>>> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
>>> session
>>> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
>>> java.lang.Exception: Internal FreeSwitch failure while streamming file,
>>> see
>>> FreeSwitch logs for details
>>>         at
>>>
>>> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>>>         at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>>>         at org.starpound.fs2agi.Translator.run(Translator.java:56)
>>>         at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>>>         at
>>>
>>> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>>>         at
>>>
>>> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>>>         at java.lang.reflect.Method.invoke(Method.java:597)
>>>         at org.freeswitch.Launcher.launch(Launcher.java:80)
>>> 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179
>>> Hangup
>>> sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application
>>> agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for
>>> details.
>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup
>>> sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE]
>>> [DESTINATION_OUT_OF_ORDER]
>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17
>>> (sofia/external/6786081291 at 66.19.38.143) Ended
>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
>>> Channel
>>> sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY]
>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18
>>> (sofia/internal/2001) Ended
>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
>>> Channel
>>> sofia/internal/2001 [CS_DESTROY]
>>>
>>>
>>>
>>> Message
>>> "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>>> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
>>> session
>>> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
>>> ..."
>>> is sent from my app upon the onHangup().`
>>>
>>> And here is a good_case:
>>>
>>> 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute
>>> java(/usr/local/freeswitch/scripts/fs2agi.jar
>>> org.starpound.fs2agi.Translator
>>> ${agi_url})
>>> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run
>>> INFO: ***************************************************
>>> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run
>>> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for
>>> session
>>> 7c37369b-ffb2-4436-9288-a640047d0e5e
>>> 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI
>>> application
>>> agi://localhost:4573/hello.agi?callId=932
>>> 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream
>>> handle!
>>>
>>> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>>> 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel
>>> sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488]
>>> 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready
>>> sofia/internal/2001!
>>> 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel
>>> [sofia/internal/2001] has
>>> been answered
>>> Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>>> INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for
>>> session
>>> 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is:
>>> java.lang.Exception: Internal FreeSwitch failure while streamming file,
>>> see
>>> FreeSwitch logs for details
>>>         at
>>>
>>> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>>>         at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>>>         at org.starpound.fs2agi.Translator.run(Translator.java:56)
>>>         at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>>>         at
>>>
>>> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>>>         at
>>>
>>> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>>>         at java.lang.reflect.Method.invoke(Method.java:597)
>>>         at org.freeswitch.Launcher.launch(Launcher.java:80)
>>> 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application
>>> agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for
>>> details.
>>> 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port
>>> from
>>> 172.26.10.39:26402 to 91.190.120.190:26402
>>>
>>>
>>>
>>> Suggestions?
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>>
>>>> Anthony,
>>>>
>>>> >>As soon as you call uuid_bridge you are transferring both legs of the
>>>> call to bridge to each other.
>>>> >>This means your java app must exit so the channels can connect to each
>>>> other.
>>>>
>>>> I didn't know that. Now my java app is exiting upon the onHangup() call
>>>> so everything has become "ok". Thank you much.
>>>> I'll add note to the wiki about this issue.
>>>>
>>>> Artem
>>>>
>>>>
>>>>
>>>>
>>>> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> Your "annoying behaviour" is the exact behavior you should be getting
>>>>> considering what you told FS to do.
>>>>>
>>>>> As soon as you call uuid_bridge you are transferring both legs of the
>>>>> call to bridge to each other.
>>>>> This means your java app must exit so the channels can connect to each
>>>>> other.
>>>>>
>>>>> remember that you hangup hook can be called when the channel is
>>>>> transferred not only when it hangs up.
>>>>> you have to test which is happening based on the input to your
>>>>> callback.
>>>>>
>>>>>
>>>>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>>>>
>>>>>> Hi there!
>>>>>>
>>>>>> I've got annoying FS behavior:
>>>>>> There are 2 channels executing the same Java application (application
>>>>>> itself is an IVR). If I try to bridge them with uuid_bridged then both
>>>>>> channels are killed. Here is a log from FS console:
>>>>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165
>>>>>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE ->
>>>>>> CS_HIBERNATE
>>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>>>> called
>>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done
>>>>>> playing file
>>>>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done
>>>>>> playing file
>>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send
>>>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK]
>>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167
>>>>>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change
>>>>>> CS_EXECUTE -> CS_HIBERNATE
>>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>>>> called
>>>>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output:
>>>>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>>>>
>>>>>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG]
>>>>>> switch_core_session.c:933 Send signal
>>>>>> sofia/internal/1001 at master.agent.starpoundtec
>>>>>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send
>>>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK]
>>>>>>
>>>>>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking
>>>>>> stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>>>>>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send
>>>>>> signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK]
>>>>>>
>>>>>> (FS version is 1.0.4)
>>>>>>
>>>>>> Any thoughts?
>>>>>>
>>>>>>
>>>>>> Artem
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091204/d3f7e531/attachment-0001.html 

From shiyanov at gmail.com  Fri Dec  4 03:08:12 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Fri, 4 Dec 2009 14:08:12 +0300
Subject: [Freeswitch-dev] uuid_bridge kills both channels if they are
	executing java app
In-Reply-To: <d211f54a0912040220u72c3d25fqe559fe93d1c759e3@mail.gmail.com>
References: <d211f54a0911191146m6cc8fec7tebb26fffb1f639a2@mail.gmail.com>
	<191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com>
	<d211f54a0911210158x74005ea6mccbdf58948e7deba@mail.gmail.com>
	<d211f54a0912021108v52e8eebeu696ea40df358dd9d@mail.gmail.com>
	<191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com>
	<d211f54a0912031016u562c4e94n638f40620bce166@mail.gmail.com>
	<191c3a030912031041i1a941f6ev970c687d124b34d6@mail.gmail.com>
	<d211f54a0912040220u72c3d25fqe559fe93d1c759e3@mail.gmail.com>
Message-ID: <d211f54a0912040308w36320d8bv8d20b932ded4fae0@mail.gmail.com>

Bug created (FSCORE-508)
http://jira.freeswitch.org/browse/FSCORE-508




On Fri, Dec 4, 2009 at 1:20 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:

> Anthony,
>
> in pastebined log you could search "DESTINATION_OUT_OF_ORDER" and you'll
> find it.
> Anyway, I'll fire a bug in Jira.
>
> Artem
>
>
>
>
> On Thu, Dec 3, 2009 at 9:41 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> The case of the channel hangup with DESTINATION_OUT_OF_ORDER is not
>> depicted in your log.
>> can you capture the log of the entire procedure top to bottom and file a
>> jira ticket on http://jira.freeswitch.org and attach the trace as a .txt
>> file.
>>
>>
>>
>> On Thu, Dec 3, 2009 at 12:16 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>
>>> Hi there!
>>>
>>> This message is a forward from user-mail-list.
>>> I'm trying to fix such a problem:
>>> FreSwithch compiled from SVN-trunk, date = 11/02/2009.
>>>
>>> What is need: connect two users, initially one is on the home-grown
>>> java-based IVR and other party is off hook.
>>>
>>> What is done/got:
>>> User1 is on the java application, it represents simple IVR system, and
>>> the most used FS API operation is "streamFile".
>>> User2 is off hook.
>>> next:
>>> (mod_socket) create_uuid
>>> <get uuid_x>
>>> bgapi originate
>>> {origination_caller_id_name=User1}[origination_uuid=uuid_x]User1 &park()
>>> <User1 answers incomming call>
>>> <get event channel_User1 answered>
>>> <get event channel_User1 parked>
>>> uuid_bridge uuid_User1 uuid_User2
>>> <get event channel_User1 hangup, cause=NORMAL_CLEARING>
>>> <get event channel_User2 hangup, cause=*DESTINATION_OUT_OF_ORDER*>
>>> <both channel_User1 and channel_User2 are down>
>>>
>>>
>>> FS log is here: http://pastebin.freeswitch.org/11380
>>>
>>>
>>> Thank you much for any help,
>>> Artem
>>>
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>> Date: Wed, Dec 2, 2009 at 10:24 PM
>>> Subject: Re: [Freeswitch-users] uuid_bridge kills both channels if they
>>> are executing java app
>>> To: freeswitch-users at lists.freeswitch.org
>>>
>>>
>>> you should be working on SVN trunk if you are doing development, we are
>>> so far forward from 1.0.4 we can't do debugging very easily.
>>>
>>> I don't know all of the details of what you are trying to do but you are
>>> hitting some race conditions because of the async nature of the socket
>>> connection and the way you are using it.
>>>
>>>
>>>
>>>
>>> On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>>
>>>> I'm back again with the same issue.
>>>> Now it is became worse: it reproduces occasionally.
>>>> [FS version is 1.04, test_load = 2 active calls]
>>>>
>>>> I've got 2 logs: successful and not.
>>>> Here is a bad_case:
>>>>
>>>> 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute
>>>> java(/usr/local/freeswitch/scripts/fs2agi.jar
>>>> org.starpound.fs2agi.Translator
>>>> ${agi_url})
>>>> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
>>>> INFO: ***************************************************
>>>> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run
>>>> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for
>>>> session
>>>> 2898ad41-4ec1-4628-89fd-651a93a7221d
>>>> 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI
>>>> application
>>>> agi://localhost:4573/hello.agi?callId=929
>>>> 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream
>>>> handle!
>>>>
>>>> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>>>> 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel
>>>> sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17]
>>>> 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready
>>>> sofia/internal/2001!
>>>> 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel
>>>> [sofia/internal/2001] has
>>>> been answered
>>>> Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>>>> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
>>>> session
>>>> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
>>>> java.lang.Exception: Internal FreeSwitch failure while streamming file,
>>>> see
>>>> FreeSwitch logs for details
>>>>         at
>>>>
>>>> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>>>>         at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>>>>         at org.starpound.fs2agi.Translator.run(Translator.java:56)
>>>>         at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>>>>         at
>>>>
>>>> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>>>>         at
>>>>
>>>> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>>>>         at java.lang.reflect.Method.invoke(Method.java:597)
>>>>         at org.freeswitch.Launcher.launch(Launcher.java:80)
>>>> 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179
>>>> Hangup
>>>> sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application
>>>> agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for
>>>> details.
>>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup
>>>> sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE]
>>>> [DESTINATION_OUT_OF_ORDER]
>>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session
>>>> 17
>>>> (sofia/external/6786081291 at 66.19.38.143) Ended
>>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
>>>> Channel
>>>> sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY]
>>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session
>>>> 18
>>>> (sofia/internal/2001) Ended
>>>> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close
>>>> Channel
>>>> sofia/internal/2001 [CS_DESTROY]
>>>>
>>>>
>>>>
>>>> Message
>>>> "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>>>> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for
>>>> session
>>>> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is:
>>>> ..."
>>>> is sent from my app upon the onHangup().`
>>>>
>>>> And here is a good_case:
>>>>
>>>> 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute
>>>> java(/usr/local/freeswitch/scripts/fs2agi.jar
>>>> org.starpound.fs2agi.Translator
>>>> ${agi_url})
>>>> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run
>>>> INFO: ***************************************************
>>>> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run
>>>> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for
>>>> session
>>>> 7c37369b-ffb2-4436-9288-a640047d0e5e
>>>> 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI
>>>> application
>>>> agi://localhost:4573/hello.agi?callId=932
>>>> 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream
>>>> handle!
>>>>
>>>> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>>>> 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel
>>>> sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488]
>>>> 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready
>>>> sofia/internal/2001!
>>>> 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel
>>>> [sofia/internal/2001] has
>>>> been answered
>>>> Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed
>>>> INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for
>>>> session
>>>> 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is:
>>>> java.lang.Exception: Internal FreeSwitch failure while streamming file,
>>>> see
>>>> FreeSwitch logs for details
>>>>         at
>>>>
>>>> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36)
>>>>         at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48)
>>>>         at org.starpound.fs2agi.Translator.run(Translator.java:56)
>>>>         at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
>>>>         at
>>>>
>>>> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39)
>>>>         at
>>>>
>>>> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25)
>>>>         at java.lang.reflect.Method.invoke(Method.java:597)
>>>>         at org.freeswitch.Launcher.launch(Launcher.java:80)
>>>> 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application
>>>> agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for
>>>> details.
>>>> 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port
>>>> from
>>>> 172.26.10.39:26402 to 91.190.120.190:26402
>>>>
>>>>
>>>>
>>>> Suggestions?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>>>
>>>>> Anthony,
>>>>>
>>>>> >>As soon as you call uuid_bridge you are transferring both legs of the
>>>>> call to bridge to each other.
>>>>> >>This means your java app must exit so the channels can connect to
>>>>> each other.
>>>>>
>>>>> I didn't know that. Now my java app is exiting upon the onHangup() call
>>>>> so everything has become "ok". Thank you much.
>>>>> I'll add note to the wiki about this issue.
>>>>>
>>>>> Artem
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> Your "annoying behaviour" is the exact behavior you should be getting
>>>>>> considering what you told FS to do.
>>>>>>
>>>>>> As soon as you call uuid_bridge you are transferring both legs of the
>>>>>> call to bridge to each other.
>>>>>> This means your java app must exit so the channels can connect to each
>>>>>> other.
>>>>>>
>>>>>> remember that you hangup hook can be called when the channel is
>>>>>> transferred not only when it hangs up.
>>>>>> you have to test which is happening based on the input to your
>>>>>> callback.
>>>>>>
>>>>>>
>>>>>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov <shiyanov at gmail.com>wrote:
>>>>>>
>>>>>>> Hi there!
>>>>>>>
>>>>>>> I've got annoying FS behavior:
>>>>>>> There are 2 channels executing the same Java application (application
>>>>>>> itself is an IVR). If I try to bridge them with uuid_bridged then both
>>>>>>> channels are killed. Here is a log from FS console:
>>>>>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165
>>>>>>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE ->
>>>>>>> CS_HIBERNATE
>>>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>>>>> called
>>>>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done
>>>>>>> playing file
>>>>>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done
>>>>>>> playing file
>>>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send
>>>>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK]
>>>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167
>>>>>>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change
>>>>>>> CS_EXECUTE -> CS_HIBERNATE
>>>>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook
>>>>>>> called
>>>>>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>>>>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output:
>>>>>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>>>>>>
>>>>>>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG]
>>>>>>> switch_core_session.c:933 Send signal
>>>>>>> sofia/internal/1001 at master.agent.starpoundtec
>>>>>>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send
>>>>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK]
>>>>>>>
>>>>>>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking
>>>>>>> stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>>>>>>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send
>>>>>>> signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK]
>>>>>>>
>>>>>>> (FS version is 1.0.4)
>>>>>>>
>>>>>>> Any thoughts?
>>>>>>>
>>>>>>>
>>>>>>> Artem
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>>> ClueCon http://www.cluecon.com/
>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>>
>>>>>> AIM: anthm
>>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> iax:guest at conference.freeswitch.org/888
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:213-799-1400
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091204/fc4d8b45/attachment-0001.html 

From msc at freeswitch.org  Fri Dec  4 09:33:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 4 Dec 2009 09:33:32 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call Starting!
Message-ID: <87f2f3b90912040933h568df38ch87ca88c205d88e8f@mail.gmail.com>

FYI,

The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_04

Please call in! :)

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091204/d981579b/attachment.html 

From rsavage at KingBallow.com  Fri Dec  4 12:07:14 2009
From: rsavage at KingBallow.com (Reece Savage)
Date: Fri, 4 Dec 2009 14:07:14 -0600
Subject: [Freeswitch-dev] Aastra XML scripts.
Message-ID: <8E4ACA7747F7F641991455BC157390C80145007D@srv-nash-ex.mail.kingballow.com>

Would anyone be willing to port the Aastra XML scripts for Asterisk to
FreeSWITCH? I would be willing to sponser.

 

Reece Savage

Information Technology Manager

King & Ballow Law Offices

315 Union Street

Suite 1100

Nashville, TN  37201

Phone (615) 726-5525

Fax (615) 254-7907

rsavage at kingballow.com <mailto:rsavage at kingballow.com> 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091204/fcfad003/attachment.html 

From anthony.minessale at gmail.com  Fri Dec  4 15:43:39 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 4 Dec 2009 17:43:39 -0600
Subject: [Freeswitch-dev] mod event multicast drops some events on
	outbound call legs ?
In-Reply-To: <2134690418.4753.1259921967638.JavaMail.root@mx.voismart.com>
References: <37787900.4751.1259921827159.JavaMail.root@mx.voismart.com>
	<2134690418.4753.1259921967638.JavaMail.root@mx.voismart.com>
Message-ID: <191c3a030912041543o19df046du31f9b57aa80cde2b@mail.gmail.com>

please try trunk,

I changed the module to do blocking reads on the socket which should work
even better than sleeping less.

so.. about this multicast presence stuff???


On Fri, Dec 4, 2009 at 4:19 AM, Matteo <mbrancaleoni at voismart.it> wrote:

> Hi again,
>
> I've found a fix, but maybe the error is hidded somewhere else.
>
> If in the function
> SWITCH_MODULE_RUNTIME_FUNCTION(mod_event_multicast_runtime),
> line 515 in mod_event_multicast.c I change switch_yield(100000);  to a
> lower
> time, for example switch_yield(1000); it works ok.
>
> so maybe a loo long sleep time makes the multicast receiver loose
> packets if we have many events (I'm sending all the events for now,
> maybe with filters this will not happen).
>
> So, with switch_yield(1000); works ok.
>
> But this's the right fix?
>
> regards,
> matteo
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091204/b7623945/attachment.html 

From msc at freeswitch.org  Tue Dec  8 10:19:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 8 Dec 2009 10:19:39 -0800
Subject: [Freeswitch-dev] FreeSWITCH 1.0.5 is (almost) here!
Message-ID: <87f2f3b90912081019v5f481b45mf727ec6db339ef96@mail.gmail.com>

Greetings,

The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH
1.0.5 pre-release version. Please check out the release
announcement<http://www.freeswitch.org/node/220>.
Let's all get updated as soon as possible. Also, please report bugs right
away and follow up when the developers need further information. We have had
to close out some bugs due to lack of information from the one reporting.

Of course, those running SVN trunk are asked to do a "make current" as soon
as reasonably possible. The devs love it when you are on the latest trunk.
:)

Thanks again for all of your help! Let's keep up the good work and we'll
have 1.0.5 available in no time.

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091208/1e42be41/attachment.html 

From mbrancaleoni at voismart.it  Wed Dec  9 01:57:40 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Wed, 9 Dec 2009 10:57:40 +0100 (CET)
Subject: [Freeswitch-dev] mod event multicast drops some events
	on	outbound call legs ?
In-Reply-To: <191c3a030912041543o19df046du31f9b57aa80cde2b@mail.gmail.com>
Message-ID: <1816949746.5486.1260352660356.JavaMail.root@mx.voismart.com>

Hi,

sorry for late response, was national holiday here in .it :)

----- "Anthony Minessale" <anthony.minessale at gmail.com> ha scritto:
> please try trunk,
> 
> I changed the module to do blocking reads on the socket which should
> work even better than sleeping less.

tried now, seems ok. great !

> so.. about this multicast presence stuff???

created another thread over dev ml.

regards,
matteo.


From mbrancaleoni at voismart.it  Wed Dec  9 02:23:52 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Wed, 9 Dec 2009 11:23:52 +0100 (CET)
Subject: [Freeswitch-dev] multicast presence,
	mwi and some registrations thoughts
In-Reply-To: <2068231592.5496.1260352970037.JavaMail.root@mx.voismart.com>
Message-ID: <2050276014.5543.1260354232878.JavaMail.root@mx.voismart.com>

Hi all,

as promised, I've created a small patch to handle PRESENCE_IN
and MWI events over multicast (using mod event multicast).

When used along registrations sharing with multicast, you
can send presence info over the mcast-bus :) and deliver
the information to the phones, even if the call is not
handled on the same switch the phone sent the subscription to.
To make it work, the sip domain must be the all for all the machines
and on the phones, otherwise the phones will ignore them.

This allows distributed BLF and MWI notifications :)

BUT doing that I noticed a issue that must be addressed imho,
which arises when you handle NAT.

"distributed blf" works behind nat right now, since
the event will get sent to all switches, but delivered
only to the phone were fs knows about the subscription.
Which is the one where the phone registers to.

"distributed mwi" results into sending N copies of
same MWI notifications , where N is the number of switches.
Behind NAT the phone will receive only the one sent
by the switch where it's registered to.

We can avoid sending multiple event if we're able to know
which is the "original" server where the phone registered to.

This is also needed to call the phone:
think about a distributed registration cluster.
if Alice (reg'ed on FS1) calls Bob on (reg'ed on FS2), right now
the call is routed from FS1 to Bob, since FS1 knows where Bob is.
This works ok without NAT.

But with NAT FS1 cannot contact Bob, because Bob is on FS2 "really"
and can receive calls from FS2 (nat pinhole).

The solution is to check where Bob is (looking to registration table),
get the "real" server and route the call from FS1 to FS2 then to Bob.

But right now, sofia registration does not have a field holding
the "original" server.
If we want to leverage on distributed registration using multicast,
I think knowing the original server is "a must".

the patch is pretty trivial, but we must add a field onto the db.

Doing that, we can also filter MWI notification, and send them
only if the phone is "really" registered on the server which receives
the MWI event over the mcast-bus .

What do you think about it?

Regards,
mat

P.S. this is my first patch to FS, so please be kind on my mistakes :)

-------------- next part --------------
A non-text attachment was scrubbed...
Name: sofia.c.diff
Type: text/x-patch
Size: 3702 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091209/6b750999/attachment.bin 

From mike at jerris.com  Wed Dec  9 05:51:59 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 9 Dec 2009 08:51:59 -0500
Subject: [Freeswitch-dev] multicast presence,
	mwi and some registrations thoughts
In-Reply-To: <2050276014.5543.1260354232878.JavaMail.root@mx.voismart.com>
References: <2050276014.5543.1260354232878.JavaMail.root@mx.voismart.com>
Message-ID: <75609FB0-69FE-46DC-A95F-A00667F9C861@jerris.com>

Please post this on a bug on jira.freeswitch.org for further discussion.

Mike

On Dec 9, 2009, at 5:23 AM, Matteo wrote:

> Hi all,
> 
> as promised, I've created a small patch to handle PRESENCE_IN
> and MWI events over multicast (using mod event multicast).
> 
> When used along registrations sharing with multicast, you
> can send presence info over the mcast-bus :) and deliver
> the information to the phones, even if the call is not
> handled on the same switch the phone sent the subscription to.
> To make it work, the sip domain must be the all for all the machines
> and on the phones, otherwise the phones will ignore them.
> 
> This allows distributed BLF and MWI notifications :)
> 
> BUT doing that I noticed a issue that must be addressed imho,
> which arises when you handle NAT.
> 
> "distributed blf" works behind nat right now, since
> the event will get sent to all switches, but delivered
> only to the phone were fs knows about the subscription.
> Which is the one where the phone registers to.
> 
> "distributed mwi" results into sending N copies of
> same MWI notifications , where N is the number of switches.
> Behind NAT the phone will receive only the one sent
> by the switch where it's registered to.
> 
> We can avoid sending multiple event if we're able to know
> which is the "original" server where the phone registered to.
> 
> This is also needed to call the phone:
> think about a distributed registration cluster.
> if Alice (reg'ed on FS1) calls Bob on (reg'ed on FS2), right now
> the call is routed from FS1 to Bob, since FS1 knows where Bob is.
> This works ok without NAT.
> 
> But with NAT FS1 cannot contact Bob, because Bob is on FS2 "really"
> and can receive calls from FS2 (nat pinhole).
> 
> The solution is to check where Bob is (looking to registration table),
> get the "real" server and route the call from FS1 to FS2 then to Bob.
> 
> But right now, sofia registration does not have a field holding
> the "original" server.
> If we want to leverage on distributed registration using multicast,
> I think knowing the original server is "a must".
> 
> the patch is pretty trivial, but we must add a field onto the db.
> 
> Doing that, we can also filter MWI notification, and send them
> only if the phone is "really" registered on the server which receives
> the MWI event over the mcast-bus .
> 
> What do you think about it?
> 
> Regards,
> mat
> 
> P.S. this is my first patch to FS, so please be kind on my mistakes :)
> 
> <sofia.c.diff>_______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From brian at freeswitch.org  Wed Dec  9 05:52:18 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 9 Dec 2009 07:52:18 -0600
Subject: [Freeswitch-dev] multicast presence,
	mwi and some registrations thoughts
In-Reply-To: <2050276014.5543.1260354232878.JavaMail.root@mx.voismart.com>
References: <2050276014.5543.1260354232878.JavaMail.root@mx.voismart.com>
Message-ID: <8F7B1E92-1681-4F23-8B70-97208798E0B5@freeswitch.org>

Please post it on http://jira.freeswitch.org

Thanks,
Brian

On Dec 9, 2009, at 4:23 AM, Matteo wrote:

> Hi all,
>
> as promised, I've created a small patch to handle PRESENCE_IN
> and MWI events over multicast (using mod event multicast).
>
> When used along registrations sharing with multicast, you
> can send presence info over the mcast-bus :) and deliver
> the information to the phones, even if the call is not
> handled on the same switch the phone sent the subscription to.
> To make it work, the sip domain must be the all for all the machines
> and on the phones, otherwise the phones will ignore them.
>
> This allows distributed BLF and MWI notifications :)
>
> BUT doing that I noticed a issue that must be addressed imho,
> which arises when you handle NAT.
>
> "distributed blf" works behind nat right now, since
> the event will get sent to all switches, but delivered
> only to the phone were fs knows about the subscription.
> Which is the one where the phone registers to.
>
> "distributed mwi" results into sending N copies of
> same MWI notifications , where N is the number of switches.
> Behind NAT the phone will receive only the one sent
> by the switch where it's registered to.
>
> We can avoid sending multiple event if we're able to know
> which is the "original" server where the phone registered to.
>
> This is also needed to call the phone:
> think about a distributed registration cluster.
> if Alice (reg'ed on FS1) calls Bob on (reg'ed on FS2), right now
> the call is routed from FS1 to Bob, since FS1 knows where Bob is.
> This works ok without NAT.
>
> But with NAT FS1 cannot contact Bob, because Bob is on FS2 "really"
> and can receive calls from FS2 (nat pinhole).
>
> The solution is to check where Bob is (looking to registration table),
> get the "real" server and route the call from FS1 to FS2 then to Bob.
>
> But right now, sofia registration does not have a field holding
> the "original" server.
> If we want to leverage on distributed registration using multicast,
> I think knowing the original server is "a must".
>
> the patch is pretty trivial, but we must add a field onto the db.
>
> Doing that, we can also filter MWI notification, and send them
> only if the phone is "really" registered on the server which receives
> the MWI event over the mcast-bus .
>
> What do you think about it?
>
> Regards,
> mat
>
> P.S. this is my first patch to FS, so please be kind on my mistakes :)
>
> <sofia.c.diff>_______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From mbrancaleoni at voismart.it  Wed Dec  9 06:17:55 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Wed, 9 Dec 2009 15:17:55 +0100 (CET)
Subject: [Freeswitch-dev] multicast presence,
 mwi and some registrations thoughts
In-Reply-To: <75609FB0-69FE-46DC-A95F-A00667F9C861@jerris.com>
Message-ID: <1169924018.5745.1260368275221.JavaMail.root@mx.voismart.com>

Hi,

----- "Michael Jerris" <mike at jerris.com> ha scritto:
> Please post this on a bug on jira.freeswitch.org for further
> discussion.

Here we go:
http://jira.freeswitch.org/browse/MODSOFIA-46

Regards,
mat


From anthony.minessale at gmail.com  Wed Dec  9 07:54:34 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Dec 2009 09:54:34 -0600
Subject: [Freeswitch-dev] multicast presence,
	mwi and some registrations 	thoughts
In-Reply-To: <2050276014.5543.1260354232878.JavaMail.root@mx.voismart.com>
References: <2068231592.5496.1260352970037.JavaMail.root@mx.voismart.com>
	<2050276014.5543.1260354232878.JavaMail.root@mx.voismart.com>
Message-ID: <191c3a030912090754v6d0c99a6qc8800822de5ebe59@mail.gmail.com>

I'm open to the idea as long as the patch properly deals with the new col to
make it upgrade itself.


On Wed, Dec 9, 2009 at 4:23 AM, Matteo <mbrancaleoni at voismart.it> wrote:

> Hi all,
>
> as promised, I've created a small patch to handle PRESENCE_IN
> and MWI events over multicast (using mod event multicast).
>
> When used along registrations sharing with multicast, you
> can send presence info over the mcast-bus :) and deliver
> the information to the phones, even if the call is not
> handled on the same switch the phone sent the subscription to.
> To make it work, the sip domain must be the all for all the machines
> and on the phones, otherwise the phones will ignore them.
>
> This allows distributed BLF and MWI notifications :)
>
> BUT doing that I noticed a issue that must be addressed imho,
> which arises when you handle NAT.
>
> "distributed blf" works behind nat right now, since
> the event will get sent to all switches, but delivered
> only to the phone were fs knows about the subscription.
> Which is the one where the phone registers to.
>
> "distributed mwi" results into sending N copies of
> same MWI notifications , where N is the number of switches.
> Behind NAT the phone will receive only the one sent
> by the switch where it's registered to.
>
> We can avoid sending multiple event if we're able to know
> which is the "original" server where the phone registered to.
>
> This is also needed to call the phone:
> think about a distributed registration cluster.
> if Alice (reg'ed on FS1) calls Bob on (reg'ed on FS2), right now
> the call is routed from FS1 to Bob, since FS1 knows where Bob is.
> This works ok without NAT.
>
> But with NAT FS1 cannot contact Bob, because Bob is on FS2 "really"
> and can receive calls from FS2 (nat pinhole).
>
> The solution is to check where Bob is (looking to registration table),
> get the "real" server and route the call from FS1 to FS2 then to Bob.
>
> But right now, sofia registration does not have a field holding
> the "original" server.
> If we want to leverage on distributed registration using multicast,
> I think knowing the original server is "a must".
>
> the patch is pretty trivial, but we must add a field onto the db.
>
> Doing that, we can also filter MWI notification, and send them
> only if the phone is "really" registered on the server which receives
> the MWI event over the mcast-bus .
>
> What do you think about it?
>
> Regards,
> mat
>
> P.S. this is my first patch to FS, so please be kind on my mistakes :)
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091209/61e6411d/attachment.html 

From mbrancaleoni at voismart.it  Wed Dec  9 10:56:41 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Wed, 9 Dec 2009 19:56:41 +0100 (CET)
Subject: [Freeswitch-dev] multicast presence,
 mwi and some registrations 	thoughts
In-Reply-To: <191c3a030912090754v6d0c99a6qc8800822de5ebe59@mail.gmail.com>
Message-ID: <715964499.5931.1260385001802.JavaMail.root@mx.voismart.com>

Hi,

----- "Anthony Minessale" <anthony.minessale at gmail.com> ha scritto:
> I'm open to the idea as long as the patch properly deals with the new
> col to make it upgrade itself.

I've checked mod sofia, and no table updates occur right now.

What do you think can be the best approach ?
Write a test sql in sofia_glue_init_sql and if failed alter
the table with the new col ?

of course the create will have the new col by default.

is this the right way to go ?

thanks,
mat


From vipkilla at gmail.com  Wed Dec  9 11:06:09 2009
From: vipkilla at gmail.com (vip killa)
Date: Wed, 9 Dec 2009 14:06:09 -0500
Subject: [Freeswitch-dev] dynamic update page
Message-ID: <957f61370912091106w57f78f7etdbbc722a86734a8@mail.gmail.com>

could someone tell me how it would be possible to have a webpage reload when
an extension is dialed. for example a user is looking at a webpage
displaying conference information. when someone enters the conference, that
webpage is reloaded displaying updated information about the conference
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091209/d818eabb/attachment.html 

From mike at jerris.com  Wed Dec  9 11:36:44 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 9 Dec 2009 14:36:44 -0500
Subject: [Freeswitch-dev] multicast presence,
	mwi and some registrations  thoughts
In-Reply-To: <715964499.5931.1260385001802.JavaMail.root@mx.voismart.com>
References: <715964499.5931.1260385001802.JavaMail.root@mx.voismart.com>
Message-ID: <A324E4EC-6095-4A7D-AECE-AA5D67BA6897@jerris.com>

There are certainly things in there to create the tables right if they have changed. grep for test_reactive

Mike

On Dec 9, 2009, at 1:56 PM, Matteo wrote:

> Hi,
> 
> ----- "Anthony Minessale" <anthony.minessale at gmail.com> ha scritto:
>> I'm open to the idea as long as the patch properly deals with the new
>> col to make it upgrade itself.
> 
> I've checked mod sofia, and no table updates occur right now.
> 
> What do you think can be the best approach ?
> Write a test sql in sofia_glue_init_sql and if failed alter
> the table with the new col ?
> 
> of course the create will have the new col by default.
> 
> is this the right way to go ?
> 
> thanks,
> mat
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Wed Dec  9 11:53:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Dec 2009 13:53:03 -0600
Subject: [Freeswitch-dev] multicast presence,
	mwi and some registrations 	thoughts
In-Reply-To: <A324E4EC-6095-4A7D-AECE-AA5D67BA6897@jerris.com>
References: <715964499.5931.1260385001802.JavaMail.root@mx.voismart.com>
	<A324E4EC-6095-4A7D-AECE-AA5D67BA6897@jerris.com>
Message-ID: <191c3a030912091153q77d162e1q4da164bc951dab2f@mail.gmail.com>

yes just mix the new col into the statement that it tests with then that
will make it fail and recreate the tables.


On Wed, Dec 9, 2009 at 1:36 PM, Michael Jerris <mike at jerris.com> wrote:

> There are certainly things in there to create the tables right if they have
> changed. grep for test_reactive
>
> Mike
>
> On Dec 9, 2009, at 1:56 PM, Matteo wrote:
>
> > Hi,
> >
> > ----- "Anthony Minessale" <anthony.minessale at gmail.com> ha scritto:
> >> I'm open to the idea as long as the patch properly deals with the new
> >> col to make it upgrade itself.
> >
> > I've checked mod sofia, and no table updates occur right now.
> >
> > What do you think can be the best approach ?
> > Write a test sql in sofia_glue_init_sql and if failed alter
> > the table with the new col ?
> >
> > of course the create will have the new col by default.
> >
> > is this the right way to go ?
> >
> > thanks,
> > mat
> >
> > _______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091209/4a5dbd59/attachment-0001.html 

From anthony.minessale at gmail.com  Wed Dec  9 17:52:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 9 Dec 2009 19:52:05 -0600
Subject: [Freeswitch-dev] dynamic update page
In-Reply-To: <957f61370912091106w57f78f7etdbbc722a86734a8@mail.gmail.com>
References: <957f61370912091106w57f78f7etdbbc722a86734a8@mail.gmail.com>
Message-ID: <191c3a030912091752h2d87bdc4k4906fd7111164764@mail.gmail.com>

you would need to use comet and a server side event socket client to detect
changes in FS and push events to your web app.
otherwise you would have to do polling

On Wed, Dec 9, 2009 at 1:06 PM, vip killa <vipkilla at gmail.com> wrote:

> could someone tell me how it would be possible to have a webpage reload
> when an extension is dialed. for example a user is looking at a webpage
> displaying conference information. when someone enters the conference, that
> webpage is reloaded displaying updated information about the conference
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091209/c39b95ca/attachment.html 

From vipkilla at gmail.com  Thu Dec 10 06:59:08 2009
From: vipkilla at gmail.com (vip killa)
Date: Thu, 10 Dec 2009 09:59:08 -0500
Subject: [Freeswitch-dev] dynamic update page
In-Reply-To: <191c3a030912091752h2d87bdc4k4906fd7111164764@mail.gmail.com>
References: <957f61370912091106w57f78f7etdbbc722a86734a8@mail.gmail.com>
	<191c3a030912091752h2d87bdc4k4906fd7111164764@mail.gmail.com>
Message-ID: <957f61370912100659y80675fdg8a9563d94fc37bd3@mail.gmail.com>

has anyone done this using comet and a server side event socket?

On Wed, Dec 9, 2009 at 8:52 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you would need to use comet and a server side event socket client to detect
> changes in FS and push events to your web app.
> otherwise you would have to do polling
>
> On Wed, Dec 9, 2009 at 1:06 PM, vip killa <vipkilla at gmail.com> wrote:
>
>> could someone tell me how it would be possible to have a webpage reload
>> when an extension is dialed. for example a user is looking at a webpage
>> displaying conference information. when someone enters the conference, that
>> webpage is reloaded displaying updated information about the conference
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091210/0b092cec/attachment.html 

From mbrancaleoni at voismart.it  Thu Dec 10 09:23:47 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Thu, 10 Dec 2009 18:23:47 +0100 (CET)
Subject: [Freeswitch-dev] multicast presence,
 mwi and some registrations 	thoughts
In-Reply-To: <191c3a030912091153q77d162e1q4da164bc951dab2f@mail.gmail.com>
Message-ID: <161820946.6418.1260465827585.JavaMail.root@mx.voismart.com>

Hi,

----- "Anthony Minessale" <anthony.minessale at gmail.com> ha scritto:
> yes just mix the new col into the statement that it tests with then
> that will make it fail and recreate the tables.

ok, I just finished a patch where the db is updated correctly
and mcast registration stores the original server_host and
original hostname.

Now I'm adding the possibility to tell sofia to dial
user at original_server_host instead to real user contact
(useful for nat).

Hope to finish soon :)

regards,
mat


From msc at freeswitch.org  Thu Dec 10 16:49:23 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 10 Dec 2009 16:49:23 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call Agenda For Dec 11
Message-ID: <87f2f3b90912101649x500c0bc9kfaefc770fbf25a8@mail.gmail.com>

FYI,

Here's the agenda for tomorrow's conference call:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_11

Please be ready to join at 11AM CST! :) Don't forget to bring your agenda
items, questions, and a willingness to help out with our various janitor
projects.

Thanks!
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091210/256f225a/attachment.html 

From mbrancaleoni at voismart.it  Fri Dec 11 01:57:52 2009
From: mbrancaleoni at voismart.it (Matteo)
Date: Fri, 11 Dec 2009 10:57:52 +0100 (CET)
Subject: [Freeswitch-dev] multicast presence,
 mwi and some registrations 	thoughts
In-Reply-To: <161820946.6418.1260465827585.JavaMail.root@mx.voismart.com>
Message-ID: <1351296982.6611.1260525472579.JavaMail.root@mx.voismart.com>

Hi,

final(?) patch on jira, along with "what changed"

http://jira.freeswitch.org/browse/MODSOFIA-46

patch => mod_sofia.diff, 11/Dec/09 03:54 AM

regards,
matteo


From msc at freeswitch.org  Fri Dec 11 09:04:07 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 11 Dec 2009 09:04:07 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call Starting!
Message-ID: <87f2f3b90912110904y716e85f9o6afa42a1960da70f@mail.gmail.com>

Come one, come all!
http://bit.ly/8KzHCZ

Talk to you soon!
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091211/75833c93/attachment.html 

From msc at freeswitch.org  Mon Dec 14 11:45:12 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 14 Dec 2009 11:45:12 -0800
Subject: [Freeswitch-dev] FreeSWITCH 1.0.5pre9 is now available
Message-ID: <87f2f3b90912141145t6c193b52yf464f208db7b5d59@mail.gmail.com>

FYI,

The latest pre-release is now available. Usual information is available
here: http://www.freeswitch.org/node/222

Please update as soon as you can. (SVN trunk users do the "make current"
thing please.) We need your testing and feedback please!

Many thanks for continuing to support FreeSWITCH.
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091214/34da2dd4/attachment.html 

From msc at freeswitch.org  Tue Dec 15 11:35:45 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 15 Dec 2009 11:35:45 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call Agenda For Dec
	18 - Need Your Items
Message-ID: <87f2f3b90912151135m4677a553k397b36369a924add@mail.gmail.com>

Hello friends,

Just to let you know, I have posted the FS weekly conf call agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14

It's pretty clean at this point so if you've got things that you'd like to
discuss with the group then please add your items to the list. If you have
items that require attention, like documentation and janitorial items then
by all means drop those on the list as well. One thing we do need to discuss
is how we will accomplish screen casting. We are going to have presentations
and we want as many people as possible to be able to view the screen while
listening to the conference.

Thanks,
Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091215/81c8634a/attachment.html 

From kristoff.bonne at skypro.be  Wed Dec 16 00:33:18 2009
From: kristoff.bonne at skypro.be (Kristoff Bonne)
Date: Wed, 16 Dec 2009 09:33:18 +0100
Subject: [Freeswitch-dev] coredump (dingaling)
Message-ID: <4B289B4E.1070305@skypro.be>

Hi All,


I don't know if this is the correct place for this, but I have a problem
so perhaps somebody here can help.

I've been playing around with freeswitch for about a week now using an
old mac-mini (ppc G3) running debian. Yesterday, I wanted to give
googletalk (dingaling) a try but got a coredump on freeswitch when I
dialed by gtalk account from outside.


Is there somewhere I can open a "ticket" for this? Can I upload the
corefile (about 1.2 MB compressed) somewhere?


This is on a PPC platform so I don't know if you people will have any
use of it.



Cheerio! Kr. Bonne.

-- 
jabber/gtalk: kristoff at krbonne.net

-------------- next part --------------
A non-text attachment was scrubbed...
Name: kristoff_bonne.vcf
Type: text/x-vcard
Size: 190 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091216/d99789d4/attachment-0001.vcf 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 261 bytes
Desc: OpenPGP digital signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091216/d99789d4/attachment-0001.bin 

From brian at freeswitch.org  Wed Dec 16 06:54:12 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 16 Dec 2009 08:54:12 -0600
Subject: [Freeswitch-dev] coredump (dingaling)
In-Reply-To: <4B289B4E.1070305@skypro.be>
References: <4B289B4E.1070305@skypro.be>
Message-ID: <6FBB10D9-76D4-4B44-AFA9-2CD86523BAAA@freeswitch.org>

Please see http://wiki.freeswitch.org/wiki/Reporting_Bugs

The core file has NO use at all without running the gdb commands on  
the same box that generated the core file... Uploading it is a futile  
attempt.

Please read the reporting bugs guide and it will tell you what steps  
and commands to take to collect the info up and report the issue on  
jira.

Thanks,
Brian

On Dec 16, 2009, at 2:33 AM, Kristoff Bonne wrote:

> Hi All,
>
>
> I don't know if this is the correct place for this, but I have a  
> problem
> so perhaps somebody here can help.
>
> I've been playing around with freeswitch for about a week now using an
> old mac-mini (ppc G3) running debian. Yesterday, I wanted to give
> googletalk (dingaling) a try but got a coredump on freeswitch when I
> dialed by gtalk account from outside.
>
>
> Is there somewhere I can open a "ticket" for this? Can I upload the
> corefile (about 1.2 MB compressed) somewhere?
>
>
> This is on a PPC platform so I don't know if you people will have any
> use of it.
>
>
>
> Cheerio! Kr. Bonne.
>
> -- 
> jabber/gtalk: kristoff at krbonne.net
>
> <kristoff_bonne.vcf>_______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Wed Dec 16 08:18:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Dec 2009 10:18:24 -0600
Subject: [Freeswitch-dev] coredump (dingaling)
In-Reply-To: <6FBB10D9-76D4-4B44-AFA9-2CD86523BAAA@freeswitch.org>
References: <4B289B4E.1070305@skypro.be>
	<6FBB10D9-76D4-4B44-AFA9-2CD86523BAAA@freeswitch.org>
Message-ID: <191c3a030912160818g7443c131o78e252d0384f6890@mail.gmail.com>

also make sure its the SVN release because we dont take bug reports on
anything but latest trunk.


On Wed, Dec 16, 2009 at 8:54 AM, Brian West <brian at freeswitch.org> wrote:

> Please see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> The core file has NO use at all without running the gdb commands on
> the same box that generated the core file... Uploading it is a futile
> attempt.
>
> Please read the reporting bugs guide and it will tell you what steps
> and commands to take to collect the info up and report the issue on
> jira.
>
> Thanks,
> Brian
>
> On Dec 16, 2009, at 2:33 AM, Kristoff Bonne wrote:
>
> > Hi All,
> >
> >
> > I don't know if this is the correct place for this, but I have a
> > problem
> > so perhaps somebody here can help.
> >
> > I've been playing around with freeswitch for about a week now using an
> > old mac-mini (ppc G3) running debian. Yesterday, I wanted to give
> > googletalk (dingaling) a try but got a coredump on freeswitch when I
> > dialed by gtalk account from outside.
> >
> >
> > Is there somewhere I can open a "ticket" for this? Can I upload the
> > corefile (about 1.2 MB compressed) somewhere?
> >
> >
> > This is on a PPC platform so I don't know if you people will have any
> > use of it.
> >
> >
> >
> > Cheerio! Kr. Bonne.
> >
> > --
> > jabber/gtalk: kristoff at krbonne.net
> >
> > <kristoff_bonne.vcf>_______________________________________________
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091216/d349c0a0/attachment.html 

From kristoff.bonne at skypro.be  Wed Dec 16 11:52:41 2009
From: kristoff.bonne at skypro.be (Kristoff Bonne)
Date: Wed, 16 Dec 2009 20:52:41 +0100
Subject: [Freeswitch-dev] coredump (dingaling)
In-Reply-To: <191c3a030912160818g7443c131o78e252d0384f6890@mail.gmail.com>
References: <4B289B4E.1070305@skypro.be>	<6FBB10D9-76D4-4B44-AFA9-2CD86523BAAA@freeswitch.org>
	<191c3a030912160818g7443c131o78e252d0384f6890@mail.gmail.com>
Message-ID: <4B293A89.2050802@skypro.be>

Hi all,


OK. Here is how far I have got sofar.

- first: "make current" -> installed latest version

- problem still there. However, not always. I do have had cases where
freeswitch did not crash, or crashed when the gtalk-session is terminated.

- The "console loglvel 7" doesn't really help. Freeswitch bombs out
before having the change to print any debug-info.
(or is there any logfile I should look for?)


- I have run "tcpdump" on both sides (both on the server and a client) a
number of times.
-> The positive side is that -as I do have some sessions that did work-
I can compair with a "good" session.
-> The negative side is that it doesn't really help that much.

The problem seams to be somewhere in, or and the end of the jabber
session. Looking at the "good" session, I see some traffic to the
stun-server. In the traces where the freeswitch crashes, I never see
these STUN-messages.
So it looks like the problem is somewhere (perhaps with the exchange of
IP-addresses for the ICE-session).

However, I cannot see that for sure. The jabber-session with googletalk
is SSL-encrypted so I cannot see what is exactly in there.


BTW. My voip-client is "Empaty" on ubuntu 9.10

Now, I did not additional test. I disabled IPv6 on my server and client
(I run a IPv6-enabled network at home). I tried with other clients (like
the voice-chat webapplication of google on my wive's mac; (both using
firefox and google chrome).
Same problem. Sometimes it works, but 3 out of 4 times it does not.



Anybody any ideas on how to procede with this? Anybody any ideas of what
to try next?

How does it come that I do not have any debug-information on the
freeswitch console?




Cheerio! Kr. Bonne.


Anthony Minessale schreef:
> also make sure its the SVN release because we dont take bug reports on
> anything but latest trunk.
>
>
> On Wed, Dec 16, 2009 at 8:54 AM, Brian West <brian at freeswitch.org> wrote:
>
>   
>> Please see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>> The core file has NO use at all without running the gdb commands on
>> the same box that generated the core file... Uploading it is a futile
>> attempt.
>>
>> Please read the reporting bugs guide and it will tell you what steps
>> and commands to take to collect the info up and report the issue on
>> jira.
>>
>> Thanks,
>> Brian
>>
>> On Dec 16, 2009, at 2:33 AM, Kristoff Bonne wrote:
>>
>>     
>>> Hi All,
>>>
>>>
>>> I don't know if this is the correct place for this, but I have a
>>> problem
>>> so perhaps somebody here can help.
>>>
>>> I've been playing around with freeswitch for about a week now using an
>>> old mac-mini (ppc G3) running debian. Yesterday, I wanted to give
>>> googletalk (dingaling) a try but got a coredump on freeswitch when I
>>> dialed by gtalk account from outside.
>>>
>>>
>>> Is there somewhere I can open a "ticket" for this? Can I upload the
>>> corefile (about 1.2 MB compressed) somewhere?
>>>
>>>
>>> This is on a PPC platform so I don't know if you people will have any
>>> use of it.
>>>
>>>
>>>
>>> Cheerio! Kr. Bonne.
>>>
>>> --
>>> jabber/gtalk: kristoff at krbonne.net
>>>
>>> <kristoff_bonne.vcf>_______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>       
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   


-- 
jabber/gtalk: kristoff at krbonne.net

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091216/202da92c/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kristoff_bonne.vcf
Type: text/x-vcard
Size: 190 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091216/202da92c/attachment.vcf 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 261 bytes
Desc: OpenPGP digital signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091216/202da92c/attachment.bin 

From msc at freeswitch.org  Wed Dec 16 12:04:49 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 16 Dec 2009 12:04:49 -0800
Subject: [Freeswitch-dev] coredump (dingaling)
In-Reply-To: <4B293A89.2050802@skypro.be>
References: <4B289B4E.1070305@skypro.be>
	<6FBB10D9-76D4-4B44-AFA9-2CD86523BAAA@freeswitch.org>
	<191c3a030912160818g7443c131o78e252d0384f6890@mail.gmail.com>
	<4B293A89.2050802@skypro.be>
Message-ID: <87f2f3b90912161204v652ac2a9s64d5d41f71b52bba@mail.gmail.com>

Well, now that you have demonstrated that you can repeat this on the latest
trunk you can now do the gdb stuff on the core.xxxx file that gets created.
Check out this page that has some information on gathering info:
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Advanced_Debugging_Techniques

That page also has info on how to properly submit a JIRA, pastebin, etc.
-MC

On Wed, Dec 16, 2009 at 11:52 AM, Kristoff Bonne
<kristoff.bonne at skypro.be>wrote:

>  Hi all,
>
>
> OK. Here is how far I have got sofar.
>
> - first: "make current" -> installed latest version
>
> - problem still there. However, not always. I do have had cases where
> freeswitch did not crash, or crashed when the gtalk-session is terminated.
>
> - The "console loglvel 7" doesn't really help. Freeswitch bombs out before
> having the change to print any debug-info.
> (or is there any logfile I should look for?)
>
>
> - I have run "tcpdump" on both sides (both on the server and a client) a
> number of times.
> -> The positive side is that -as I do have some sessions that did work- I
> can compair with a "good" session.
> -> The negative side is that it doesn't really help that much.
>
> The problem seams to be somewhere in, or and the end of the jabber session.
> Looking at the "good" session, I see some traffic to the stun-server. In the
> traces where the freeswitch crashes, I never see these STUN-messages.
> So it looks like the problem is somewhere (perhaps with the exchange of
> IP-addresses for the ICE-session).
>
> However, I cannot see that for sure. The jabber-session with googletalk is
> SSL-encrypted so I cannot see what is exactly in there.
>
>
> BTW. My voip-client is "Empaty" on ubuntu 9.10
>
> Now, I did not additional test. I disabled IPv6 on my server and client (I
> run a IPv6-enabled network at home). I tried with other clients (like the
> voice-chat webapplication of google on my wive's mac; (both using firefox
> and google chrome).
> Same problem. Sometimes it works, but 3 out of 4 times it does not.
>
>
>
> Anybody any ideas on how to procede with this? Anybody any ideas of what to
> try next?
>
> How does it come that I do not have any debug-information on the freeswitch
> console?
>
>
>
>
> Cheerio! Kr. Bonne.
>
>
> Anthony Minessale schreef:
>
> also make sure its the SVN release because we dont take bug reports on
> anything but latest trunk.
>
>
> On Wed, Dec 16, 2009 at 8:54 AM, Brian West <brian at freeswitch.org> <brian at freeswitch.org> wrote:
>
>
>
>  Please see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> The core file has NO use at all without running the gdb commands on
> the same box that generated the core file... Uploading it is a futile
> attempt.
>
> Please read the reporting bugs guide and it will tell you what steps
> and commands to take to collect the info up and report the issue on
> jira.
>
> Thanks,
> Brian
>
> On Dec 16, 2009, at 2:33 AM, Kristoff Bonne wrote:
>
>
>
>  Hi All,
>
>
> I don't know if this is the correct place for this, but I have a
> problem
> so perhaps somebody here can help.
>
> I've been playing around with freeswitch for about a week now using an
> old mac-mini (ppc G3) running debian. Yesterday, I wanted to give
> googletalk (dingaling) a try but got a coredump on freeswitch when I
> dialed by gtalk account from outside.
>
>
> Is there somewhere I can open a "ticket" for this? Can I upload the
> corefile (about 1.2 MB compressed) somewhere?
>
>
> This is on a PPC platform so I don't know if you people will have any
> use of it.
>
>
>
> Cheerio! Kr. Bonne.
>
> --
> jabber/gtalk: kristoff at krbonne.net
>
> <kristoff_bonne.vcf>_______________________________________________
> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>  _______________________________________________
> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>        ------------------------------
>
> _______________________________________________
> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>
>
> --
> jabber/gtalk: kristoff at krbonne.net
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091216/708f782b/attachment-0001.html 

From anthony.minessale at gmail.com  Wed Dec 16 12:13:11 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 16 Dec 2009 14:13:11 -0600
Subject: [Freeswitch-dev] coredump (dingaling)
In-Reply-To: <87f2f3b90912161204v652ac2a9s64d5d41f71b52bba@mail.gmail.com>
References: <4B289B4E.1070305@skypro.be>
	<6FBB10D9-76D4-4B44-AFA9-2CD86523BAAA@freeswitch.org>
	<191c3a030912160818g7443c131o78e252d0384f6890@mail.gmail.com>
	<4B293A89.2050802@skypro.be>
	<87f2f3b90912161204v652ac2a9s64d5d41f71b52bba@mail.gmail.com>
Message-ID: <191c3a030912161213y6d23e5f4j5d7ed3197e63e51f@mail.gmail.com>

I'm going to guess its the good old crappy gnutls crap.
noobuntu bleeding probably updated it and broke our support.
get the backtrace, my money is on something in gcrypt or gnutls


On Wed, Dec 16, 2009 at 2:04 PM, Michael Collins <msc at freeswitch.org> wrote:

> Well, now that you have demonstrated that you can repeat this on the latest
> trunk you can now do the gdb stuff on the core.xxxx file that gets created.
> Check out this page that has some information on gathering info:
>
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Advanced_Debugging_Techniques
>
> That page also has info on how to properly submit a JIRA, pastebin, etc.
> -MC
>
>
> On Wed, Dec 16, 2009 at 11:52 AM, Kristoff Bonne <kristoff.bonne at skypro.be
> > wrote:
>
>>  Hi all,
>>
>>
>> OK. Here is how far I have got sofar.
>>
>> - first: "make current" -> installed latest version
>>
>> - problem still there. However, not always. I do have had cases where
>> freeswitch did not crash, or crashed when the gtalk-session is terminated.
>>
>> - The "console loglvel 7" doesn't really help. Freeswitch bombs out before
>> having the change to print any debug-info.
>> (or is there any logfile I should look for?)
>>
>>
>> - I have run "tcpdump" on both sides (both on the server and a client) a
>> number of times.
>> -> The positive side is that -as I do have some sessions that did work- I
>> can compair with a "good" session.
>> -> The negative side is that it doesn't really help that much.
>>
>> The problem seams to be somewhere in, or and the end of the jabber
>> session. Looking at the "good" session, I see some traffic to the
>> stun-server. In the traces where the freeswitch crashes, I never see these
>> STUN-messages.
>> So it looks like the problem is somewhere (perhaps with the exchange of
>> IP-addresses for the ICE-session).
>>
>> However, I cannot see that for sure. The jabber-session with googletalk is
>> SSL-encrypted so I cannot see what is exactly in there.
>>
>>
>> BTW. My voip-client is "Empaty" on ubuntu 9.10
>>
>> Now, I did not additional test. I disabled IPv6 on my server and client (I
>> run a IPv6-enabled network at home). I tried with other clients (like the
>> voice-chat webapplication of google on my wive's mac; (both using firefox
>> and google chrome).
>> Same problem. Sometimes it works, but 3 out of 4 times it does not.
>>
>>
>>
>> Anybody any ideas on how to procede with this? Anybody any ideas of what
>> to try next?
>>
>> How does it come that I do not have any debug-information on the
>> freeswitch console?
>>
>>
>>
>>
>> Cheerio! Kr. Bonne.
>>
>>
>> Anthony Minessale schreef:
>>
>> also make sure its the SVN release because we dont take bug reports on
>> anything but latest trunk.
>>
>>
>> On Wed, Dec 16, 2009 at 8:54 AM, Brian West <brian at freeswitch.org> <brian at freeswitch.org> wrote:
>>
>>
>>
>>  Please see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>> The core file has NO use at all without running the gdb commands on
>> the same box that generated the core file... Uploading it is a futile
>> attempt.
>>
>> Please read the reporting bugs guide and it will tell you what steps
>> and commands to take to collect the info up and report the issue on
>> jira.
>>
>> Thanks,
>> Brian
>>
>> On Dec 16, 2009, at 2:33 AM, Kristoff Bonne wrote:
>>
>>
>>
>>  Hi All,
>>
>>
>> I don't know if this is the correct place for this, but I have a
>> problem
>> so perhaps somebody here can help.
>>
>> I've been playing around with freeswitch for about a week now using an
>> old mac-mini (ppc G3) running debian. Yesterday, I wanted to give
>> googletalk (dingaling) a try but got a coredump on freeswitch when I
>> dialed by gtalk account from outside.
>>
>>
>> Is there somewhere I can open a "ticket" for this? Can I upload the
>> corefile (about 1.2 MB compressed) somewhere?
>>
>>
>> This is on a PPC platform so I don't know if you people will have any
>> use of it.
>>
>>
>>
>> Cheerio! Kr. Bonne.
>>
>> --
>> jabber/gtalk: kristoff at krbonne.net
>>
>> <kristoff_bonne.vcf>_______________________________________________
>> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>>
>>  _______________________________________________
>> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>>
>>        ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>>
>>
>>
>> --
>> jabber/gtalk: kristoff at krbonne.net
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091216/8dd3a640/attachment.html 

From kristoff.bonne at skypro.be  Thu Dec 17 14:41:37 2009
From: kristoff.bonne at skypro.be (Kristoff Bonne)
Date: Thu, 17 Dec 2009 23:41:37 +0100
Subject: [Freeswitch-dev] coredump (dingaling)
In-Reply-To: <191c3a030912161213y6d23e5f4j5d7ed3197e63e51f@mail.gmail.com>
References: <4B289B4E.1070305@skypro.be>	<6FBB10D9-76D4-4B44-AFA9-2CD86523BAAA@freeswitch.org>	<191c3a030912160818g7443c131o78e252d0384f6890@mail.gmail.com>	<4B293A89.2050802@skypro.be>	<87f2f3b90912161204v652ac2a9s64d5d41f71b52bba@mail.gmail.com>
	<191c3a030912161213y6d23e5f4j5d7ed3197e63e51f@mail.gmail.com>
Message-ID: <4B2AB3A1.5000000@skypro.be>

Hi all,


Bug-rapport is submitted:
http://jira.freeswitch.org/browse/LBDING-18


This is a traceback I got when the problem accured when freeswitch
crashed when the gtalk-session was TERMINATED.

One of the things I did notice was this:
2009-12-17 11:42:45.461750 [ERR] switch_rtp.c:370 Invalid STUN/ICE
packet received
During some other voip-sessions, I got loads of these messages; however
I do not have this problem always.

One of the strange things I did notice is that in the packet captures I
took (on the server) I see UDP-packets for 69.0.208.27 udp/3478 (that's
stun) with incorrect CRC-checksum; or at least, that's what fireshark
tells me. The strange this is that I do get a UDP STUN-packets back
(which is strange as normally a server on the other side should ignore
UDP-packets with errors in it).


BTW.
Another elements this is strange:
Audio-traffic from the sip ATA to the mac flash/webinterface is almost
immediatly. Audio-traffic in the other side has about a 1 second delay
But this could be a problem on the webinterface flash-application, not
network related; so I do not know if this is related to this problem.



Cheerio! Kr. Bonne.



Anthony Minessale schreef:
> I'm going to guess its the good old crappy gnutls crap.
> noobuntu bleeding probably updated it and broke our support.
> get the backtrace, my money is on something in gcrypt or gnutls
>
>
> On Wed, Dec 16, 2009 at 2:04 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>   
>> Well, now that you have demonstrated that you can repeat this on the latest
>> trunk you can now do the gdb stuff on the core.xxxx file that gets created.
>> Check out this page that has some information on gathering info:
>>
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Advanced_Debugging_Techniques
>>
>> That page also has info on how to properly submit a JIRA, pastebin, etc.
>> -MC
>>
>>
>> On Wed, Dec 16, 2009 at 11:52 AM, Kristoff Bonne <kristoff.bonne at skypro.be
>>     
>>> wrote:
>>>       
>>>  Hi all,
>>>
>>>
>>> OK. Here is how far I have got sofar.
>>>
>>> - first: "make current" -> installed latest version
>>>
>>> - problem still there. However, not always. I do have had cases where
>>> freeswitch did not crash, or crashed when the gtalk-session is terminated.
>>>
>>> - The "console loglvel 7" doesn't really help. Freeswitch bombs out before
>>> having the change to print any debug-info.
>>> (or is there any logfile I should look for?)
>>>
>>>
>>> - I have run "tcpdump" on both sides (both on the server and a client) a
>>> number of times.
>>> -> The positive side is that -as I do have some sessions that did work- I
>>> can compair with a "good" session.
>>> -> The negative side is that it doesn't really help that much.
>>>
>>> The problem seams to be somewhere in, or and the end of the jabber
>>> session. Looking at the "good" session, I see some traffic to the
>>> stun-server. In the traces where the freeswitch crashes, I never see these
>>> STUN-messages.
>>> So it looks like the problem is somewhere (perhaps with the exchange of
>>> IP-addresses for the ICE-session).
>>>
>>> However, I cannot see that for sure. The jabber-session with googletalk is
>>> SSL-encrypted so I cannot see what is exactly in there.
>>>
>>>
>>> BTW. My voip-client is "Empaty" on ubuntu 9.10
>>>
>>> Now, I did not additional test. I disabled IPv6 on my server and client (I
>>> run a IPv6-enabled network at home). I tried with other clients (like the
>>> voice-chat webapplication of google on my wive's mac; (both using firefox
>>> and google chrome).
>>> Same problem. Sometimes it works, but 3 out of 4 times it does not.
>>>
>>>
>>>
>>> Anybody any ideas on how to procede with this? Anybody any ideas of what
>>> to try next?
>>>
>>> How does it come that I do not have any debug-information on the
>>> freeswitch console?
>>>
>>>
>>>
>>>
>>> Cheerio! Kr. Bonne.
>>>
>>>
>>> Anthony Minessale schreef:
>>>
>>> also make sure its the SVN release because we dont take bug reports on
>>> anything but latest trunk.
>>>
>>>
>>> On Wed, Dec 16, 2009 at 8:54 AM, Brian West <brian at freeswitch.org> <brian at freeswitch.org> wrote:
>>>
>>>
>>>
>>>  Please see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>
>>> The core file has NO use at all without running the gdb commands on
>>> the same box that generated the core file... Uploading it is a futile
>>> attempt.
>>>
>>> Please read the reporting bugs guide and it will tell you what steps
>>> and commands to take to collect the info up and report the issue on
>>> jira.
>>>
>>> Thanks,
>>> Brian
>>>
>>> On Dec 16, 2009, at 2:33 AM, Kristoff Bonne wrote:
>>>
>>>
>>>
>>>  Hi All,
>>>
>>>
>>> I don't know if this is the correct place for this, but I have a
>>> problem
>>> so perhaps somebody here can help.
>>>
>>> I've been playing around with freeswitch for about a week now using an
>>> old mac-mini (ppc G3) running debian. Yesterday, I wanted to give
>>> googletalk (dingaling) a try but got a coredump on freeswitch when I
>>> dialed by gtalk account from outside.
>>>
>>>
>>> Is there somewhere I can open a "ticket" for this? Can I upload the
>>> corefile (about 1.2 MB compressed) somewhere?
>>>
>>>
>>> This is on a PPC platform so I don't know if you people will have any
>>> use of it.
>>>
>>>
>>>
>>> Cheerio! Kr. Bonne.
>>>
>>> --
>>> jabber/gtalk: kristoff at krbonne.net
>>>
>>> <kristoff_bonne.vcf>_______________________________________________
>>> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>>>
>>>  _______________________________________________
>>> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>>>
>>>        ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> jabber/gtalk: kristoff at krbonne.net
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>>     
>
>
>   
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   


-- 
jabber/gtalk: kristoff at krbonne.net

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091217/0e0574fe/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kristoff_bonne.vcf
Type: text/x-vcard
Size: 190 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091217/0e0574fe/attachment-0001.vcf 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 261 bytes
Desc: OpenPGP digital signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091217/0e0574fe/attachment-0001.bin 

From ste.bossi at gmail.com  Fri Dec 18 02:38:33 2009
From: ste.bossi at gmail.com (stefano bossi)
Date: Fri, 18 Dec 2009 11:38:33 +0100
Subject: [Freeswitch-dev] thesis on live sip calls migration
In-Reply-To: <8ab521c30912171527l7008635dybb4a7ff4667eb952@mail.gmail.com>
References: <8ab521c30912171527l7008635dybb4a7ff4667eb952@mail.gmail.com>
Message-ID: <8ab521c30912180238j17c3ea09y4e996eab8213426f@mail.gmail.com>

Hi all,

as promised in #freeswitch-dev I'll try to explain better what I've done and
I would like to achieve for my thesis.


At the moment I successfully patched sofsip_cli and now it can do something
near active call migration :)
Thanks to the versatility of nua api I could achieve my objective with
little effort.
Basically I call someone, then I kill sofsip_cli and I restart it. I launch
my "reinvite" command and the call can continue normally. The called client
doesn't realize what happened.
In this case I write call data in a file and when I launch the reinvite I
read directly from this file.
Sofsip_cli sends an INVITE message with some TAG appended to the nua_handle.
So the called phone thinks to receive an in-session invite and simply
refreshes the call, but Sofsip_cli instead allocates all the stuff for a new
call. In this way I can avoid to modify directly the RTP part. It seems to
be all simpler.

I even wrote a little module for FS able to monitor(using the SIP OPTION
message) the life of a specific sofia profile.. It's just a proof of concept
but it can do failover(without live call migration) between 2 machines in
about 50ms.
This is possible with these actions:

   - registrations are shared with odbc


   - on the start of the module(present only on the backup machine):


   1. I set an arp rule blocking the arp response for the virtual IP (set on
   the primary machine)
   2. I set the virtual IP on the backup machine( but no one knows thanks to
   the arp rule and so I can bind to vIP)
   3. I prepare and run the sip profile (on the backup)... loading here the
   profile permits to save a lot of time during reaction


   - on the reaction


   1. I remove the arp rule
   2. I send a gratuitous arp request

As I said in about 50ms sip clients can call again. Maybe this time can
decrease using NETLINK socket for the arp table.

The union of these 2 works will give us a very fast "live profile migration"
:D

There some points to discuss:

   - switch_core_session_resurrect_channel: I didn't know this function, I
   need to understand what it offers, maybe tomorrow ;)
   - the propagation of call state variables: my first idea was to use the
   multicast events adding some headers to send all the necessary data (but
   anthm proposed XML)
   - I thought only to sofia aspects..

In next days I'll try to understand where to hook in FS to try these ideas.
When I'll have something ready (and without very very big errors:) I'll be
happy to send you the patch.
This is my first work on something real like FS.. I'm opened to any kind of
suggestions!*

*please contact me for further clarifications

Thanks

Stefano Bossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091218/b22c0cde/attachment.html 

From msc at freeswitch.org  Fri Dec 18 08:54:23 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 18 Dec 2009 08:54:23 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call Starting Shortly!
Message-ID: <87f2f3b90912180854h59b651c2t289f5a42ebec3973@mail.gmail.com>

Hello everyone!

Today's agenda is listed here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14

Also, we are going to be giving away goodies on some of the upcoming
conferences, so call in and see what we've got in store. :)

For the first 15 minutes we'll let everyone mingle and then we'll get into
the agenda.

Talk to you all soon!
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091218/df3e27df/attachment.html 

From rentmycoder at gmail.com  Sat Dec 19 09:09:00 2009
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Sat, 19 Dec 2009 18:09:00 +0100
Subject: [Freeswitch-dev] multiple actions in an extension with different
	condition executing each after possible???
Message-ID: <50e456910912190909y4a8bc11wd0aaf2d2e8a8b47@mail.gmail.com>

Hi pals,

I would like to convert the following dialplan int fs xml dialplan, is
this possible???

	;----- check for ivr vars -----
	exten => s,16,GotoIf($["${IvrID}" = ""]?17:18)
	exten => s,17,Set(IvrID=0)
	exten => s,18,GotoIf($["${IvrActionID}" = ""]?19:20)
	exten => s,19,Set(IvrActionID=0)
	exten => s,20,GotoIf($["${QueeID}" = ""]?21:22)
	exten => s,21,Set(QueeID=0)
	exten => s,22,GotoIf($["${ChannelID}" = ""]?23:24)
	exten => s,23,Set(ChannelID=${CHANNEL_INNER})
	exten => s,24,Set(CallStateID=${CALLSTATE_DIALING})

It would be easy if we could use nested conditions...

    <extension name="test">
      <condition field="destination_number" expression="^9999$">

      <condition field="${IvrID}" expression="^$">
        <action application="set" data="IvrID=1"/>
      </condition>

      <condition field="${QueeID}" expression="^$">
        <action application="set" data="QueeID=1"/>
      </condition>

      </condition>
    </extension>

But we can't:(

    <extension name="test">
      <condition field="destination_number" expression="^9999$" break="never"/>

      <condition field="${IvrID}" expression="^$">
        <action application="set" data="IvrID=1"/>
      </condition>

      <condition field="${QueeID}" expression="^$">
        <action application="set" data="QueeID=1"/>
      </condition>

    </extension>

I'we tried this trick also, but second and third condition get's never
executed...


As a workaround I could define different extensions and use transfer
to jump to the next
but I have a lot of thing to check resulting a lot of extensions...

There should be a more elegant way... isn't it?


From kristoff.bonne at skypro.be  Sat Dec 19 16:37:33 2009
From: kristoff.bonne at skypro.be (Kristoff Bonne)
Date: Sun, 20 Dec 2009 01:37:33 +0100
Subject: [Freeswitch-dev] iksemel
Message-ID: <4B2D71CD.9080009@skypro.be>

Hi all,


Can somebody please explain how exactly the "iksemel" library works.

I've been trying to hunt down the bug that causes freeswitch to crash
when making a jingle/gtalk call from ubuntu Empathy instead of the
google talk web/flash application.


I have narrowed the issue down to the difference in jingle-messages
call-setup messages send over the jabber-channel by empathy, compaired
to what the google webapplet sends.
(see http://jira.freeswitch.org/browse/LBDING-18 for more info on the
exact differences).



However, I'm trying to understand how exactly freeswitch processes these
messages.

So, if I get it correct, it goes like that:
- jabber xml-messages are treated by the iksemel library.
- this library is based on "filters", which are configured beforehands.
- when a jabber xml-message comes it, these filters are called. (va the
function "iks_parse").
- these filters call, in their turn, the function then call a number of
times the function "iks_find_attrib".

In this bug, the problem seams to be when a call-setup packet comes in
from empathy, on of these filters call the function "find_with_attrib",
however pointing to a datarecord which is not a valid iks-record,
resulting in a memory violation and hence a core-dump.


But can somebody explain how exactly the "iksemel" library works. There
seams to be a system of "hooks" used to attach pointers dynamically
created filters-functions. But how exactly it all fits together is not
very clear to me).

If I understand it well, the problem needs to be solved somewhere in the
code that creates these filters and define on what datarecord they
should be called. However, I got lost in the code.


Can somebody explain this a little bit.



Cheerio! Kr. Bonne.

-- 
jabber/gtalk: kristoff at krbonne.net

-------------- next part --------------
A non-text attachment was scrubbed...
Name: kristoff_bonne.vcf
Type: text/x-vcard
Size: 190 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091220/8dd3392b/attachment.vcf 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 261 bytes
Desc: OpenPGP digital signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091220/8dd3392b/attachment.bin 

From math.parent at gmail.com  Mon Dec 21 05:36:55 2009
From: math.parent at gmail.com (Mathieu Parent)
Date: Mon, 21 Dec 2009 14:36:55 +0100
Subject: [Freeswitch-dev] Implementation of Skinny Call Control Protocol
Message-ID: <960738410912210536ke82ea43lfafb2a9a3057099e@mail.gmail.com>

Hello,

I'm searching for a company or somebody to implement an SCCP (aka
Skinny) endpoints module in FreeSWITCH (directly or via OPAL). The
module should support Cisco IP Phones including Cisco IP Phone 7905,
Cisco IP Phone 7920 and Cisco IP Phone 7960G.

NB: I am not on freeswitch-biz list.

Regards

Mathieu Parent


From sky.side at hotmail.com  Sun Dec 20 02:36:41 2009
From: sky.side at hotmail.com (sky tan)
Date: Sun, 20 Dec 2009 10:36:41 +0000
Subject: [Freeswitch-dev] =?iso-2022-jp?b?SGVscCAbJEIhKhsoQiBIb3cgdG8gYWRq?=
 =?iso-2022-jp?b?dXN0IHRoZSBzcGVleCBDb2RlYyAncyBwYXJhbWV0ZXIgZm9yIG15IHNp?=
 =?iso-2022-jp?b?cCBwaG9uZQ==?=
Message-ID: <SNT125-W156FE35799BF09AAA4ECFA9B830@phx.gbl>


hi, all

    I used my soft phone with speex phone in speex, 8000hz , my softphone ' codec is base on opal lib.
but when I register it in freeswitch and call in a conference . I fine the voice quality is not good enough?

   I did print the parameter in switch_speex_decode() function , I find that my softphone sent  encoded_data_len=21.  if I send frame as encoded_data_len=20, then the voice quality become nice.

   So ,who can tell me how to change the freeswitch's speex parameter, then I can operate the encoded_data_len=21  ?


best regards .
your freeswitch' fan.

 		 	   		  
_________________________________________________________________
Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091220/044b4928/attachment.html 

From brian at freeswitch.org  Mon Dec 21 07:03:02 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 21 Dec 2009 09:03:02 -0600
Subject: [Freeswitch-dev] Implementation of Skinny Call Control Protocol
In-Reply-To: <960738410912210536ke82ea43lfafb2a9a3057099e@mail.gmail.com>
References: <960738410912210536ke82ea43lfafb2a9a3057099e@mail.gmail.com>
Message-ID: <30E12B1F-AAB5-49AA-958F-74CE77E69306@freeswitch.org>

Email consulting at freeswitch.org we can find someone to do this.

/b

On Dec 21, 2009, at 7:36 AM, Mathieu Parent wrote:

> Hello,
> 
> I'm searching for a company or somebody to implement an SCCP (aka
> Skinny) endpoints module in FreeSWITCH (directly or via OPAL). The
> module should support Cisco IP Phones including Cisco IP Phone 7905,
> Cisco IP Phone 7920 and Cisco IP Phone 7960G.
> 
> NB: I am not on freeswitch-biz list.
> 
> Regards
> 
> Mathieu Parent



From math.parent at gmail.com  Mon Dec 21 07:08:27 2009
From: math.parent at gmail.com (Mathieu Parent)
Date: Mon, 21 Dec 2009 16:08:27 +0100
Subject: [Freeswitch-dev] [Freeswitch-biz] Fwd: Implementation of Skinny
	Call Control Protocol
In-Reply-To: <d3ceea380912210656y3ce609b1t7723ef059bbd2295@mail.gmail.com>
References: <960738410912210536ke82ea43lfafb2a9a3057099e@mail.gmail.com>
	<960738410912210542x31f353c4h820eb7fca358723@mail.gmail.com>
	<d3ceea380912210656y3ce609b1t7723ef059bbd2295@mail.gmail.com>
Message-ID: <960738410912210708t77bf8d7dh30540c803fbc4e63@mail.gmail.com>

Hello,

On Mon, Dec 21, 2009 at 3:56 PM, Itamar Reis Peixoto
<itamar at ispbrasil.com.br> wrote:
> On Mon, Dec 21, 2009 at 11:42 AM, Mathieu Parent <math.parent at gmail.com> wrote:
>> Hello,
>>
>> I'm searching for a company or somebody to implement an SCCP (aka
>> Skinny) endpoints module in FreeSWITCH (directly or via OPAL). The
>> module should support Cisco IP Phones including Cisco IP Phone 7905,
>> Cisco IP Phone 7920 and Cisco IP Phone 7960G.
>>
>> Regards
>>
>> Mathieu Parent
>>
>
> why you don't flash your phones with the sip firmware ?

because:
- no french translation (english only)
- no way to configure SpeedDial Buttons (remotely)
- no way to configure SoftKeys (remotely)
- according to <http://www.voip-info.org/wiki/view/Setup+SiP+on+7940+-+7960>:
"the firmware runs into intermittent timeouts upon pulling URLs"

Regards

Mathieu Parent


From bossekr at debian.org  Mon Dec 21 11:54:56 2009
From: bossekr at debian.org (bossekr at debian.org)
Date: Mon, 21 Dec 2009 20:54:56 +0100 (CET)
Subject: [Freeswitch-dev] Changes for files within freeswitch/libs/*
	compared with official version
Message-ID: <g3ho1vv7exoze5inj5UYAxe124vaj_firegpg@mail.gmail.com>

Hi developers,

I'm new to FreeSWITCH and would like to know if changes on the freeswitch/libs/* files are done ? I would like to know if it makes sense to use the original sources while prepare a FreeSWITCH packages for e.g. Debian distribution where most of these libraries exists (for some with an different version).
From quality point of view it could be a bad idea to do so.

Greetings,
Raphael

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 272 bytes
Desc: OpenPGP digital signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091221/57a0c716/attachment-0001.bin 

From mike at jerris.com  Mon Dec 21 12:11:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 21 Dec 2009 15:11:51 -0500
Subject: [Freeswitch-dev] Changes for files within freeswitch/libs/*
	compared with official version
In-Reply-To: <g3ho1vv7exoze5inj5UYAxe124vaj_firegpg@mail.gmail.com>
References: <g3ho1vv7exoze5inj5UYAxe124vaj_firegpg@mail.gmail.com>
Message-ID: <5F7E6BEB-1AF9-461E-94D1-649740779AA0@jerris.com>

Yes, we have modification, no, you can not use distro libs to replace any of our in tree libs, with the exception of libcurl, which there is a configure flag to specify.  Please see the debian directory in tree for a mostly working deb package that is in process of being improved.

Mike

On Dec 21, 2009, at 2:54 PM, bossekr at debian.org wrote:

> Hi developers,
> 
> I'm new to FreeSWITCH and would like to know if changes on the freeswitch/libs/* files are done ? I would like to know if it makes sense to use the original sources while prepare a FreeSWITCH packages for e.g. Debian distribution where most of these libraries exists (for some with an different version).
> From quality point of view it could be a bad idea to do so.
> 
> Greetings,
> Raphael



From quentusrex at gmail.com  Mon Dec 21 12:39:30 2009
From: quentusrex at gmail.com (William King)
Date: Mon, 21 Dec 2009 12:39:30 -0800
Subject: [Freeswitch-dev] Changes for files within freeswitch/libs/*
 compared with official version
In-Reply-To: <g3ho1vv7exoze5inj5UYAxe124vaj_firegpg@mail.gmail.com>
References: <g3ho1vv7exoze5inj5UYAxe124vaj_firegpg@mail.gmail.com>
Message-ID: <1261427970.2265.1.camel@quentusrex-asus>

Raphael, 

There are a few of us working on getting the debian/ubuntu package of
freeswitch debian ready. Check what is in tree in the debian directory
for the WIP of the package.

-William King

On Mon, 2009-12-21 at 20:54 +0100, bossekr at debian.org wrote:
> Hi developers,
> 
> I'm new to FreeSWITCH and would like to know if changes on the freeswitch/libs/* files are done ? I would like to know if it makes sense to use the original sources while prepare a FreeSWITCH packages for e.g. Debian distribution where most of these libraries exists (for some with an different version).
> From quality point of view it could be a bad idea to do so.
> 
> Greetings,
> Raphael
> 
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org




From sky.side at hotmail.com  Mon Dec 21 18:09:43 2009
From: sky.side at hotmail.com (sky tan)
Date: Tue, 22 Dec 2009 02:09:43 +0000
Subject: [Freeswitch-dev]
 =?iso-2022-jp?b?SGVscCAbJEIhKhsoQiBIb3cgdG8gYWRq?=
 =?iso-2022-jp?b?dXN0IHRoZSBmcmVlc3dpdGNoIHNwZWV4IENvZGVjICdzIHBhcmFtZXRl?=
 =?iso-2022-jp?b?ciBmb3IgbXkgc2lwIHBob25l?=
In-Reply-To: <mailman.4246.1261425407.2495.freeswitch-dev@lists.freeswitch.org>
References: <mailman.4246.1261425407.2495.freeswitch-dev@lists.freeswitch.org>
Message-ID: <SNT125-W4892394ABD92012330547A9B810@phx.gbl>


So I  explain my question as:
  if I use x-lite softphone in speex codec(it must be narrowband), the vioce quality is pool .
 but when I use X-lite in speex wideband codec , the voice quality is very nice.
I do try to change the parameter of default_codec_settings in mod_speex.c , but it seen don't
affect anything.
  so who can tell me how to ajust the freeswitch side to fit the X-lite softphone in speex narrowband?

best regards .
your freeswitch' fan.



--Forwarded Message Attachment--
From: sky.side at hotmail.com
To: freeswitch-dev at lists.freeswitch.org
Date: Sun, 20 Dec 2009 10:36:41 +0000
Subject: [Freeswitch-dev] Help ? How to adjust the speex Codec 's parameter for my sip phone

hi, all

    I used my soft phone with codec in speex, 8000hz , my softphone ' codec is base on opal lib.
but when I register it in freeswitch and call in a conference . I fine the voice quality is not good enough?

   I did print the parameter in switch_speex_decode() function , I find that my softphone sent  encoded_data_len=21.  if I send frame as encoded_data_len=20, then the voice quality become nice.

   So ,who can tell me how to change the freeswitch's speex parameter, then I can operate the encoded_data_len=21  ?


best regards .
your freeswitch' fan.

 		 	   		  
 		 	   		  
_________________________________________________________________
Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091222/e8cf0a8d/attachment.html 

From anthony.minessale at gmail.com  Tue Dec 22 07:20:19 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 22 Dec 2009 09:20:19 -0600
Subject: [Freeswitch-dev]
	=?utf-8?q?Help_=EF=BC=81_How_to_adjust_the_frees?=
	=?utf-8?q?witch_speex_Codec_=27s_parameter_for_my_sip_phone?=
In-Reply-To: <SNT125-W4892394ABD92012330547A9B810@phx.gbl>
References: <mailman.4246.1261425407.2495.freeswitch-dev@lists.freeswitch.org>
	<SNT125-W4892394ABD92012330547A9B810@phx.gbl>
Message-ID: <191c3a030912220720i64295864s4139db863aa9fa32@mail.gmail.com>

add speex at 16000h instead of just speex to the codec list.


2009/12/21 sky tan <sky.side at hotmail.com>

>  So I  explain my question as:
>   if I use x-lite softphone in speex codec(it must be narrowband), the
> vioce quality is pool .
>  but when I use X-lite in speex wideband codec , the voice quality is very
> nice.
> I do try to change the parameter of default_codec_settings in mod_speex.c ,
> but it seen don't
> affect anything.
>   so who can tell me how to ajust the freeswitch side to fit the X-lite
> softphone in speex narrowband?
>
> best regards .
> your freeswitch' fan.
>
>
>
> --Forwarded Message Attachment--
> From: sky.side at hotmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Date: Sun, 20 Dec 2009 10:36:41 +0000
> Subject: [Freeswitch-dev] Help ? How to adjust the speex Codec 's parameter
> for my sip phone
>
> hi, all
>
>     I used my soft phone with codec in speex, 8000hz , my softphone ' codec
> is base on opal lib.
> but when I register it in freeswitch and call in a conference . I fine the
> voice quality is not good enough?
>
>    I did print the parameter in switch_speex_decode() function , I find
> that my softphone sent  encoded_data_len=21.  if I send frame as
> encoded_data_len=20, then the voice quality become nice.
>
>    So ,who can tell me how to change the freeswitch's speex parameter, then
> I can operate the encoded_data_len=21  ?
>
>
> best regards .
> your freeswitch' fan.
>
>
>
> ------------------------------
> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they
> e-mail you.<http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091222/ee3dad04/attachment.html 

From bkarvandi at yahoo.com  Tue Dec 22 09:26:42 2009
From: bkarvandi at yahoo.com (Babak Karvandi)
Date: Tue, 22 Dec 2009 09:26:42 -0800 (PST)
Subject: [Freeswitch-dev] "a1-has" param in gateway setting
Message-ID: <179369.24879.qm@web110203.mail.gq1.yahoo.com>

Hi,

Does any body know or has test the "a1-hash" parameter with gateway
setting?  I am not sure if it is even allowed. I have the following
gateway setting but when the freeswitch starts up it simply ignores this
provider without any error message or attempt to register in the log
file.  Thank you for your help in advance.

<include>
  <gateway name="iptel">
    <param name="username" value="MY-USERNAME"/>
    <param name="realm" value="iptel.org"/>
    <!-- <param name="password" value="MY_PASSWORD"/>  -->

    <!-- replaced the password with MD5 encrypted -->
    <!-- openssl dgst -md5 < filename, or echo
"username:domain:password" | openssldgst -md5 -->

    <param name="a1-hash" value="30f610a85e973f2b29b75ddc1ec3450e"/>
    <param name="proxy" value="sip.iptel.org"/>
  </gateway>
</include>


From brian at freeswitch.org  Tue Dec 22 10:04:30 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 22 Dec 2009 12:04:30 -0600
Subject: [Freeswitch-dev] "a1-has" param in gateway setting
In-Reply-To: <179369.24879.qm@web110203.mail.gq1.yahoo.com>
References: <179369.24879.qm@web110203.mail.gq1.yahoo.com>
Message-ID: <ABB0946C-5FC2-4C81-9588-25BC70323965@freeswitch.org>

I'm not too sure you can put an a1-hash on outbound auth.

/b

On Dec 22, 2009, at 11:26 AM, Babak Karvandi wrote:

> Hi,
> 
> Does any body know or has test the "a1-hash" parameter with gateway
> setting?  I am not sure if it is even allowed. I have the following
> gateway setting but when the freeswitch starts up it simply ignores this
> provider without any error message or attempt to register in the log
> file.  Thank you for your help in advance.
> 
> <include>
>  <gateway name="iptel">
>    <param name="username" value="MY-USERNAME"/>
>    <param name="realm" value="iptel.org"/>
>    <!-- <param name="password" value="MY_PASSWORD"/>  -->
> 
>    <!-- replaced the password with MD5 encrypted -->
>    <!-- openssl dgst -md5 < filename, or echo
> "username:domain:password" | openssldgst -md5 -->
> 
>    <param name="a1-hash" value="30f610a85e973f2b29b75ddc1ec3450e"/>
>    <param name="proxy" value="sip.iptel.org"/>
>  </gateway>
> </include>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091222/1f3a4443/attachment.html 

From msc at freeswitch.org  Tue Dec 22 14:17:23 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 22 Dec 2009 14:17:23 -0800
Subject: [Freeswitch-dev] FreeSWITCH 1.0.5pre10 is now available
Message-ID: <87f2f3b90912221417l3b4b7f00pe7a4c7775ce2d85a@mail.gmail.com>

It's upgrade-and-test time! The new release announcement is on the main
FreeSWITCH page:
http://www.freeswitch.org/node/224

Please update, test, and report back bugs and questions.

Thanks!

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091222/118432aa/attachment.html 

From pvolchek at voicemobility.com  Tue Dec 22 15:31:16 2009
From: pvolchek at voicemobility.com (Peter Volchek)
Date: Tue, 22 Dec 2009 15:31:16 -0800
Subject: [Freeswitch-dev] Silence detection and reporting
In-Reply-To: <87f2f3b90912221417l3b4b7f00pe7a4c7775ce2d85a@mail.gmail.com>
References: <87f2f3b90912221417l3b4b7f00pe7a4c7775ce2d85a@mail.gmail.com>
Message-ID: <4B3156C4.4010400@voicemobility.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091222/f89dd32e/attachment-0001.html 

From sky.side at hotmail.com  Tue Dec 22 19:17:00 2009
From: sky.side at hotmail.com (sky tan)
Date: Wed, 23 Dec 2009 03:17:00 +0000
Subject: [Freeswitch-dev]
 =?utf-8?q?Help_=EF=BC=81_How_to_adjust_the_frees?=
 =?utf-8?q?witch_speex_Codec_=27s_parameter_for_my_sip_phone?=
In-Reply-To: <mailman.4298.1261524711.2495.freeswitch-dev@lists.freeswitch.org>
References: <mailman.4298.1261524711.2495.freeswitch-dev@lists.freeswitch.org>
Message-ID: <SNT125-W39098E96C40EDE5B1227E59B800@phx.gbl>


hi,Minessale 

   thank for your aswer, I show my setting below:
my previous setting in vars.xml is?
 <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,speex at 8000h@20i,speex at 16000h@20i,speex at 32000h@20i,G7221 at 16000h,G722,PCMU,PCMA,iLBC at 30i,GSM"/>

then I change my setting as:
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,speex at 8000h,speex at 16000h,speex at 32000h,G7221 at 16000h,G722,PCMU,PCMA,iLBC at 30i,GSM"/>

the problem still exist. the appearance is : the x-lite (8000h) calling side sound good, and the x-lite called side's voice quality is very poor.

also I test many sip softphone, and get my analyze as:
the media consultation in SDP(Session Discription Protocol) is not work correctly in speex at 8000.
 
can you get me more suggestion?
or any one who have such experience in it ?

thanks a lot !



--Forwarded Message Attachment--
From: anthony.minessale at gmail.com
To: freeswitch-dev at lists.freeswitch.org
Date: Tue, 22 Dec 2009 09:20:19 -0600
Subject: Re: [Freeswitch-dev]	Help ? How to adjust the freeswitch speex Codec 's parameter for my sip phone

add speex at 16000h instead of just speex to the codec list.


2009/12/21 sky tan <sky.side at hotmail.com>






So I  explain my question as:
  if I use x-lite softphone in speex codec(it must be narrowband), the vioce quality is pool .
 but when I use X-lite in speex wideband codec , the voice quality is very nice.
I do try to change the parameter of default_codec_settings in mod_speex.c , but it seen don't

affect anything.
  so who can tell me how to ajust the freeswitch side to fit the X-lite softphone in speex narrowband?

best regards .
your freeswitch' fan.


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900






 		 	   		  
_________________________________________________________________
Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail?.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091223/9375a671/attachment.html 

From juanbackson at gmail.com  Wed Dec 23 04:56:46 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Wed, 23 Dec 2009 20:56:46 +0800
Subject: [Freeswitch-dev] error with referencing channel->direction
Message-ID: <27c25bc40912230456x2fd968cav74ab077500e02591@mail.gmail.com>

Hi,

The following lines give me some strange error:


    switch_channel_t *channel = switch_core_session_get_channel(session);





        if (channel->direction==SWITCH_CALL_DIRECTION_OUTBOUND) {


error: dereferencing pointer to incomplete type
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091223/43cc6a7a/attachment.html 

From mrene_lists at avgs.ca  Wed Dec 23 06:22:54 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 23 Dec 2009 09:22:54 -0500
Subject: [Freeswitch-dev] error with referencing channel->direction
In-Reply-To: <27c25bc40912230456x2fd968cav74ab077500e02591@mail.gmail.com>
References: <27c25bc40912230456x2fd968cav74ab077500e02591@mail.gmail.com>
Message-ID: <DB2807EB-DF07-4933-B5C4-15ED3046536B@avgs.ca>

Hi,

You may not dereference switch_channel_t or switch_session_t. They are  
opaque structure to force people to use wrapper functions in  
switch_channel.h and in switch_core_session.h.

The function you are looking for is:
SWITCH_DECLARE(switch_call_direction_t)  
switch_channel_direction(switch_channel_t *channel);

Cheers,

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 23-Dec-09, at 7:56 AM, Juan Backson wrote:

> Hi,
>
> The following lines give me some strange error:
>
>
>     switch_channel_t *channel =  
> switch_core_session_get_channel(session);
>
>
>
>
>
>         if (channel->direction==SWITCH_CALL_DIRECTION_OUTBOUND) {
>
>
> error: dereferencing pointer to incomplete type
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From msc at freeswitch.org  Wed Dec 23 14:00:01 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 23 Dec 2009 14:00:01 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call - Holiday
	Schedule
Message-ID: <87f2f3b90912231400q65fefa87g778d1bf4857db142@mail.gmail.com>

Hello all!

Because the holidays fall on consecutive Fridays this year we decided to
have a single conference call on Wednesday Dec 30th at the usual time of
11AM CST. The agenda is posted here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30

Thanks for supporting the weekly calls. Don't forget that we will soon be
having giveaways and fun stuff on the calls so be sure to plan on joining
us!

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091223/5c47f7e1/attachment.html 

From lei.tlfly at gmail.com  Thu Dec 24 03:06:41 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Thu, 24 Dec 2009 19:06:41 +0800
Subject: [Freeswitch-dev] FS doesn't update rtp port when sdp changed in
	200OK response for invite message.
Message-ID: <50c41b4e0912240306h61b09d43rf68d4d5b28b4e219@mail.gmail.com>

Merry Christmas everyone!

I'm using FS 1.0.5pre9, I think there is a bug in mod_sofia when SDP changed
in 200 OK response for INVITE message,

the scenario is :
A(FreeSwitch)                      B
      ------INVITE ------->
      <----100 Tring--------
      <----180 Ring --------   with sdp m=audio 55066 RTP/AVP 0 120  c=IN
IP4 10.36.143.76
      <----UPDATE -------    with sdp m=audio 45486 RTP/AVP 0 120  c=IN IP4
10.36.143.76
      -----200 OK  ------>     response for UPDATE message
      <---- 200 OK--------     response for INVITE message,  with sdp
m=audio 45486 RTP/AVP 0 120  c=IN IP4 10.36.143.76
      --------ACK --------->

The problem is, B changed the rtp port in UPDATE message and "200 OK"
response message, but FS didn't do update, so it still send and receive data
from port 55066.
I read the mod_sofia source code and debug log, I found that  when get 200OK
response ,mod_sofia should update rtp session when sdp information is
different of the the pre sdp received in 180 ring response,
I'm using proxy_media mode, and I think there is the same problem in normal
mode.

here is the outbound channel state change log
==============================
entering state [received][100]
entering state [calling][0]
entering state [proceeding][180]
entering state [proceeding][200]
entering state [completing][200]
entering state [ready][200]
entering state [completed][200]
entering state [ready][200]


The problem is in state [proceeding][200], The code is as follow:
sofia.c line 3876
                if (switch_channel_test_flag(channel, CF_PROXY_MEDIA)) {
                    if (sofia_glue_activate_rtp(tech_pvt, 0) !=
SWITCH_STATUS_SUCCESS) {
                        goto done;
                    }
                }

Since the rtp session is initialized when 180 RING response received,
sofia_glue_activate_rtp will return SWITCH_STATUS_SUCCESS without change any
thing,
So, I think the code should be as follow

3876                 if (switch_channel_test_flag(channel, CF_PROXY_MEDIA))
{
3877                     if (tech_pvt->rtp_session &&
!is_dup_sdp){

3878                         sofia_set_flag_locked(tech_pvt,
TFLAG_REINVITE);
3879                         if (sofia_glue_tech_proxy_remote_addr(tech_pvt)
!= SWITCH_STATUS_SUCCESS || sofia_glue_activate_rtp(tec     h_pvt, 0) !=
SWITCH_STATUS_SUCCESS)
{

3880                             sofia_clear_flag_locked(tech_pvt,
TFLAG_REINVITE);
3881                             goto
done;

3882
}

3883                         sofia_clear_flag_locked(tech_pvt,
TFLAG_REINVITE);
3884
}else{

3885                         if (sofia_glue_activate_rtp(tech_pvt, 0) !=
SWITCH_STATUS_SUCCESS) {
3886                             goto
done;

3887
}

3888
}

3889                 }

When get 200OK reponse and sdp is changed, we should update remote port and
set reinvite flag before call  sofia_glue_activate_rtp.

Because I'm not so familiy with sofia and mod_sofia's source code, this code
is only for PROXY MEDIA mode and doesn't check for codec changed.(in my
scenario, only the remote port and ip are changed).

Could some fs dev check this problem, and give a offical patch?

BTW, I guess maybe mod_sofia is just ignore UPDATE message too. (just ignore
it if I was wrong)

Best Regards
-- 
Lei.Tang
lei.tlfly at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091224/54d5911c/attachment-0001.html 

From msc at freeswitch.org  Thu Dec 24 11:19:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 24 Dec 2009 11:19:51 -0800
Subject: [Freeswitch-dev] FS doesn't update rtp port when sdp changed in
	200OK response for invite message.
In-Reply-To: <50c41b4e0912240306h61b09d43rf68d4d5b28b4e219@mail.gmail.com>
References: <50c41b4e0912240306h61b09d43rf68d4d5b28b4e219@mail.gmail.com>
Message-ID: <87f2f3b90912241119t26fb6061p4b853576c7762841@mail.gmail.com>

Can you update to latest trunk and see if this still occurs? There have been
many updates since pre9.
Thanks,
MC

2009/12/24 Lei Tang <lei.tlfly at gmail.com>

> Merry Christmas everyone!
>
> I'm using FS 1.0.5pre9, I think there is a bug in mod_sofia when SDP
> changed in 200 OK response for INVITE message,
>
> the scenario is :
> A(FreeSwitch)                      B
>       ------INVITE ------->
>       <----100 Tring--------
>       <----180 Ring --------   with sdp m=audio 55066 RTP/AVP 0 120  c=IN
> IP4 10.36.143.76
>       <----UPDATE -------    with sdp m=audio 45486 RTP/AVP 0 120  c=IN IP4
> 10.36.143.76
>       -----200 OK  ------>     response for UPDATE message
>       <---- 200 OK--------     response for INVITE message,  with sdp
> m=audio 45486 RTP/AVP 0 120  c=IN IP4 10.36.143.76
>       --------ACK --------->
>
> The problem is, B changed the rtp port in UPDATE message and "200 OK"
> response message, but FS didn't do update, so it still send and receive data
> from port 55066.
> I read the mod_sofia source code and debug log, I found that  when get
> 200OK response ,mod_sofia should update rtp session when sdp information is
> different of the the pre sdp received in 180 ring response,
> I'm using proxy_media mode, and I think there is the same problem in normal
> mode.
>
> here is the outbound channel state change log
> ==============================
> entering state [received][100]
> entering state [calling][0]
> entering state [proceeding][180]
> entering state [proceeding][200]
> entering state [completing][200]
> entering state [ready][200]
> entering state [completed][200]
> entering state [ready][200]
>
>
> The problem is in state [proceeding][200], The code is as follow:
> sofia.c line 3876
>                 if (switch_channel_test_flag(channel, CF_PROXY_MEDIA)) {
>                     if (sofia_glue_activate_rtp(tech_pvt, 0) !=
> SWITCH_STATUS_SUCCESS) {
>                         goto done;
>                     }
>                 }
>
> Since the rtp session is initialized when 180 RING response received,
> sofia_glue_activate_rtp will return SWITCH_STATUS_SUCCESS without change any
> thing,
> So, I think the code should be as follow
>
> 3876                 if (switch_channel_test_flag(channel, CF_PROXY_MEDIA))
> {
> 3877                     if (tech_pvt->rtp_session &&
> !is_dup_sdp){
>
> 3878                         sofia_set_flag_locked(tech_pvt,
> TFLAG_REINVITE);
> 3879                         if
> (sofia_glue_tech_proxy_remote_addr(tech_pvt) != SWITCH_STATUS_SUCCESS ||
> sofia_glue_activate_rtp(tec     h_pvt, 0) != SWITCH_STATUS_SUCCESS)
> {
>
> 3880                             sofia_clear_flag_locked(tech_pvt,
> TFLAG_REINVITE);
> 3881                             goto
> done;
>
> 3882
> }
>
> 3883                         sofia_clear_flag_locked(tech_pvt,
> TFLAG_REINVITE);
> 3884
> }else{
>
> 3885                         if (sofia_glue_activate_rtp(tech_pvt, 0) !=
> SWITCH_STATUS_SUCCESS) {
> 3886                             goto
> done;
>
> 3887
> }
>
> 3888
> }
>
> 3889                 }
>
> When get 200OK reponse and sdp is changed, we should update remote port and
> set reinvite flag before call  sofia_glue_activate_rtp.
>
> Because I'm not so familiy with sofia and mod_sofia's source code, this
> code is only for PROXY MEDIA mode and doesn't check for codec changed.(in my
> scenario, only the remote port and ip are changed).
>
> Could some fs dev check this problem, and give a offical patch?
>
> BTW, I guess maybe mod_sofia is just ignore UPDATE message too. (just
> ignore it if I was wrong)
>
> Best Regards
> --
> Lei.Tang
> lei.tlfly at gmail.com
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091224/26a1674e/attachment.html 

From paulo.pizarro at gmail.com  Thu Dec 24 06:29:52 2009
From: paulo.pizarro at gmail.com (Paulo Pizarro)
Date: Thu, 24 Dec 2009 12:29:52 -0200
Subject: [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip library
	(off-topic)
Message-ID: <8a384d790912240629q77a367eeg8f048fb3de232943@mail.gmail.com>

Hi all,

We like to present to all Sofia SIP developers a Lua bind to Sofia SIP,
Luasofia. The objective of this project is to be able to use Sofia on a more
high level language, Lua was chosen because of its easy integration with C,
a lot of our projects are starting to use Lua integrated with C to make
development and maintenance easier and faster, because of that we wanted to
have a good SIP library to use in Lua, the decision was to make a bind to
the already mature Sofia library. The bind is far from being complete, we
only bonded the code that we need right now, there's a lot of work to do. We
expect that the project will be useful, so a lot of documentation was made
to help someone who wants to bind more Sofia code. There is good
documentation on how to use it too.

To develop we used git to make version control, and we hosted the project at
github:

http://github.com/ppizarro/luasofia

The wiki is at github too:

http://wiki.github.com/ppizarro/luasofia

We hope Luasofia can be useful to a lot of people, and eventually we get
some help making it better and more complete.

Best regards,

Luasofia team
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091224/ddfd50ac/attachment.html 

From davis.erwin at gmail.com  Fri Dec 25 12:54:57 2009
From: davis.erwin at gmail.com (Erwin Davis)
Date: Fri, 25 Dec 2009 12:54:57 -0800
Subject: [Freeswitch-dev] Generic Question on switch_rtp.c and STFU
Message-ID: <a38a91d50912251254h7c77a0ffi23671f1328265b42@mail.gmail.com>

Hi, I have a question on the time period when stfu_read_a_frame(), which
reads a frame from a jiffer buffer within an interval T1,  is called.
Ideally the frames in the jitter buffer are transmitted within interval T1
to avoid the jitter.  However, if I understand the code correctly, the
interval is determined by the latency between two read_pollfd triggering
instead of T1. The reason is that, at switch_rtp.c, Proc stfu_read_a_frame()
is called by read_rtp_packet() which is called when  read_pollfd is
triggered. However, If I understand correctly, read_pollfd is triggeredd
only when there is something on the sockets. For example, if the latency of
two read_pollfd triggerings is 50 ms, the actual transmit of next new frame
from the jitter buffer should be 50 ms delay instead of T1. Please correct
me if I am missing something. Thanks,

Regards,

e
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091225/03683d72/attachment.html 

From 2012.miracle at gmail.com  Sun Dec 27 23:21:49 2009
From: 2012.miracle at gmail.com (miracle 2012)
Date: Mon, 28 Dec 2009 12:51:49 +0530
Subject: [Freeswitch-dev] pls help
Message-ID: <df38f5d30912272321l5fc3b324v34b66bf221e81df3@mail.gmail.com>

Hi,

please provide any example c programs , APIs info, reference as i want to
write a dialer in c with freeswitch.
i am a newbie for free switch... :|

thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091228/8d2538fe/attachment.html 

From brian at freeswitch.org  Mon Dec 28 06:44:36 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 28 Dec 2009 08:44:36 -0600
Subject: [Freeswitch-dev] FS doesn't update rtp port when sdp changed in
	200OK response for invite message.
In-Reply-To: <87f2f3b90912241119t26fb6061p4b853576c7762841@mail.gmail.com>
References: <50c41b4e0912240306h61b09d43rf68d4d5b28b4e219@mail.gmail.com>
	<87f2f3b90912241119t26fb6061p4b853576c7762841@mail.gmail.com>
Message-ID: <FBA0B35D-49C6-4B60-8529-F0622B6FAC74@freeswitch.org>

Is it even legal to change the port like that?  I recall a discussion and change to the code to allow this but as far as I can recall its not legal to do that... I may be wrong.

/b

On Dec 24, 2009, at 1:19 PM, Michael Collins wrote:

> Can you update to latest trunk and see if this still occurs? There have been many updates since pre9.
> Thanks,
> MC



From lei.tlfly at gmail.com  Mon Dec 28 07:09:32 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Mon, 28 Dec 2009 23:09:32 +0800
Subject: [Freeswitch-dev] FS doesn't update rtp port when sdp changed in
	200OK response for invite message.
In-Reply-To: <FBA0B35D-49C6-4B60-8529-F0622B6FAC74@freeswitch.org>
References: <50c41b4e0912240306h61b09d43rf68d4d5b28b4e219@mail.gmail.com>
	<87f2f3b90912241119t26fb6061p4b853576c7762841@mail.gmail.com>
	<FBA0B35D-49C6-4B60-8529-F0622B6FAC74@freeswitch.org>
Message-ID: <50c41b4e0912280709n172d6617x55b94f7f49d540a9@mail.gmail.com>

Hi Brian, In my scenario, I connect FS to a  softswitch(bell alcatel),  the
port change because the callee is using *CRBT(*Coloring Ring Back Tone*), I
don't known much detail of the softswitch, I guessthere is a server process
CRBT  and  another process **audio stream, so the port is changed, when I
call a number without CRBT, the port is not changed.**   *

2009/12/28 Brian West <brian at freeswitch.org>

> Is it even legal to change the port like that?  I recall a discussion and
> change to the code to allow this but as far as I can recall its not legal to
> do that... I may be wrong.
>
> /b
>
> On Dec 24, 2009, at 1:19 PM, Michael Collins wrote:
>
> > Can you update to latest trunk and see if this still occurs? There have
> been many updates since pre9.
> > Thanks,
> > MC
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Lei.Tang
lei.tlfly at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091228/9f3d6e8d/attachment.html 

From tonhudung at gmail.com  Mon Dec 28 07:53:52 2009
From: tonhudung at gmail.com (Alex To)
Date: Mon, 28 Dec 2009 23:53:52 +0800
Subject: [Freeswitch-dev] pls help
In-Reply-To: <df38f5d30912272321l5fc3b324v34b66bf221e81df3@mail.gmail.com>
References: <df38f5d30912272321l5fc3b324v34b66bf221e81df3@mail.gmail.com>
Message-ID: <000c01ca87d5$fa220dc0$ee662940$@com>

I have been with FreeSwitch for not so long but why don't you provide more
details about exactly what you want to do? Maybe I will be able to share
some thing.

 

Regards

 

Alex To

 

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of miracle
2012
Sent: Monday, December 28, 2009 3:22 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] pls help

 

Hi,

please provide any example c programs , APIs info, reference as i want to
write a dialer in c with freeswitch. 
i am a newbie for free switch... :|

thanks. 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091228/e5501bf4/attachment.html 

From brian at freeswitch.org  Mon Dec 28 08:08:55 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 28 Dec 2009 10:08:55 -0600
Subject: [Freeswitch-dev] pls help
In-Reply-To: <000c01ca87d5$fa220dc0$ee662940$@com>
References: <df38f5d30912272321l5fc3b324v34b66bf221e81df3@mail.gmail.com>
	<000c01ca87d5$fa220dc0$ee662940$@com>
Message-ID: <F05F42D2-5B3D-4A0B-9C7E-1587E6ED2FD9@freeswitch.org>

Maybe someone can get with mercutioviz on IRC and write up a how-to on the proper way to ask questions on the list and what info to provide... I have noticed an up tick in these short "HELP ME NOW PLEASE" emails and they have gotta stop.  Its noise and I just ignore them for the most part.  If you're not willing to do a little work providing details its only going to bog down our already limited resources trying to extract the info from a user.  I already do this but its like pulling teeth sometimes... VERY PAINFUL. 

Meeting Agenda items:

We need community volunteers.
Bug Tracker, Wiki, IRC. 
Project Etiquette (irc, wiki, tracker and lists)

The project is growing by leaps and bounds and very few are stepping up to the plate to help out.  

I thank everyone that has so far been helping out.  Even if its a little here and there because it helps greatly.  

Thank you,
/b

On Dec 28, 2009, at 9:53 AM, Alex To wrote:

> I have been with FreeSwitch for not so long but why don?t you provide more details about exactly what you want to do? Maybe I will be able to share some thing.
>  
> Regards
>  
> Alex To



From mike at jerris.com  Mon Dec 28 09:39:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 28 Dec 2009 12:39:10 -0500
Subject: [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip library
	(off-topic)
In-Reply-To: <8a384d790912240629q77a367eeg8f048fb3de232943@mail.gmail.com>
References: <8a384d790912240629q77a367eeg8f048fb3de232943@mail.gmail.com>
Message-ID: <6D058946-1D88-4C75-9B8D-E739AC1669B1@jerris.com>

how do you handle the passing of tags in lua?  does this end up being sane?

Mike

On Dec 24, 2009, at 9:29 AM, Paulo Pizarro wrote:

> Hi all,
> 
> We like to present to all Sofia SIP developers a Lua bind to Sofia SIP, Luasofia. The objective of this project is to be able to use Sofia on a more high level language, Lua was chosen because of its easy integration with C, a lot of our projects are starting to use Lua integrated with C to make development and maintenance easier and faster, because of that we wanted to have a good SIP library to use in Lua, the decision was to make a bind to the already mature Sofia library. The bind is far from being complete, we only bonded the code that we need right now, there's a lot of work to do. We expect that the project will be useful, so a lot of documentation was made to help someone who wants to bind more Sofia code. There is good documentation on how to use it too.
> 
> To develop we used git to make version control, and we hosted the project at github: 
> 
> http://github.com/ppizarro/luasofia
> 
> The wiki is at github too:
> 
> http://wiki.github.com/ppizarro/luasofia
> 
> We hope Luasofia can be useful to a lot of people, and eventually we get some help making it better and more complete.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091228/54e72ff2/attachment.html 

From msc at freeswitch.org  Mon Dec 28 10:21:02 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 28 Dec 2009 10:21:02 -0800
Subject: [Freeswitch-dev] pls help
In-Reply-To: <df38f5d30912272321l5fc3b324v34b66bf221e81df3@mail.gmail.com>
References: <df38f5d30912272321l5fc3b324v34b66bf221e81df3@mail.gmail.com>
Message-ID: <87f2f3b90912281021l3fab5b43se6b7596943075e84@mail.gmail.com>

On Sun, Dec 27, 2009 at 11:21 PM, miracle 2012 <2012.miracle at gmail.com>wrote:

> Hi,
>
> please provide any example c programs , APIs info, reference as i want to
> write a dialer in c with freeswitch.
> i am a newbie for free switch... :|
>
> thanks.
>

First things first. Make sure you get a working install of FreeSWITCH and
that you can do the basics. Get the latest SVN trunk and install it using
this procedure (if you haven't done so already):
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install

Then brush up on the basics of FS by considering this article:
http://bit.ly/EpVrv*

*Once you are comfortable with the basics of operating FreeSWITCH then think
about whether or not you really need to write your dialer module in C.
FreeSWITCH comes with an awesome event socket and corresponding library
(ESL) that lets you control FS from one or more separate systems. That might
be a better fit for you depending on your circumstances. More info on event
socket here:
http://wiki.freeswitch.org/wiki/Event_Socket

You have your homework assignment now. :) Also, if you want to talk in
real-time about this join the main channel and the dev channel in IRC:
#freeswitch and #freeswitch-dev on irc.freenode.net.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091228/670c2031/attachment.html 

From brian at freeswitch.org  Mon Dec 28 13:05:21 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 28 Dec 2009 15:05:21 -0600
Subject: [Freeswitch-dev] twitter.com/freeswitch (its not ours)
Message-ID: <F52EFF11-B321-4CA7-B3C3-BDFA221A9445@freeswitch.org>

Dear FreeSWITCHers,
	
	Someone has registered the freeswitch name and is squatting on twitter with it.  They haven't used it in over a year and I would like to have this for our project as its clearly confusing.

If you own this account please contact me off list.

Thanks,
Brian



From paulo.pizarro at gmail.com  Tue Dec 29 08:10:35 2009
From: paulo.pizarro at gmail.com (Paulo Pizarro)
Date: Tue, 29 Dec 2009 14:10:35 -0200
Subject: [Freeswitch-dev] LuaSofia is a Lua binding of Sofia-Sip library
	(off-topic)
In-Reply-To: <6D058946-1D88-4C75-9B8D-E739AC1669B1@jerris.com>
References: <8a384d790912240629q77a367eeg8f048fb3de232943@mail.gmail.com>
	<6D058946-1D88-4C75-9B8D-E739AC1669B1@jerris.com>
Message-ID: <8a384d790912290810h410fe9b2n40186b62ea289b25@mail.gmail.com>

Perhapses links below can help:

http://wiki.github.com/ppizarro/luasofia/tag-table

http://wiki.github.com/ppizarro/luasofia/tags-proxy

2009/12/28 Michael Jerris <mike at jerris.com>

> how do you handle the passing of tags in lua?  does this end up being sane?
>
> Mike
>
> On Dec 24, 2009, at 9:29 AM, Paulo Pizarro wrote:
>
> Hi all,
>
> We like to present to all Sofia SIP developers a Lua bind to Sofia SIP,
> Luasofia. The objective of this project is to be able to use Sofia on a more
> high level language, Lua was chosen because of its easy integration with C,
> a lot of our projects are starting to use Lua integrated with C to make
> development and maintenance easier and faster, because of that we wanted to
> have a good SIP library to use in Lua, the decision was to make a bind to
> the already mature Sofia library. The bind is far from being complete, we
> only bonded the code that we need right now, there's a lot of work to do. We
> expect that the project will be useful, so a lot of documentation was made
> to help someone who wants to bind more Sofia code. There is good
> documentation on how to use it too.
>
> To develop we used git to make version control, and we hosted the project
> at github:
>
> http://github.com/ppizarro/luasofia
>
> The wiki is at github too:
>
> http://wiki.github.com/ppizarro/luasofia
>
> We hope Luasofia can be useful to a lot of people, and eventually we get
> some help making it better and more complete.
>
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091229/676d8182/attachment-0001.html 

From lei.tlfly at gmail.com  Wed Dec 30 06:44:46 2009
From: lei.tlfly at gmail.com (Lei Tang)
Date: Wed, 30 Dec 2009 22:44:46 +0800
Subject: [Freeswitch-dev] FS doesn't update rtp port when sdp changed in
	200OK response for invite message.
In-Reply-To: <50c41b4e0912280709n172d6617x55b94f7f49d540a9@mail.gmail.com>
References: <50c41b4e0912240306h61b09d43rf68d4d5b28b4e219@mail.gmail.com>
	<87f2f3b90912241119t26fb6061p4b853576c7762841@mail.gmail.com>
	<FBA0B35D-49C6-4B60-8529-F0622B6FAC74@freeswitch.org>
	<50c41b4e0912280709n172d6617x55b94f7f49d540a9@mail.gmail.com>
Message-ID: <50c41b4e0912300644l639ac161k49a1181aae06904e@mail.gmail.com>

Hi  Brian, Do you think it is a bug of FS, or  is a by-design problem? How
ever, I think fs should update rtp port when receive UPDATE message, but it
seems doesn't.

2009/12/28 Lei Tang <lei.tlfly at gmail.com>

> Hi Brian, In my scenario, I connect FS to a  softswitch(bell alcatel),  the
> port change because the callee is using *CRBT(*Coloring Ring Back Tone*),
> I don't known much detail of the softswitch, I guessthere is a server
> process CRBT  and  another process **audio stream, so the port is changed,
> when I call a number without CRBT, the port is not changed.**   *
>
> 2009/12/28 Brian West <brian at freeswitch.org>
>
> Is it even legal to change the port like that?  I recall a discussion and
>> change to the code to allow this but as far as I can recall its not legal to
>> do that... I may be wrong.
>>
>> /b
>>
>> On Dec 24, 2009, at 1:19 PM, Michael Collins wrote:
>>
>> > Can you update to latest trunk and see if this still occurs? There have
>> been many updates since pre9.
>> > Thanks,
>> > MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Lei.Tang
> lei.tlfly at gmail.com
>



-- 
Lei.Tang
lei.tlfly at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091230/ad80ea12/attachment.html 

From msc at freeswitch.org  Wed Dec 30 09:00:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 30 Dec 2009 09:00:18 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <87f2f3b90912300900i678303e3ofcbba57fce2a0faa@mail.gmail.com>

Please join us today for a special Wednesday edition of the weekly
FreeSWITCH conference call:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091230/cd533d43/attachment.html 

From vipkilla at gmail.com  Wed Dec 30 09:49:19 2009
From: vipkilla at gmail.com (vip killa)
Date: Wed, 30 Dec 2009 12:49:19 -0500
Subject: [Freeswitch-dev] conference member's volume
Message-ID: <957f61370912300949qe99966wdd688468cf348cb4@mail.gmail.com>

is there anyway to auto-balance volume levels of members of the conference?
some members seem to be loud and overdrive.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091230/b3a81a45/attachment.html 

From msc at freeswitch.org  Wed Dec 30 16:02:16 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 30 Dec 2009 16:02:16 -0800
Subject: [Freeswitch-dev] Special Announcement: Latest FreeSWITCH Files
Message-ID: <87f2f3b90912301602u4697b0ccjf89e6ce89d8a6fec@mail.gmail.com>

Greetings all!

We would like to let everyone know that we have a new place for you to
download the latest FreeSWITCH source packages: http://latest.freeswitch.org.
The official announcement can be read here<http://www.freeswitch.org/node/225>.


Thanks for all your help in making FreeSWITCH a great project and a great
community.
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20091230/6d5adb95/attachment.html