date:20090327


Please direct the report to http://jira.freeswitch.org

/b

On Mar 27, 2009, at 2:58 AM, Jason White wrote:

While I was trying to obtain more detailed logs of my portaudio  
problems,

FreeSWITCH crashed, leaving a core file.

The backtraces are here:
http://pastebin.freeswitch.org/7998


Brian West
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Re: [Freeswitch-users] rubymod - ESL compile error


http://jira.freeswitch.org/browse/ESL-7

I think this might apply to you.

/b

On Mar 27, 2009, at 6:48 AM, Matthew Fong wrote:


/usr/bin/ld: cannot find -lruby
collect2: ld returned 1 exit status


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Re: [Freeswitch-users] sipp emulating a registered end point

2009-03-27 Thread Jonas Gauffin

thanks!

2009/3/27 Thomas Troy ttro...@gmail.com

 When I need to do something like this, what I do is set sipp2 to have 2
 scripts both using the same IP and port.

 One sends the register and deals with authentication \ OK.
 The other is then run after this to wait and receive the incoming call.

 So you would run them like

 sipp2_register   - Performs registration and ends
 sipp2_receiveCall  - Waits for incoming call, while listening
 on same IP \ Port as sipp2_register
 sipp1_makeCall- Makes outgoing call


 A sample register scenario is


 scenario name=Register
   send start_rtd=1 retrans=500
 ![CDATA[

   REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
   Via: SIP/2.0/[transport]
 [local_ip]:[local_port];branch=[branch]1;rport
   Max-Forwards: 70
   To: TestUser1 sip:u...@[remote_ip]:[remote_port]
   From: TestUser1 sip:user@
 [remote_ip]:[remote_port];tag=[pid][call_number]
   Call-ID: [call_id]
   CSeq: 1 REGISTER
   Contact: sip:u...@[local_ip]:[local_port]
   Expires: 240
   User-Agent: SIPp
   Content-Length: 0

 ]]
   /send

   recv response=401 rtd=1 auth=true
   /recv

   send start_rtd=2 retrans=500
 ![CDATA[

   REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
   Via: SIP/2.0/[transport]
 [local_ip]:[local_port];branch=[branch]2;rport
   Max-Forwards: 70
   To: TestUser1 sip:u...@[remote_ip]:[remote_port]
   From: TestUser1 sip:user@
 [remote_ip]:[remote_port];tag=[pid][call_number]
   Call-ID: [call_id]
   CSeq: 2 REGISTER
   Contact: sip:u...@[local_ip]:[local_port]
   Expires: 240
   User-Agent: SIPp
   [authentication username=user password=pass]
   Content-Length: 0

 ]]
   /send

   recv response=200 rtd=2
   /recv

   ResponseTimeRepartition value=10, 20, 30, 40, 50, 100, 250, 500/

 /scenario






 2009/3/26 Jonas Gauffin jonas.gauf...@gmail.com

 Hello
 I want to achive this: Sipp1 - FS - Sipp2

 Sipp1 emulates a inbound calls (easy to achive)
 Sipp2 should emulate a registered user (i.e. register with FS and then
 just wait for calls and hangup when sipp1 hangsup)

 How do I configure sipp as Sipp2?

 Thanks,
   Jonas

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Re: [Freeswitch-users] Multiple calls with PortAudio

Are you updating with make current each time?


On Fri, Mar 27, 2009 at 2:58 AM, Jason White ja...@jasonjgw.net wrote:

 While I was trying to obtain more detailed logs of my portaudio problems,
 FreeSWITCH crashed, leaving a core file.

 The backtraces are here:
  http://pastebin.freeswitch.org/7998

 As far as I can remember, at the time of the segfault, one channel was
 trying
 to connect and not succeeding; I had just issued a pa hangup command on it
 and
 then a pa call to try connecting again. Since my memory of exactly what was
 happening isn't as reliable as it should be, the value of the backtraces
 may
 be diminished.

 As to the portaudio problem, with rev. 12701 (Debian Sid, kernel 2.6.29,
 x86_64 architecture), the situation appears to be that the second and
 subsequent concurrent portaudio calls sometimes wait for a long time after
 issuing a log message such as the following:

  [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel
 sofia/internal/1...@192.168.0.2:5070[d6e56642-1a9b-11de-b23e-c5a9450df57d]

 These calls do not always complete successfully, but I'm still trying to
 collect more precise details of when and why they fail.

 With apologies for being unable to use Jira, if anything valuable appears
 in
 the backtraces, you are welcome to let me know via the list or by e-mail.

 I have previously seen crashes while working with multiple portaudio calls,
 but I don't yet have a reliable means to reproduce them.

 If the backtraces are revealing, that's good, but if not, that's fine too
 and
 I'll collect better particulars next time.


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Re: [Freeswitch-users] SIP INFO - RFC2833



On Mar 27, 2009, at 2:40 AM, rod wrote:


Hi,

I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
versa) and it's working fine when FS stays in the media path with
default configuration.

But my setup is the following:
   - Core network requires SIP INFO
   - Peerings require RFC2833

all would be fine with FS if my SIP Peers were not enforcing G729
(discarding G711) so that I have to use the directive  action
application=set data=proxy_media=true/ in my dialplan cause FS
can't deal with G729 except in pass-through.


Can't use proxy media in this case.  (I highly recommend you not use  
Proxy Media mode)




It's sad, but G729 is still a reality in Telco World.


Coming soon!



So do you think there could be a way to deal with DTMF even if not
analyzing RTP for transcoding. My commercial SBC is doing this, but it
sucks and that's the last step before final migration to FS.

regards,
rod


Brian West
br...@freeswitch.org

-- Meet us a ClueCon!  http://www.cluecon.com



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Re: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based

Sort of silly?,

I am not sure what you are talking about.
I
t's called *event*Consumer right? what do you mean by event based?

There is no need to create a session?

con = freeswitch.EventConsumer(all);

now you have a consumer obj

every time you call con:pop() with no arg you will either get an event or
nil when there are no events to consume.
every time you call con:pop(1) the consumer object will block until there is
an event.

So you use the first way in conjunction with some other lock to do async or
the 2nd way you do a dedicated blocking loop.

I don't know what you said in #lua but, umm du I think we have an event
driven programming under control.
We have a dedicated eventing engine in the core with scaling backend
dispatcher threads that can handle hundereds of thousand
of events at a time.

There is also Event Socket (the word *event* again) that can connect to tcp
and listen for *events*

you can also write your code in C with the trivial module API that allows
you to bind to an event internally and pretty much do whatever you want.

2009/3/27 Matthew Fong mattdf...@gmail.com

I've been playing around with using freeswitch.EventConsumer in a lua
process that starts-up when FS boots, and stays in the background. I've
setup the example on the wiki, but the example uses
session:execute(sleep,1000), and essentially loops every second until an
event is fired. I'm wondering if there is a more event-driven way to
accomplish this?
I tried asking for help in #lua, but they said the project (FS) needed to
implement event-driven programming for this to work. To me, it seems sort of
silly to implement freeswitch.EventConsumer without a way for it to be
executed event-wise

Is using lua ESL the only option? There isn't any lua example scripts in
libs/esl/lua to demonstrate how to handle events.

if mod_lua can't handle events, can the mod_javascript utilize it? Thanks.

--matt

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Re: [Freeswitch-users] High CPU load but only few sessions

2009-03-27 Thread Helmut Kuper

Hello,

today I killed that special thread via kill -9 a simple kill didn't
helped. Unfortunately this led to a normal shutdown of FS although I
killed not the parent process. :(

After restart of FS the server has a normal load again.

regards and a nice weekend
Helmut


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Re: [Freeswitch-users] SIP INFO - RFC2833

if you enable mod_g729 you can use freeswitch normally with that g729 codec
as long
as no transcoding is enabled (same passthru concept as proxy_media_mode)


On Fri, Mar 27, 2009 at 10:07 AM, rod kawa...@laposte.net wrote:

 Hi Brian,

 don't understand very well your advice:
 -- Can't use proxy media in this case.  (I highly recommend you not use
 Proxy Media mode)

 If i want to hide my topology network and deal with G729, I must use
 proxy media ?
 Why is Proxy media mode not recommended ??

 regards.
 rod



 Brian West wrote:
 
  On Mar 27, 2009, at 2:40 AM, rod wrote:
 
  Hi,
 
  I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
  versa) and it's working fine when FS stays in the media path with
  default configuration.
 
  But my setup is the following:
 - Core network requires SIP INFO
 - Peerings require RFC2833
 
  all would be fine with FS if my SIP Peers were not enforcing G729
  (discarding G711) so that I have to use the directive  action
  application=set data=proxy_media=true/ in my dialplan cause FS
  can't deal with G729 except in pass-through.
 
  Can't use proxy media in this case.  (I highly recommend you not use
  Proxy Media mode)
 
 
  It's sad, but G729 is still a reality in Telco World.
 
  Coming soon!
 
 
  So do you think there could be a way to deal with DTMF even if not
  analyzing RTP for transcoding. My commercial SBC is doing this, but it
  sucks and that's the last step before final migration to FS.
 
  regards,
  rod
 
  Brian West
  br...@freeswitch.org mailto:br...@freeswitch.org
 
  -- Meet us a ClueCon!  http://www.cluecon.com http://www.cluecon.com/
 
 
 
  
 
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[Freeswitch-users] generating RFC 3966 and RFC 4694 calls

2009-03-27 Thread James H Thompson

I need to generate calls with Invite URIs in this format:

INVITE sip:9085551212;npdi=yes;rn=9083820...@204.123.123.123:5060 SIP/2.0 

Is there an easy way to do this?

Thanks.
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[Freeswitch-users] differences between mod_fifo and asterisk queues

2009-03-27 Thread Francois Delawarde

Hello and welcome me into FreeSWITCH's world! = sorry that was rude

I am (hoping to say I was soon) a heavy user of Asterisk's call queues
for small call centers with sometimes empty queues and all agents idle
for a few seconds.

FreeSWITCH's mod_fifo algorithm is apparently quite different than
Asterisk's app_queue. Instead of choosing an agent for a each call once
it gets to the bottom of the queue given a specific strategy, FreeSWITCH
does the inverse and finds a call once an agent is free given a strategy
(the call that has waited longer from all the agent's queues, or the
call in the queue that currently has more calls waiting). Am I right?

If the above deduction is correct, while it seems a MUCH better choice
for heavier call centers that always have calls in their queues (in
queue calls are not delayed by the processing of the call at the end of
the queue), I have a few doubts for what would happen in small call
centers when those queues sometimes get empty and several agents fight
for the incoming calls. My questions are following:

- If for example 4 agents are connected (fifo out) to an empty queue,
what happens when a call arrives? Do the 4 agents ring? If not, how do
we know which agent get the call?
- Is there an [easy] way (with some javascript or similar) to emulate
Asterisk's distribution strategies to agents (by amount of time without
calls, total number of answered calls, round robing, ...) in this cases?

A couple of other newbie questions that has nothing to do with the
above:
- Is there a way to execute some PHP scripts for each call that would do
the bridging or call applications (like Asterisk's AGI)?
- What is the recommended language for features, speed, and programming
ease (not a priority) for this kind of scripts (C? LUA?,
Javascript?, ..)?

Thanks in advance,
François.
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Re: [Freeswitch-users] Losing Gateway registration

Look closer at the logs or sip trace, this sounds like a failed  
session timer to me.


Mike

On Mar 27, 2009, at 8:34 AM, Andy Ayers wrote:

Thanks for your help folks, the ping parameter seems to have  
resolved the gateway connection issue but I now seem to be having a  
related issue with calls being cut off after a number of seconds.  
The freeswitch logs show a normal call clearing. I am indeed behind  
a NAT firewall which I'm assuming is the main issue. do you have any  
further tips to make this more stable and prevent the call cut off?


Many thanks
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Mathieu Rene

Sent: 18 March 2009 14:46
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Losing Gateway registration

if you are behind NAT it is possible that your router forgot the  
mapping betweeen FS and your provider, try addingparam  
name=ping value=30 / to your gateway.


Math

On 18-Mar-09, at 10:07 AM, Brian West wrote:


Upgrade to 1.03 or SVN Trunk

/b

On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote:


Hi,

I've recently ugrade to version 1.02 of freeswitch and am having  
some problems with my gateway registrations. The gateway  
successfully registers with my voip provider when freeswitch first  
starts but if left running it seems to loose it's connection to my  
voip provider. I can get it to reconnect with a sofia restart. I'm  
using the same provider and user account as with the old version  
of the software. Can you suggest any reaosn why this may be  
happening and how I can prevent it?


Many thanks
Andy


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Re: [Freeswitch-users] rubymod - ESL compile error


fixed revision 12805.

Mike

On Mar 27, 2009, at 9:49 AM, Brian West wrote:


http://jira.freeswitch.org/browse/ESL-7

I think this might apply to you.

/b

On Mar 27, 2009, at 6:48 AM, Matthew Fong wrote:


/usr/bin/ld: cannot find -lruby
collect2: ld returned 1 exit status


Brian West
br...@freeswitch.org

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Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls

You seem to be confusing your standards, those 2 specs are about tel:  
uri's not sip: uris.  Sending a tel uri I am not sure we can do, where  
would we send it to?


Mike

On Mar 27, 2009, at 6:11 AM, James H Thompson wrote:


I need to generate calls with Invite URIs in this format:

INVITE sip:9085551212;npdi=yes;rn=9083820...@204.123.123.123:5060  
SIP/2.0


Is there an easy way to do this?

Thanks.

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Re: [Freeswitch-users] differences between mod_fifo and asterisk queues

2009/3/27 Francois Delawarde fdelawa...@wirelessmundi.com

  Hello and welcome me into FreeSWITCH's world! = sorry that was rude

 I am (hoping to say I was soon) a heavy user of Asterisk's call queues
 for small call centers with sometimes empty queues and all agents idle for a
 few seconds.

 FreeSWITCH's mod_fifo algorithm is apparently quite different than
 Asterisk's app_queue. Instead of choosing an agent for a each call once it
 gets to the bottom of the queue given a specific strategy, FreeSWITCH does
 the inverse and finds a call once an agent is free given a strategy (the
 call that has waited longer from all the agent's queues, or the call in the
 queue that currently has more calls waiting). Am I right?

 If the above deduction is correct, while it seems a MUCH better choice for
 heavier call centers that always have calls in their queues (in queue
 calls are not delayed by the processing of the call at the end of the
 queue), I have a few doubts for what would happen in small call centers when
 those queues sometimes get empty and several agents fight for the incoming
 calls. My questions are following:

 - If for example 4 agents are connected (fifo out) to an empty queue,
 what happens when a call arrives? Do the 4 agents ring? If not, how do we
 know which agent get the call?


If you are using on-hook agents, it will place as many outbound calls as
there are people waiting.
If you are using off-hook agents it will just connect the first free agent.



 - Is there an [easy] way (with some javascript or similar) to emulate
 Asterisk's distribution strategies to agents (by amount of time without
 calls, total number of answered calls, round robing, ...) in this cases?


Easiest way would be to write a patch in C to mod_fifo it'self or propose a
bounty for features and see if you can get the change approved by the
developers.




 A couple of other newbie questions that has nothing to do with the above:
 - Is there a way to execute some PHP scripts for each call that would do
 the bridging or call applications (like Asterisk's AGI)?


Your best bet would be to not try to do anything like asterisk FreeSWITCH
is a paradigm shift from asterisk and you may defeat yourself trying to do
anything the same way.

That said, yes, look at Event Socket and ESL, (using asterisk terminology,
it's a combination of AGI and manager).



 - What is the recommended language for features, speed, and programming
 ease (not a priority) for this kind of scripts (C? LUA?, Javascript?, ..)?


C




 Thanks in advance,
 François.

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Re: [Freeswitch-users] Building on Ubuntu Intrepid

2009-03-27 Thread dujinfang



On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote:

We do not support ubuntu interpid, it has at least 3 known fatal  
issues not experienced by all but nonetheless enough to make us  
unwilling to support it.


I use Ubuntu gutsy in production  and interipid in test env. It works  
well. Can you briefly explain the 3 fatal issues Anthony? It will help  
me know potential risks.





It's use at your own risk or use the stable branch hardy for any  
support.




On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds tre...@concipient.net 
 wrote:

Has there been any progress getting FreeSWITCH to build on Ubuntu
Intrepid without downgrading libtool?



I successfully built FS on intrepid. I simply done this by changing  
the apt-source to Hardy and installed libtool. Obviously I changed the  
apt-source back to intrepid after I installed libtool.


And, another approach. Install libtool from source should be as easy  
as  configure  make  make install. I done this on a new CentOS4  
because the default yum install of libtool on CentOS4 is old than FS  
required.




Thanks!

Sincerely,
Trevor Hammonds

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Re: [Freeswitch-users] SIP INFO - RFC2833

2009-03-27 Thread rod

Hello,

I have this error when not enablig proxy_media:
2009-03-27 19:54:44 [ERR] mod_g729.c:145 switch_g729_decode() This codec 
is only usable in passthrough mode!
2009-03-27 19:54:44 [ERR] switch_core_io.c:723 
switch_core_session_write_frame() Codec G.729 decoder error!

Sure there is an option to check. Any pointers.

regards.




Anthony Minessale wrote:
 if you enable mod_g729 you can use freeswitch normally with that g729 
 codec as long
 as no transcoding is enabled (same passthru concept as proxy_media_mode)


 On Fri, Mar 27, 2009 at 10:07 AM, rod kawa...@laposte.net 
 mailto:kawa...@laposte.net wrote:

 Hi Brian,

 don't understand very well your advice:
 -- Can't use proxy media in this case.  (I highly recommend you
 not use
 Proxy Media mode)

 If i want to hide my topology network and deal with G729, I must use
 proxy media ?
 Why is Proxy media mode not recommended ??

 regards.
 rod



 Brian West wrote:
 
  On Mar 27, 2009, at 2:40 AM, rod wrote:
 
  Hi,
 
  I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
  versa) and it's working fine when FS stays in the media path with
  default configuration.
 
  But my setup is the following:
 - Core network requires SIP INFO
 - Peerings require RFC2833
 
  all would be fine with FS if my SIP Peers were not enforcing G729
  (discarding G711) so that I have to use the directive  action
  application=set data=proxy_media=true/ in my dialplan cause FS
  can't deal with G729 except in pass-through.
 
  Can't use proxy media in this case.  (I highly recommend you not use
  Proxy Media mode)
 
 
  It's sad, but G729 is still a reality in Telco World.
 
  Coming soon!
 
 
  So do you think there could be a way to deal with DTMF even if not
  analyzing RTP for transcoding. My commercial SBC is doing this,
 but it
  sucks and that's the last step before final migration to FS.
 
  regards,
  rod
 
  Brian West
  br...@freeswitch.org mailto:br...@freeswitch.org
 mailto:br...@freeswitch.org mailto:br...@freeswitch.org
 
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 http://www.cluecon.com/
 
 
 
 
 
 
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[Freeswitch-users] condition matching on variables which have been set in the dialplan

2009-03-27 Thread Peter P GMX

I try to use speed dialling and masked numbers in a dialplan through
xml-curl. For the XML I use templates which I fill with variables. The
numbering plan is set up in a way that any number can be a speed
dialling or masked number, so I cannot parse them via Regex in the XML
part of the dialplan. E.g.

* 12345 is a normal phone
* 12346 is a speed dialling number = 0049xx
* 12347 is a normal phone
* 4 is a speed dialling number = 0049xx

So I need to substitute a variable with the final number to be dialled.
This final number then needs to be parsed in the dialplan to indentify
how to handle it (bridge, conference, voicemail etc.)
I have special reasons to do that, so please do not ask me why.

So the dialplan is as following

extension name=Any !-- Set the variables --
  condition field=destination_number expression=^[0-9]\d[0,16}$
continue=true.
action application=set data=destination_number=0049x/.
action application=export
data=destination_number=0049x/.
action application=info/
  /condition
  condition field=${variable_destination_number}
expression=^(00[1-9]\d{4,13})$ !-- Now parse the new variables --
   action application=set
data=effective_caller_id_number=unknown/.
   action application=set data=effective_caller_id_name=unknown/.
   action application=bridge
data=sofia/gateway/QSC_DE/$...@sip.qsc.de/.
  /condition
  condition 
   .
   .
/extension

In the first condition I set the substituted final destination number.
This is dynamically substituted in the template in my application via
xml-curl dependend on which kind of number is dialled. In this case a
German number is substituted.
In the following conditions I would like to set the gateways.

What is happening in the logs?

* I dial e.g. 12346 for a speed dialling number
* The first condition is parsed correctly, and the variables are set
  (Action set(destination_number=0049)
* in the second condition ${variable_destination_number} is not set
  to the new value. It's still 12346.(I also tried conditions
  based on ${destination_number} and destination_number).
* In the logs the execution of set and export in fact is shown
  after the whole conditions are parsed. Also the info application
  is outputted after all conditions are parsed.
  E.g. EXECUTE sofia/internal/10...@sip.domain.de
  set(destination_number=0049)
* the info app shows me that variable_destination_number is set
  to the right number, but it seems to be too late?

Question: Are these lines not handled sequentially (I am using a quad
core machine)? Any other idea how to solve this?

Best regards
Peter



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Re: [Freeswitch-users] condition matching on variables which have been set in the dialplan

Remember the dialplan is NOT executed when its parsed so you can't set  
a var then condition on that exact var on the next line.. that var  
doesn't exist.


/b


On Mar 27, 2009, at 11:25 AM, Peter P GMX wrote:


   extension name=Any !-- Set the variables --
 condition field=destination_number  
expression=^[0-9]\d[0,16}$

continue=true.
   action application=set  
data=destination_number=0049x/.

   action application=export
data=destination_number=0049x/.
   action application=info/
 /condition
 condition field=${variable_destination_number}


Its ${destination_number}


expression=^(00[1-9]\d{4,13})$ !-- Now parse the new variables --
  action application=set
data=effective_caller_id_number=unknown/.
  action application=set  
data=effective_caller_id_name=unknown/.

  action application=bridge
data=sofia/gateway/QSC_DE/$...@sip.qsc.de/.
 /condition
 condition 
  .
  .
   /extension


Brian West
br...@freeswitch.org

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Re: [Freeswitch-users] Building on Ubuntu Intrepid

Another example of a fatal issue was the optimizer in gcc was breaking  
openzap code even with -O2.


Mike

On Mar 27, 2009, at 12:12 PM, dujinfang wrote:



On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote:
We do not support ubuntu interpid, it has at least 3 known fatal  
issues not experienced by all but nonetheless enough to make us  
unwilling to support it.


I use Ubuntu gutsy in production  and interipid in test env. It  
works well. Can you briefly explain the 3 fatal issues Anthony? It  
will help me know potential risks.


It's use at your own risk or use the stable branch hardy for  
any support.


On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds tre...@concipient.net 
 wrote:

Has there been any progress getting FreeSWITCH to build on Ubuntu
Intrepid without downgrading libtool?



I successfully built FS on intrepid. I simply done this by changing  
the apt-source to Hardy and installed libtool. Obviously I changed  
the apt-source back to intrepid after I installed libtool.


And, another approach. Install libtool from source should be as easy  
as  configure  make  make install. I done this on a new CentOS4  
because the default yum install of libtool on CentOS4 is old than FS  
required.
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Re: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based

2009-03-27 Thread Michael Collins

 con = freeswitch.EventConsumer(all);

 now you have a consumer obj

 every time you call con:pop() with no arg you will either get an event or
 nil when there are no events to consume.
 every time you call con:pop(1) the consumer object will block until there is
 an event.

 So you use the first way in conjunction with some other lock to do async or
 the 2nd way you do a dedicated blocking loop.

FYI, I added this information to the wiki page for freeswitch.EventConsumer.
-MC

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Re: [Freeswitch-users] Building on Ubuntu Intrepid

2009-03-27 Thread Gabriel Kuri

regarding gcc compiler optimizations, are they generally compatible with
FS or should they be removed or does the configure strip them out? just
curious, as I run Gentoo and use such optimizations as -march=nocona
-O2 -pipe -fomit-frame-pointer

not sure if they break things or I should be removing them before
compiling FS?

Gabe

Michael Jerris wrote:
 Another example of a fatal issue was the optimizer in gcc was breaking
 openzap code even with -O2.
 
 Mike
 
 On Mar 27, 2009, at 12:12 PM, dujinfang wrote:
 

 On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote:
 We do not support ubuntu interpid, it has at least 3 known fatal
 issues not experienced by all but nonetheless enough to make us
 unwilling to support it.

 I use Ubuntu gutsy in production  and interipid in test env. It works
 well. Can you briefly explain the 3 fatal issues Anthony? It will help
 me know potential risks.

 It's use at your own risk or use the stable branch hardy for any
 support.

 On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds
 tre...@concipient.net mailto:tre...@concipient.net wrote:

 Has there been any progress getting FreeSWITCH to build on Ubuntu
 Intrepid without downgrading libtool?


 I successfully built FS on intrepid. I simply done this by changing
 the apt-source to Hardy and installed libtool. Obviously I changed the
 apt-source back to intrepid after I installed libtool.

 And, another approach. Install libtool from source should be as easy
 as  configure  make  make install. I done this on a new CentOS4
 because the default yum install of libtool on CentOS4 is old than FS
 required.
 
 
 
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Re: [Freeswitch-users] Building on Ubuntu Intrepid

Usually if you don't know what they do... then you shouldn't use  
them!  ;)



/b

On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote:

regarding gcc compiler optimizations, are they generally compatible  
with
FS or should they be removed or does the configure strip them out?  
just

curious, as I run Gentoo and use such optimizations as -march=nocona
-O2 -pipe -fomit-frame-pointer

not sure if they break things or I should be removing them before
compiling FS?

Gabe


Brian West
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Re: [Freeswitch-users] Building on Ubuntu Intrepid

2009-03-27 Thread Gabriel Kuri

I'm not asking what they do, I'm asking those more familiar with FS
whether the optimization flags are too aggressive for FS. What do you
guys (developers) normalize use, just your basic -march=i686 -pipe ?

Gabe


Brian West wrote:
 Usually if you don't know what they do... then you shouldn't use them!  ;)
 
 
 /b
 
 On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote:
 
 regarding gcc compiler optimizations, are they generally compatible with
 FS or should they be removed or does the configure strip them out? just
 curious, as I run Gentoo and use such optimizations as -march=nocona
 -O2 -pipe -fomit-frame-pointer

 not sure if they break things or I should be removing them before
 compiling FS?

 Gabe
 
 Brian West
 br...@freeswitch.org mailto:br...@freeswitch.org
 
 -- Meet us a ClueCon!  http://www.cluecon.com http://www.cluecon.com/
 
 
 
 
 
 
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Re: [Freeswitch-users] SIP INFO - RFC2833

you have to set disable-transcoding as well to avoid any transcoding
situations

On Fri, Mar 27, 2009 at 11:23 AM, rod kawa...@laposte.net wrote:

 Hello,

 I have this error when not enablig proxy_media:
 2009-03-27 19:54:44 [ERR] mod_g729.c:145 switch_g729_decode() This codec
 is only usable in passthrough mode!
 2009-03-27 19:54:44 [ERR] switch_core_io.c:723
 switch_core_session_write_frame() Codec G.729 decoder error!

 Sure there is an option to check. Any pointers.

 regards.




 Anthony Minessale wrote:
  if you enable mod_g729 you can use freeswitch normally with that g729
  codec as long
  as no transcoding is enabled (same passthru concept as proxy_media_mode)
 
 
  On Fri, Mar 27, 2009 at 10:07 AM, rod kawa...@laposte.net
  mailto:kawa...@laposte.net wrote:
 
  Hi Brian,
 
  don't understand very well your advice:
  -- Can't use proxy media in this case.  (I highly recommend you
  not use
  Proxy Media mode)
 
  If i want to hide my topology network and deal with G729, I must use
  proxy media ?
  Why is Proxy media mode not recommended ??
 
  regards.
  rod
 
 
 
  Brian West wrote:
  
   On Mar 27, 2009, at 2:40 AM, rod wrote:
  
   Hi,
  
   I did some tests with FS to transcode SIP INFO to RFC2833 (and
 vice
   versa) and it's working fine when FS stays in the media path with
   default configuration.
  
   But my setup is the following:
  - Core network requires SIP INFO
  - Peerings require RFC2833
  
   all would be fine with FS if my SIP Peers were not enforcing G729
   (discarding G711) so that I have to use the directive  action
   application=set data=proxy_media=true/ in my dialplan cause
 FS
   can't deal with G729 except in pass-through.
  
   Can't use proxy media in this case.  (I highly recommend you not
 use
   Proxy Media mode)
  
  
   It's sad, but G729 is still a reality in Telco World.
  
   Coming soon!
  
  
   So do you think there could be a way to deal with DTMF even if not
   analyzing RTP for transcoding. My commercial SBC is doing this,
  but it
   sucks and that's the last step before final migration to FS.
  
   regards,
   rod
  
   Brian West
   br...@freeswitch.org mailto:br...@freeswitch.org
  mailto:br...@freeswitch.org mailto:br...@freeswitch.org
  
   -- Meet us a ClueCon!  http://www.cluecon.com
  http://www.cluecon.com/
  
  
  
  
 
 
  
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Re: [Freeswitch-users] DTMF Missing Digits

2009-03-27 Thread Chris Fowler


Sent: Wednesday, March 25, 2009 12:43

btw you'll have to reinstall your phrase macros  make vm-sync I  
think should do it if it doesn't let me know... we removed the 250ms  
sleeps and that was the problem which we fixed.


I re-did the macros; the only change I could detect was the elimination
of the 250ms sleeps; and the change to:
macro name=welcome pause=250

I'm running build 12782; should this have fixed it?  If so, I will
follow the bug reporting instructions you sent earlier.

Thanks, Chris.



Here's the errors caught today on my production system.  

2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'000'
2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'000'
2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'1101'
2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'55'
2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'015'
2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'000'
2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'000'
2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'000'
2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'000'
2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
''
2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'006'
2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'006'
2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'006'
2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'006'
2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input
'006'

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Re: [Freeswitch-users] Building on Ubuntu Intrepid


We usually don't specify anything extra!

/b

On Mar 27, 2009, at 1:56 PM, Gabriel Kuri wrote:


I'm not asking what they do, I'm asking those more familiar with FS
whether the optimization flags are too aggressive for FS. What do you
guys (developers) normalize use, just your basic -march=i686 -pipe ?

Gabe


Brian West
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Re: [Freeswitch-users] Building on Ubuntu Intrepid

We've made no attempts to add any optimization flags on unix to date.
We use the defaults and always build debug binaries.

When we get some spare time we might go back and turn them on but so far
we don't have much of a need to.


On Fri, Mar 27, 2009 at 1:56 PM, Gabriel Kuri gk...@ieee.org wrote:

 I'm not asking what they do, I'm asking those more familiar with FS
 whether the optimization flags are too aggressive for FS. What do you
 guys (developers) normalize use, just your basic -march=i686 -pipe ?

 Gabe


 Brian West wrote:
  Usually if you don't know what they do... then you shouldn't use them!
  ;)
 
 
  /b
 
  On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote:
 
  regarding gcc compiler optimizations, are they generally compatible with
  FS or should they be removed or does the configure strip them out? just
  curious, as I run Gentoo and use such optimizations as -march=nocona
  -O2 -pipe -fomit-frame-pointer
 
  not sure if they break things or I should be removing them before
  compiling FS?
 
  Gabe
 
  Brian West
  br...@freeswitch.org mailto:br...@freeswitch.org
 
  -- Meet us a ClueCon!  http://www.cluecon.com http://www.cluecon.com/
 
 
 
 
  
 
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Re: [Freeswitch-users] condition matching on variables which have been set in the dialplan

2009-03-27 Thread Peter P GMX

OK, understood. I will do it in a different way then.


Brian West schrieb:
 Remember the dialplan is NOT executed when its parsed so you can't set
 a var then condition on that exact var on the next line.. that var
 doesn't exist.

 /b


 On Mar 27, 2009, at 11:25 AM, Peter P GMX wrote:

extension name=Any !-- Set the variables --
  condition field=destination_number expression=^[0-9]\d[0,16}$
 continue=true.
action application=set
 data=destination_number=0049x/.
action application=export
 data=destination_number=0049x/.
action application=info/
  /condition
  condition field=${variable_destination_number}

 Its ${destination_number}

 expression=^(00[1-9]\d{4,13})$ !-- Now parse the new variables --
   action application=set
 data=effective_caller_id_number=unknown/.
   action application=set
 data=effective_caller_id_name=unknown/.
   action application=bridge
 data=sofia/gateway/QSC_DE/$...@sip.qsc.de
 mailto:sofia/gateway/QSC_DE/$...@sip.qsc.de/.
  /condition
  condition 
   .
   .
/extension

 Brian West
 br...@freeswitch.org mailto:br...@freeswitch.org

 -- Meet us a ClueCon!  http://www.cluecon.com http://www.cluecon.com/



 

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Re: [Freeswitch-users] condition matching on variables which have been set in the dialplan

You can execute_extension to revisit the dialplan at a later time  
kinda like a macro.


/b

On Mar 27, 2009, at 2:23 PM, Peter P GMX wrote:


OK, understood. I will do it in a different way then.


Brian West
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Re: [Freeswitch-users] DTMF Missing Digits

2009-03-27 Thread Chris Fowler

 Did you provide the menu you are using and what you expect to happen?

Here's the setup;  

Caller - FlowRoute -  FreeSwitch

menu name=main_ivr
  greet-long=phrase:welcome
  greet-short=phrase:top-menu
  invalid-sound=ivr/ivr-that_was_an_invalid_entry.wav
  exit-sound=ivr/ivr-operator.wav
  timeout =1
  inter-digit-timeout=1500
  max-failures=2
  max-timeouts=7
  digit-len=4
  entry action=menu-exec-app digits=/^(10[0-2][0-9])$/
  param=transfer $1 XML public/
  entry action=menu-exec-app digits=/^(30\d{2})$/
  param=transfer $1 XML default/
  entry action=menu-exec-app digits=0 param=transfer 1000 XML
  public/  !-- Send to the operator extension --
  entry action=menu-exec-app digits=# param=transfer 6000 XML
  default/
/menu

macro name=welcome pause=250
  input pattern=(.*)
match
  action function=play-file
  data=/usr/local/freeswitch/sounds/fr1.wav/
  action function=play-file
  data=/usr/local/freeswitch/sounds/fr2.wav/
  action function=play-file
  data=/usr/local/freeswitch/sounds/if-u-know-ext-dial.wav/
  action function=play-file
  data=/usr/local/freeswitch/sounds/fr3.wav/
/match
  /input
/macro



macro name=top-menu pause=250
  input pattern=(.*)
match
  action function=play-file
  data=/usr/local/freeswitch/sounds/if-u-know-ext-dial.wav/
  action function=play-file
  data=/usr/local/freeswitch/sounds/fr3.wav/
/match
  /input
/macro



 B: Right and that is the fix for this.  If you have the sleep's in your 
 phrase macro's remove them and use the pause= param... you shouldn't have 
 any problems.

Still seeing multiple issues logged during ivr process for
mis-interpreted DTMF.  Here's today's list from our production server.

2009-03-27 06:38:59 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
'1100'
2009-03-27 07:20:33 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
'1101'
2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '55'
2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '015'
2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
''
2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 11:58:35 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '028'
2009-03-27 11:59:27 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '050'
2009-03-27 12:01:52 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 12:02:01 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 12:02:41 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 12:02:53 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'


Any other debug I can capture to assist?

Thanks, Chris.

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[Freeswitch-users] Contacting Callie

2009-03-27 Thread Kristian Kielhofner

What is the best way (if any) to contact Callie for custom prompt
work?  I can't seem to find much about her.

Thanks!

-- 
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http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls

2009-03-27 Thread James H Thompson

Calls would be sent to the IP address after the '@'  in the URI.
Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as the user 
part of a SIP URI.
My example Invite URI is the way we are receiving traffic from some of the 
major telecom carriers.
We would like be able to generate calls using the same formats.


- Original Message - 
From: Michael Jerris 
To: freeswitch-users@lists.freeswitch.org 
Sent: Friday, March 27, 2009 5:41 AM
Subject: Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls


You seem to be confusing your standards, those 2 specs are about tel: uri's not 
sip: uris.  Sending a tel uri I am not sure we can do, where would we send it 
to?


Mike


On Mar 27, 2009, at 6:11 AM, James H Thompson wrote:


  I need to generate calls with Invite URIs in this format:

  INVITE sip:9085551212;npdi=yes;rn=9083820...@204.123.123.123:5060 SIP/2.0

  Is there an easy way to do this?

  Thanks.

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Re: [Freeswitch-users] Contacting Callie


Well I do get a discount if we batch them.

I'm taking donations for this order br...@freeswitch.org is my paypal  
we are sending the order out monday but I only have a handful of stuff  
to record this go around.


http://jira.freeswitch.org/browse/FSSCRIPTS-15

William, Thanks for your donation to help pay for it.  ;)

/b


On Mar 27, 2009, at 3:04 PM, William Suffill wrote:

Good question. Last I talked to Brian about this (new prompts for  
upcoming new release) all the prompts are done by http://www.gmvoices.com/ 
. I don't know anything more about the process to get recordings  
done or if there is any preferred process if they are from users of  
FreeSwitch but be curious to find out.


-- W


Brian West
br...@freeswitch.org

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Re: [Freeswitch-users] IRC is not for all

2009-03-27 Thread Addison Martin

Also, moving the list to Google Groups would allow email OR threaded
views, and personally I like them better than nabble.

anm_


On Thu, Mar 26, 2009 at 11:06 AM, Michael Jerris m...@jerris.com wrote:
 http://n2.nabble.com/freeswitch-users-f2379917.html

 Mike


 On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote:

 Is there nothing out there that integrates a forum with a mailing
 list?
 It seems like one could display the mailing list archives exactly
 like a
 forum, and allow users to register to the forum and post (appearing to
 the mailing list as usern...@forumurl.org) in such a way that they
 don't
 have to realize it's a mailing list.

 Anthony Minessale wrote:
 The guy started a forum almost a month ago and as you can see nobody
 knows the url and it has no posts.

 http://freeswitch411.info/forum/

 This is one of the problems I was worried about when endorsing a
 forum.



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Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls

if you prefix the sofia dial string with sip: you should be able to pass
anything you want.

sofia/internal/sip:9085551212;npdi=yes;rn=9083820...@204.123.123.123:5060rn=9083820...@204.123.123.123:5060

2009/3/27 James H Thompson j...@lj.net

  Calls would be sent to the IP address after the '@' %...@%27  in the
 URI.
 Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as the
 user part of a SIP URI.
 My example Invite URI is the way we are receiving traffic from some of the
 major telecom carriers.
 We would like be able to generate calls using the same formats.


 - Original Message - *From:* Michael Jerris m...@jerris.com
 *To:* freeswitch-users@lists.freeswitch.org
 *Sent:* Friday, March 27, 2009 5:41 AM
 *Subject:* Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls

 You seem to be confusing your standards, those 2 specs are about tel: uri's
 not sip: uris.  Sending a tel uri I am not sure we can do, where would we
 send it to?
 Mike

  On Mar 27, 2009, at 6:11 AM, James H Thompson wrote:

  I need to generate calls with Invite URIs in this format:

 INVITE 
 sip:9085551212;npdi=yes;rn=9083820...@204.123.123.123:5060rn=9083820...@204.123.123.123:5060
  SIP/2.0

 Is there an easy way to do this?

 Thanks.

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googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] High CPU load but only few sessions

2009-03-27 Thread Rupa Schomaker

kill -9 on a thread will kill the process which kills freeswitch.

On Fri, Mar 27, 2009 at 10:09 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:

 Hello,

 today I killed that special thread via kill -9 a simple kill didn't
 helped. Unfortunately this led to a normal shutdown of FS although I
 killed not the parent process. :(

 After restart of FS the server has a normal load again.

 regards and a nice weekend
 Helmut




-- 
-Rupa
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Re: [Freeswitch-users] IRC is not for all

2009-03-27 Thread James H Thompson

The freeswitch user mailing list is also on:
http://news.gmane.org/gmane.comp.telephony.freeswitch.user

There are several forums packages that allow feeding in a mailing list, 
although
not many people seem to do it.  Google Groups and Yahoo Groups are also
alternatives.

I've been considering mirroring some of the major voip mailing lists on 
voip-info.org
into a  forum of somekind.

If this would be of interest let me know.

Jim


- Original Message - 
From: Addison Martin freeswi...@servercorps.com
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, March 27, 2009 10:14 AM
Subject: Re: [Freeswitch-users] IRC is not for all


 Also, moving the list to Google Groups would allow email OR threaded
 views, and personally I like them better than nabble.

 anm_


 On Thu, Mar 26, 2009 at 11:06 AM, Michael Jerris m...@jerris.com wrote:
 http://n2.nabble.com/freeswitch-users-f2379917.html

 Mike


 On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote:

 Is there nothing out there that integrates a forum with a mailing
 list?
 It seems like one could display the mailing list archives exactly
 like a
 forum, and allow users to register to the forum and post (appearing to
 the mailing list as usern...@forumurl.org) in such a way that they
 don't
 have to realize it's a mailing list.

 Anthony Minessale wrote:
 The guy started a forum almost a month ago and as you can see nobody
 knows the url and it has no posts.

 http://freeswitch411.info/forum/

 This is one of the problems I was worried about when endorsing a
 forum.



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Re: [Freeswitch-users] IRC is not for all

2009-03-27 Thread Jason White

James H Thompson j...@lj.net wrote:
 I've been considering mirroring some of the major voip mailing lists on 
 voip-info.org
 into a  forum of somekind.

Have a look at http://www.gmane.org/
and note that you can post via NNTP or via the WEb.

This mailing list is subscribed to gmane.


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Re: [Freeswitch-users] IRC is not for all

2009-03-27 Thread Jason White

Addison Martin freeswi...@servercorps.com wrote:
 Also, moving the list to Google Groups would allow email OR threaded
 views, and personally I like them better than nabble.

http://dir.gmane.org/gmane.comp.telephony.freeswitch.user

Would any of those views suffice?


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Re: [Freeswitch-users] Building on Ubuntu Intrepid

2009-03-27 Thread dujinfang


Thanks.

I was thinking  created a new server with Ubuntu intrepid, seems I'd  
like back to  Hardy.
Even on hardy the default libtool is version 2. replace to libtool 1  
should be easy as I mentioned before.



On Mar 28, 2009, at 3:04 AM, Anthony Minessale wrote:

1) There is an incompatibility on the fake ncurses wrapper that  
causes an instant seg fault unless you install the real ncurses.


On ubuntu it's libncurses5-dev, use for simple's sake.


2) The bleeding edge GCC builds an openzap binary that crashes  
instantly with no explanation in the core file from a minimal -O2  
(that's just the one copmiler bug that we know about for sure, like  
cock roaches, see one, there are probably 1000)
we don't use openzap. Is the probably 1000 all in the openzap or  
anywhere else potentially?.




3) They upgraded to libtool 2.0 which builds binaries that will not  
start. (easier said than done to upgrade ours too as we have to make  
sure we work on *every* plarform and the upgrade to make it work  
would break other operating systems we support)



Understand.

Bottom line, it's not their fault or anything but the choice to use  
all brand new versions of everything under the sun is not a good  
idea for your server, it's great that we have bleeding edge stuff or  
we would not have anyone to test stuff, we have a similar group of  
people always running SVN trunk of the day.  But it's hard to  
stabalize code when both your code and the OS may be unstable at the  
same time.


There is a reason they call it bleeding vs stable, which one would  
you rather be if you were in the hospital. =D




As you mentioned. It's not their fault. ppl want to live on the edge  
just need to install multi-versions of gcc(or other tools). Like the  
Linux kernel, to compile from source, gcc-3 was recommended for a long  
time. Don't know if it's still the case recently.





2009/3/27 dujinfang dujinf...@gmail.com

On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote:
We do not support ubuntu interpid, it has at least 3 known fatal  
issues not experienced by all but nonetheless enough to make us  
unwilling to support it.


I use Ubuntu gutsy in production  and interipid in test env. It  
works well. Can you briefly explain the 3 fatal issues Anthony? It  
will help me know potential risks.





It's use at your own risk or use the stable branch hardy for  
any support.




On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds tre...@concipient.net 
 wrote:

Has there been any progress getting FreeSWITCH to build on Ubuntu
Intrepid without downgrading libtool?



I successfully built FS on intrepid. I simply done this by changing  
the apt-source to Hardy and installed libtool. Obviously I changed  
the apt-source back to intrepid after I installed libtool.


And, another approach. Install libtool from source should be as easy  
as  configure  make  make install. I done this on a new CentOS4  
because the default yum install of libtool on CentOS4 is old than FS  
required.




Thanks!

Sincerely,
Trevor Hammonds

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Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls

2009-03-27 Thread dujinfang



On Mar 28, 2009, at 4:05 AM, Anthony Minessale wrote:

if you prefix the sofia dial string with sip: you should be able to  
pass anything you want.


sofia/internal/sip:9085551212;npdi=yes;rn=9083820...@204.123.123.123:5060



Is that similar as this?

action application=bridge data=sofia/sip/9998881...@sip.yourprovider.com 
/


got it from wiki:  
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#From_the_Dialplan

2009/3/27 James H Thompson j...@lj.net
Calls would be sent to the IP address after the '@'  in the URI.
Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as  
the user part of a SIP URI.
My example Invite URI is the way we are receiving traffic from some  
of the major telecom carriers.

We would like be able to generate calls using the same formats.


- Original Message -
From: Michael Jerris
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, March 27, 2009 5:41 AM
Subject: Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls

You seem to be confusing your standards, those 2 specs are about  
tel: uri's not sip: uris.  Sending a tel uri I am not sure we can  
do, where would we send it to?


Mike

On Mar 27, 2009, at 6:11 AM, James H Thompson wrote:


I need to generate calls with Invite URIs in this format:

INVITE sip:9085551212;npdi=yes;rn=9083820...@204.123.123.123:5060  
SIP/2.0


Is there an easy way to do this?

Thanks.

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Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls