Re: [Freeswitch-users] mod_t38gateway

2009-11-01 Thread Tihomir Culjaga
i tought so :PP

T.

On Sun, Nov 1, 2009 at 6:34 AM, Michael Jerris m...@jerris.com wrote:

 This is a non working module, just a shell for development.

 Mike

 On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote:

  does anybody know how does it work and how to use it in a dialplan?
 


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Re: [Freeswitch-users] SIP provider with extern rtp server

2009-11-01 Thread Ivan C Myrvold
No one have any idea why this is not working? I have combed through  
the log, but couldn't find any clue there.
Incoming calls from my sip provider is working perfect, but for  
outgoing calls it looks like Freeswitch is not letting the incoming  
rtp to the local sip phone.

Ivan

On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote:

 Yes, now I got a more detailed trace. Thank you for helping me with
 this.

 A new pastebin at http://pastebin.freeswitch.org/10905

 Ivan

 Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:

 fsctl loglevel debug
 console loglevel debug
 sofia profile internal siptrace on
 sofia profile external siptrace on
 sofia loglevel all 9
 ^

 Then run your call, then do this:

 sofia loglevel all 0
 sofia profile external siptrace off
 sofia profile internal siptrace off
 fsctl loglevel warning
 console loglevel warning

 On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold i...@myrvold.org
 wrote:
 I have already set debug to 9, on both profiles.

 Ivan


 Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:

 See that 200 OK that keeps coming in over and over and over and  
 over
 again? That's because they never received your ACK. If you can
 turn on
 sofia loglevel to 9 and then watch where you send the ACK, you will
 probably have your answer to why the other system did not receive
 it.
 If you're still not sure what's going on, post another pastebin  
 with
 sofia loglevel set to 9.


 On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold i...@myrvold.org
 wrote:
 Oh, what happened to it?
 Anyway, here is a new pb:
 http://pastebin.freeswitch.org/10867
 Ivan
 Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:


 On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org
 wrote:

 Here is a debug log from a call from an internal phone out to an
 external (my iPhone with nbr 91316356):
 http://pastebin.freeswitch.org/108578

 Ivan

 Uh... you wanna try that PB number again?
 -MC

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 --
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 We do not inherit the Earth from our ancestors: we borrow it from
 our
 children. ~David Brower

 I decided the words were too conservative for me. We're not
 borrowing
 from our children, we're stealing from them--and it's not even
 considered to be a crime. ~David Brower

 Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to
 live; not live to eat.) ~Marcus Tullius Cicero

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 -- 
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 children. ~David Brower

 I decided the words were too conservative for me. We're not  
 borrowing
 from our children, we're stealing from them--and it's not even
 considered to be a crime. ~David Brower

 Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to
 live; not live to eat.) ~Marcus Tullius Cicero

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[Freeswitch-users] Rejecting a call from JavaScript

2009-11-01 Thread Yehavi Bourvine
Hello,

  We would like to handle an incoming call to a busy phone according
to user's prefference:  Some want waiting call, some want to just reject the
call, and others want to send the call to voicemail.

  We have a small JavaScript which tests the status of the destination and
the user's will and tries to act accordingly. Our problem is how to send
busy. I tried session.hangup(USER_BUSY) but it always sends temporary
unavailable which causes the orignator to think that the destination is out
of order.

What is the correct way to do so?

   Thanks! __Yehavi:
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Re: [Freeswitch-users] Mod_pjsip

2009-11-01 Thread Diego Viola
Don't put me on the same leval as DelphiWorld please, I was just
curious about how this SIP stack compares to sofia.

Diego

On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns ch...@cloudtel.com wrote:
 My favorite part of this 'civilized' discussion on IRC was when DelphiWord and
 diegoviola sat around tryin to take the piss outta stkn on this issue for
 seemingly no reason. Thanks for making the channel a cool place, guys ;)

 On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:
 Anthony Minessale a écrit :
  Meftah,
  Feel free.
 
  thanks
 
  P.S.
 
  STKN was the guy who made the first mod_pjsip for FS that we abandoned
  years ago. So you should believe him.
  Both him and I agreed it was not working out.  So if you don't believe
  me, find out for yourself.
  anthony, why i don't believe  you?

 never say that.
 i believe you and all Freeswitch Staf and thank you and to all
 Freeswitch Staf.

  On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com
  mailto:tayeb.mef...@gmail.com wrote:
 
      hi Anthony
      i agry
      i say that because STKN hate all my suggestions.
      about pjsip, i will contribute aditional module in the contrib.
      thanks Anthony
 
      Anthony Minessale a écrit :
      Meftah,
 
      He is 100% correct.  Please do not insult my volunteer
      developers. Without help from him you would not have any
      FreeSWITCH right now so please drop this subject we are not using
      pjsip.
 
 
 
      On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
      tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:
 
          hi,
          Pjsip support ICE, STUN and TURN!
          to STKN:
          if you don't pjsip, please stop talking or exit the discution
          we want to kype Freeswitch Clean and universal
 
          Stefan Knoblich a écrit :
          Michael S Collins wrote:
          I can guarantee that the FS devs are well aware of pj-sip. If
  it was/ is a viable alternative then it would be considered. The fact
  that it isn't being used is a pretty good indication that it isn't
  suitable for FS at this time.
 
          -MV
 
          Sent from my iPhone
 
          We already mentioned some of the reasons why it did get
          dropped 3 years ago (first two points from memory, last two
          from old IRC logs): [License incompatible (GPL), but i think
          tony tried to negotiate on alternate license terms] Not
          possible to have multiple SIP profiles (due to global
          variables being used in the lib). A race-condition under
          high load, that couldn't be resolved back then (with the
          help of the pjsip developers). And the sofia module just
          working and surviving the scalability tests, so all efforts
          were focussed on mod_sofia and pjsip got dropped. stkn
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[Freeswitch-users] multihomed help

2009-11-01 Thread Joseph L. Casale
I am setting up fs in pfsense.
Following the multihomed tutorial (I also have a dedicated wan/lan int)
if I set directory/default.xml domain=10.0.0.1 (my lan int ip) it breaks
everything, but if I set X-PRE-PROCESS cmd=set data=domain=10.0.0.1/
in conf/vars.xml I now get audio working correctly, * etc plays MOH.

Is there still something I have done wrong?


conf/sip_profiles/internal.xml has the int lan ip's entered, and external.xml
does not have any edits.

Thanks for any info!
jlc

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[Freeswitch-users] Small bug in switch_ivr_record_file (in trunk)

2009-11-01 Thread Jonas Gauffin
switch_ivr_play_say.c, line 486.

file = switch_core_session_sprintf(session, %s%s%s%s,
switch_str_nil(tfile), tfile ? ] : , prefix, SWITCH_PATH_SEPARATOR,
file);

There should be five %s, not four.
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Re: [Freeswitch-users] Small bug in switch_ivr_record_file (in trunk)

2009-11-01 Thread Jonas Gauffin
Same bug in switch_ivr_async.c, method switch_ivr_record_session.

On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote:

 switch_ivr_play_say.c, line 486.

 file = switch_core_session_sprintf(session, %s%s%s%s,
 switch_str_nil(tfile), tfile ? ] : , prefix, SWITCH_PATH_SEPARATOR,
 file);

 There should be five %s, not four.

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Re: [Freeswitch-users] SIP Proxy with direct media path

2009-11-01 Thread Dmitry Gromov
Hi!

On Sun, Nov 1, 2009 at 01:21, Jason White ja...@jasonjgw.net wrote:


 http://wiki.freeswitch.org/ is the best there is.

 It is being improved by the community over time.

 You can also take advantage of the IRC channel, the FreeSWITCH conference
 and
 of course the mailing list.



Thank you very much for the reply... I had some experience with FreeSwitch
about a year ago, but did not have more time to check all the features. And
there wasn't much documentation available then.
I'll give it another try this time.

Thanks,
Dmitry

-- 
DG
NJ
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Re: [Freeswitch-users] Mod_pjsip

2009-11-01 Thread Giovanni Maruzzelli
On Sun, Nov 1, 2009 at 3:37 PM, Diego Viola diego.vi...@gmail.com wrote:
 Don't put me on the same leval as DelphiWorld please, I was just
 curious about how this SIP stack compares to sofia.

Smile Diego, smile. We're all just jocking!  :)

-gm



 Diego

 On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns ch...@cloudtel.com wrote:
 My favorite part of this 'civilized' discussion on IRC was when DelphiWord 
 and
 diegoviola sat around tryin to take the piss outta stkn on this issue for
 seemingly no reason. Thanks for making the channel a cool place, guys ;)

 On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:
 Anthony Minessale a écrit :
  Meftah,
  Feel free.
 
  thanks
 
  P.S.
 
  STKN was the guy who made the first mod_pjsip for FS that we abandoned
  years ago. So you should believe him.
  Both him and I agreed it was not working out.  So if you don't believe
  me, find out for yourself.
  anthony, why i don't believe  you?

 never say that.
 i believe you and all Freeswitch Staf and thank you and to all
 Freeswitch Staf.

  On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com
  mailto:tayeb.mef...@gmail.com wrote:
 
      hi Anthony
      i agry
      i say that because STKN hate all my suggestions.
      about pjsip, i will contribute aditional module in the contrib.
      thanks Anthony
 
      Anthony Minessale a écrit :
      Meftah,
 
      He is 100% correct.  Please do not insult my volunteer
      developers. Without help from him you would not have any
      FreeSWITCH right now so please drop this subject we are not using
      pjsip.
 
 
 
      On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
      tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:
 
          hi,
          Pjsip support ICE, STUN and TURN!
          to STKN:
          if you don't pjsip, please stop talking or exit the discution
          we want to kype Freeswitch Clean and universal
 
          Stefan Knoblich a écrit :
          Michael S Collins wrote:
          I can guarantee that the FS devs are well aware of pj-sip. If
  it was/ is a viable alternative then it would be considered. The fact
  that it isn't being used is a pretty good indication that it isn't
  suitable for FS at this time.
 
          -MV
 
          Sent from my iPhone
 
          We already mentioned some of the reasons why it did get
          dropped 3 years ago (first two points from memory, last two
          from old IRC logs): [License incompatible (GPL), but i think
          tony tried to negotiate on alternate license terms] Not
          possible to have multiple SIP profiles (due to global
          variables being used in the lib). A race-condition under
          high load, that couldn't be resolved back then (with the
          help of the pjsip developers). And the sofia module just
          working and surviving the scalability tests, so all efforts
          were focussed on mod_sofia and pjsip got dropped. stkn
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      Twitter: http://twitter.com/FreeSWITCH_wire
 
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      sip:8...@conference.freeswitch.org
      mailto:sip%3a...@conference.freeswitch.org
      iax:gu...@conference.freeswitch.org/888
      http://iax:gu...@conference.freeswitch.org/888
      googletalk:conf+...@conference.freeswitch.org
      mailto:googletalk%3aconf%2b...@conference.freeswitch.org
      pstn:213-799-1400
 
   

Re: [Freeswitch-users] Mod_pjsip

2009-11-01 Thread Meftah Tayeb

hi diego,
what you mean?
so my level is nothing?
my level is bad?
my level is zero?
thank to gmaruzz/MikeJ that understand me quickly/easyly
Diego Viola a écrit :

Don't put me on the same leval as DelphiWorld please, I was just
curious about how this SIP stack compares to sofia.

Diego

On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns ch...@cloudtel.com wrote:
  

My favorite part of this 'civilized' discussion on IRC was when DelphiWord and
diegoviola sat around tryin to take the piss outta stkn on this issue for
seemingly no reason. Thanks for making the channel a cool place, guys ;)

On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:


Anthony Minessale a écrit :
  

Meftah,
Feel free.

thanks

P.S.

STKN was the guy who made the first mod_pjsip for FS that we abandoned
years ago. So you should believe him.
Both him and I agreed it was not working out.  So if you don't believe
me, find out for yourself.
anthony, why i don't believe  you?


never say that.
i believe you and all Freeswitch Staf and thank you and to all
Freeswitch Staf.

  

On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com
mailto:tayeb.mef...@gmail.com wrote:

hi Anthony
i agry
i say that because STKN hate all my suggestions.
about pjsip, i will contribute aditional module in the contrib.
thanks Anthony

Anthony Minessale a écrit :


Meftah,

He is 100% correct.  Please do not insult my volunteer
developers. Without help from him you would not have any
FreeSWITCH right now so please drop this subject we are not using
pjsip.



On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:

hi,
Pjsip support ICE, STUN and TURN!
to STKN:
if you don't pjsip, please stop talking or exit the discution
we want to kype Freeswitch Clean and universal

Stefan Knoblich a écrit :
  

Michael S Collins wrote:


I can guarantee that the FS devs are well aware of pj-sip. If
it was/ is a viable alternative then it would be considered. The fact
that it isn't being used is a pretty good indication that it isn't
suitable for FS at this time.

-MV

Sent from my iPhone
  

We already mentioned some of the reasons why it did get
dropped 3 years ago (first two points from memory, last two
from old IRC logs): [License incompatible (GPL), but i think
tony tried to negotiate on alternate license terms] Not
possible to have multiple SIP profiles (due to global
variables being used in the lib). A race-condition under
high load, that couldn't be resolved back then (with the
help of the pjsip developers). And the sofia module just
working and surviving the scalability tests, so all efforts
were focussed on mod_sofia and pjsip got dropped. stkn
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Re: [Freeswitch-users] Mod_pjsip

2009-11-01 Thread Diego Viola
=D

On Sun, Nov 1, 2009 at 4:04 PM, Giovanni Maruzzelli gmar...@celliax.org wrote:
 On Sun, Nov 1, 2009 at 3:37 PM, Diego Viola diego.vi...@gmail.com wrote:
 Don't put me on the same leval as DelphiWorld please, I was just
 curious about how this SIP stack compares to sofia.

 Smile Diego, smile. We're all just jocking!  :)

 -gm



 Diego

 On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns ch...@cloudtel.com wrote:
 My favorite part of this 'civilized' discussion on IRC was when DelphiWord 
 and
 diegoviola sat around tryin to take the piss outta stkn on this issue for
 seemingly no reason. Thanks for making the channel a cool place, guys ;)

 On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:
 Anthony Minessale a écrit :
  Meftah,
  Feel free.
 
  thanks
 
  P.S.
 
  STKN was the guy who made the first mod_pjsip for FS that we abandoned
  years ago. So you should believe him.
  Both him and I agreed it was not working out.  So if you don't believe
  me, find out for yourself.
  anthony, why i don't believe  you?

 never say that.
 i believe you and all Freeswitch Staf and thank you and to all
 Freeswitch Staf.

  On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com
  mailto:tayeb.mef...@gmail.com wrote:
 
      hi Anthony
      i agry
      i say that because STKN hate all my suggestions.
      about pjsip, i will contribute aditional module in the contrib.
      thanks Anthony
 
      Anthony Minessale a écrit :
      Meftah,
 
      He is 100% correct.  Please do not insult my volunteer
      developers. Without help from him you would not have any
      FreeSWITCH right now so please drop this subject we are not using
      pjsip.
 
 
 
      On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
      tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:
 
          hi,
          Pjsip support ICE, STUN and TURN!
          to STKN:
          if you don't pjsip, please stop talking or exit the discution
          we want to kype Freeswitch Clean and universal
 
          Stefan Knoblich a écrit :
          Michael S Collins wrote:
          I can guarantee that the FS devs are well aware of pj-sip. If
  it was/ is a viable alternative then it would be considered. The fact
  that it isn't being used is a pretty good indication that it isn't
  suitable for FS at this time.
 
          -MV
 
          Sent from my iPhone
 
          We already mentioned some of the reasons why it did get
          dropped 3 years ago (first two points from memory, last two
          from old IRC logs): [License incompatible (GPL), but i think
          tony tried to negotiate on alternate license terms] Not
          possible to have multiple SIP profiles (due to global
          variables being used in the lib). A race-condition under
          high load, that couldn't be resolved back then (with the
          help of the pjsip developers). And the sofia module just
          working and surviving the scalability tests, so all efforts
          were focussed on mod_sofia and pjsip got dropped. stkn
          ___
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      --
      Anthony Minessale II
 
      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire
 
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Re: [Freeswitch-users] Mod_pjsip

2009-11-01 Thread Diego Viola
Hi Meftah,

No, of course is not, and it will never be, I actually quite admire
how you are able to do what you do.

I just wanted to say that I don't want people to put me into this
problem, because I don't have anything to do with it, I was just
curious about how pjsip compares to sofia, etc. That's all.

Apologies if I didn't expressed myself correctly.

Regards,

Diego

On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb tayeb.mef...@gmail.com wrote:
 hi diego,
 what you mean?
 so my level is nothing?
 my level is bad?
 my level is zero?
 thank to gmaruzz/MikeJ that understand me quickly/easyly
 Diego Viola a écrit :

 Don't put me on the same leval as DelphiWorld please, I was just
 curious about how this SIP stack compares to sofia.

 Diego

 On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns ch...@cloudtel.com wrote:


 My favorite part of this 'civilized' discussion on IRC was when DelphiWord
 and
 diegoviola sat around tryin to take the piss outta stkn on this issue for
 seemingly no reason. Thanks for making the channel a cool place, guys ;)

 On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:


 Anthony Minessale a écrit :


 Meftah,
 Feel free.

 thanks

 P.S.

 STKN was the guy who made the first mod_pjsip for FS that we abandoned
 years ago. So you should believe him.
 Both him and I agreed it was not working out.  So if you don't believe
 me, find out for yourself.
 anthony, why i don't believe  you?


 never say that.
 i believe you and all Freeswitch Staf and thank you and to all
 Freeswitch Staf.



 On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com
 mailto:tayeb.mef...@gmail.com wrote:

     hi Anthony
     i agry
     i say that because STKN hate all my suggestions.
     about pjsip, i will contribute aditional module in the contrib.
     thanks Anthony

     Anthony Minessale a écrit :


     Meftah,

     He is 100% correct.  Please do not insult my volunteer
     developers. Without help from him you would not have any
     FreeSWITCH right now so please drop this subject we are not using
     pjsip.



     On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
     tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:

         hi,
         Pjsip support ICE, STUN and TURN!
         to STKN:
         if you don't pjsip, please stop talking or exit the discution
         we want to kype Freeswitch Clean and universal

         Stefan Knoblich a écrit :


         Michael S Collins wrote:


         I can guarantee that the FS devs are well aware of pj-sip. If
 it was/ is a viable alternative then it would be considered. The fact
 that it isn't being used is a pretty good indication that it isn't
 suitable for FS at this time.

         -MV

         Sent from my iPhone


         We already mentioned some of the reasons why it did get
         dropped 3 years ago (first two points from memory, last two
         from old IRC logs): [License incompatible (GPL), but i think
         tony tried to negotiate on alternate license terms] Not
         possible to have multiple SIP profiles (due to global
         variables being used in the lib). A race-condition under
         high load, that couldn't be resolved back then (with the
         help of the pjsip developers). And the sofia module just
         working and surviving the scalability tests, so all efforts
         were focussed on mod_sofia and pjsip got dropped. stkn
         ___
         FreeSWITCH-users mailing list
         FreeSWITCH-users@lists.freeswitch.org
         mailto:FreeSWITCH-users@lists.freeswitch.org
         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
 s http://www.freeswitch.org



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     --
     Anthony Minessale II

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

     AIM: anthm
     MSN:anthony_miness...@hotmail.com
     mailto:msn%3aanthony_miness...@hotmail.com
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Re: [Freeswitch-users] Mod_pjsip

2009-11-01 Thread Diego Viola
I never said your level is bad or anything, I just said that I don't
want people to involve me into that problem.

Diego

On Sun, Nov 1, 2009 at 5:09 PM, Diego Viola diego.vi...@gmail.com wrote:
 Hi Meftah,

 No, of course is not, and it will never be, I actually quite admire
 how you are able to do what you do.

 I just wanted to say that I don't want people to put me into this
 problem, because I don't have anything to do with it, I was just
 curious about how pjsip compares to sofia, etc. That's all.

 Apologies if I didn't expressed myself correctly.

 Regards,

 Diego

 On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb tayeb.mef...@gmail.com wrote:
 hi diego,
 what you mean?
 so my level is nothing?
 my level is bad?
 my level is zero?
 thank to gmaruzz/MikeJ that understand me quickly/easyly
 Diego Viola a écrit :

 Don't put me on the same leval as DelphiWorld please, I was just
 curious about how this SIP stack compares to sofia.

 Diego

 On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns ch...@cloudtel.com wrote:


 My favorite part of this 'civilized' discussion on IRC was when DelphiWord
 and
 diegoviola sat around tryin to take the piss outta stkn on this issue for
 seemingly no reason. Thanks for making the channel a cool place, guys ;)

 On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:


 Anthony Minessale a écrit :


 Meftah,
 Feel free.

 thanks

 P.S.

 STKN was the guy who made the first mod_pjsip for FS that we abandoned
 years ago. So you should believe him.
 Both him and I agreed it was not working out.  So if you don't believe
 me, find out for yourself.
 anthony, why i don't believe  you?


 never say that.
 i believe you and all Freeswitch Staf and thank you and to all
 Freeswitch Staf.



 On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com
 mailto:tayeb.mef...@gmail.com wrote:

     hi Anthony
     i agry
     i say that because STKN hate all my suggestions.
     about pjsip, i will contribute aditional module in the contrib.
     thanks Anthony

     Anthony Minessale a écrit :


     Meftah,

     He is 100% correct.  Please do not insult my volunteer
     developers. Without help from him you would not have any
     FreeSWITCH right now so please drop this subject we are not using
     pjsip.



     On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
     tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:

         hi,
         Pjsip support ICE, STUN and TURN!
         to STKN:
         if you don't pjsip, please stop talking or exit the discution
         we want to kype Freeswitch Clean and universal

         Stefan Knoblich a écrit :


         Michael S Collins wrote:


         I can guarantee that the FS devs are well aware of pj-sip. If
 it was/ is a viable alternative then it would be considered. The fact
 that it isn't being used is a pretty good indication that it isn't
 suitable for FS at this time.

         -MV

         Sent from my iPhone


         We already mentioned some of the reasons why it did get
         dropped 3 years ago (first two points from memory, last two
         from old IRC logs): [License incompatible (GPL), but i think
         tony tried to negotiate on alternate license terms] Not
         possible to have multiple SIP profiles (due to global
         variables being used in the lib). A race-condition under
         high load, that couldn't be resolved back then (with the
         help of the pjsip developers). And the sofia module just
         working and surviving the scalability tests, so all efforts
         were focussed on mod_sofia and pjsip got dropped. stkn
         ___
         FreeSWITCH-users mailing list
         FreeSWITCH-users@lists.freeswitch.org
         mailto:FreeSWITCH-users@lists.freeswitch.org
         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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     --
     Anthony Minessale II

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

     AIM: anthm
     

Re: [Freeswitch-users] Small bug in switch_ivr_record_file (in trunk)

2009-11-01 Thread Anthony Minessale
thank you
fixed in r15308
We do prefer jira for this kind of thing so we can track and make accurate
change logs.


On Sun, Nov 1, 2009 at 9:14 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote:

 Same bug in switch_ivr_async.c, method switch_ivr_record_session.


 On Sun, Nov 1, 2009 at 4:06 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote:

 switch_ivr_play_say.c, line 486.

 file = switch_core_session_sprintf(session, %s%s%s%s,
 switch_str_nil(tfile), tfile ? ] : , prefix, SWITCH_PATH_SEPARATOR,
 file);

 There should be five %s, not four.



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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
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Re: [Freeswitch-users] Mod_pjsip

2009-11-01 Thread Anthony Minessale
Meftah and Diego,

If you continue to argue on our mailing lists I will be forced to moderate
you to stop you from bothering other people.
I do not want to see one more reply to this thread from either of you.
Please do not reply to apologize, simply stop sending any more email to this
topic.


On Sun, Nov 1, 2009 at 11:25 AM, Diego Viola diego.vi...@gmail.com wrote:

 I never said your level is bad or anything, I just said that I don't
 want people to involve me into that problem.

 Diego

 On Sun, Nov 1, 2009 at 5:09 PM, Diego Viola diego.vi...@gmail.com wrote:
  Hi Meftah,
 
  No, of course is not, and it will never be, I actually quite admire
  how you are able to do what you do.
 
  I just wanted to say that I don't want people to put me into this
  problem, because I don't have anything to do with it, I was just
  curious about how pjsip compares to sofia, etc. That's all.
 
  Apologies if I didn't expressed myself correctly.
 
  Regards,
 
  Diego
 
  On Sun, Nov 1, 2009 at 5:46 PM, Meftah Tayeb tayeb.mef...@gmail.com
 wrote:
  hi diego,
  what you mean?
  so my level is nothing?
  my level is bad?
  my level is zero?
  thank to gmaruzz/MikeJ that understand me quickly/easyly
  Diego Viola a écrit :
 
  Don't put me on the same leval as DelphiWorld please, I was just
  curious about how this SIP stack compares to sofia.
 
  Diego
 
  On Sun, Nov 1, 2009 at 2:24 AM, Chris Burns ch...@cloudtel.com wrote:
 
 
  My favorite part of this 'civilized' discussion on IRC was when
 DelphiWord
  and
  diegoviola sat around tryin to take the piss outta stkn on this issue
 for
  seemingly no reason. Thanks for making the channel a cool place, guys ;)
 
  On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:
 
 
  Anthony Minessale a écrit :
 
 
  Meftah,
  Feel free.
 
  thanks
 
  P.S.
 
  STKN was the guy who made the first mod_pjsip for FS that we abandoned
  years ago. So you should believe him.
  Both him and I agreed it was not working out.  So if you don't believe
  me, find out for yourself.
  anthony, why i don't believe  you?
 
 
  never say that.
  i believe you and all Freeswitch Staf and thank you and to all
  Freeswitch Staf.
 
 
 
  On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com
  mailto:tayeb.mef...@gmail.com wrote:
 
  hi Anthony
  i agry
  i say that because STKN hate all my suggestions.
  about pjsip, i will contribute aditional module in the contrib.
  thanks Anthony
 
  Anthony Minessale a écrit :
 
 
  Meftah,
 
  He is 100% correct.  Please do not insult my volunteer
  developers. Without help from him you would not have any
  FreeSWITCH right now so please drop this subject we are not using
  pjsip.
 
 
 
  On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb
  tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote:
 
  hi,
  Pjsip support ICE, STUN and TURN!
  to STKN:
  if you don't pjsip, please stop talking or exit the discution
  we want to kype Freeswitch Clean and universal
 
  Stefan Knoblich a écrit :
 
 
  Michael S Collins wrote:
 
 
  I can guarantee that the FS devs are well aware of pj-sip. If
  it was/ is a viable alternative then it would be considered. The fact
  that it isn't being used is a pretty good indication that it isn't
  suitable for FS at this time.
 
  -MV
 
  Sent from my iPhone
 
 
  We already mentioned some of the reasons why it did get
  dropped 3 years ago (first two points from memory, last two
  from old IRC logs): [License incompatible (GPL), but i think
  tony tried to negotiate on alternate license terms] Not
  possible to have multiple SIP profiles (due to global
  variables being used in the lib). A race-condition under
  high load, that couldn't be resolved back then (with the
  help of the pjsip developers). And the sofia module just
  working and surviving the scalability tests, so all efforts
  were focussed on mod_sofia and pjsip got dropped. stkn
  ___
  FreeSWITCH-users mailing list
  FreeSWITCH-users@lists.freeswitch.org
  mailto:FreeSWITCH-users@lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-11-01 Thread Anthony Minessale
#4 makes no sense to me.
Are you just trying to create a call that the channel does not participate
in?
Then for sure you want to use bgapi to cause the originate to  happen in the
background.

Also your gcore report has to be taken while you have the stuck channels to
see why they are stuck.
I can promise you this problem should be filed under misuse/abuse.


On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost d...@tel.co.th wrote:

 How to use bgapi in my flow.
 1. user call did
 2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey)
 3. Hangup (by use or timeout)
 4. FS callback to user and bridge to IVR

 I'm not sure  bgapi can do after channel hangup.

 Dome C.


 2009/11/1 Anthony Minessale anthony.miness...@gmail.com:
  Use bgapi originate ...
 
  On Oct 31, 2009 9:09 PM, Dome Charoenyost d...@tel.co.th wrote:
 
  Yes. i user api_hangup_hook for do callback.
  action application=set data=api_hangup_hook=originate
 
 {origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
  call_ivr${caller_id_number} XML public /
 
  may be need originate_timeout
 
  Dome C.
 
 
  2009/11/1 Rupa Schomaker r...@rupa.com:
 
  fscore_pb has been updated.   Next time put in the pastebin url.  This
  one was 10911.   Ok, a ...
 
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Re: [Freeswitch-users] Rejecting a call from JavaScript

2009-11-01 Thread Anthony Minessale
try session.execute(hangup, user_busy);


On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine
yehavi.bourv...@gmail.comwrote:

 Hello,

   We would like to handle an incoming call to a busy phone according
 to user's prefference:  Some want waiting call, some want to just reject the
 call, and others want to send the call to voicemail.

   We have a small JavaScript which tests the status of the destination and
 the user's will and tries to act accordingly. Our problem is how to send
 busy. I tried session.hangup(USER_BUSY) but it always sends temporary
 unavailable which causes the orignator to think that the destination is out
 of order.

 What is the correct way to do so?

Thanks! __Yehavi:

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Re: [Freeswitch-users] CDR CSV variables

2009-11-01 Thread Anthony Minessale
you can make up your own variable and set whatever you want in there then
add it to the template.


On Fri, Oct 30, 2009 at 10:04 AM, DJB djbin...@yahoo.com wrote:

 I wonder if I don't want to have b-leg in cdr csv, is there any variables
 that can give me the actual gateway ip address that is actually went out.

 For instance, if I have this in my dialplan:
 action application=bridge data=sofia/default/$...@192.168.1.4
 |sofia/default/$...@192.168.1.5/
 the only value that I can think of from cdr csv is to get
 remote_ip_last_arg, but it would contains the whole line of both ip
 addresses.

 Thank you,
 Dorn B.





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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-11-01 Thread Dome Charoenyost
2009/11/2 Anthony Minessale anthony.miness...@gmail.com:
 #4 makes no sense to me.
This solution call miss callback. some contry cost for toll free
number is so high. but call to user cheaper. so when user want to
use international call. they call to my DID and my system callback to them.

 Are you just trying to create a call that the channel does not participate
 in?
 Then for sure you want to use bgapi to cause the originate to  happen in the
 background.
Give me example how bgapi work when user hangup call



Dome C.


 Also your gcore report has to be taken while you have the stuck channels to
 see why they are stuck.
 I can promise you this problem should be filed under misuse/abuse.


 On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost d...@tel.co.th wrote:

 How to use bgapi in my flow.
 1. user call did
 2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey)
 3. Hangup (by use or timeout)
 4. FS callback to user and bridge to IVR

 I'm not sure  bgapi can do after channel hangup.

 Dome C.


 2009/11/1 Anthony Minessale anthony.miness...@gmail.com:
  Use bgapi originate ...
 
  On Oct 31, 2009 9:09 PM, Dome Charoenyost d...@tel.co.th wrote:
 
  Yes. i user api_hangup_hook for do callback.
  action application=set data=api_hangup_hook=originate
 
  {origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
  call_ivr${caller_id_number} XML public /
 
  may be need originate_timeout
 
  Dome C.
 
 
  2009/11/1 Rupa Schomaker r...@rupa.com:
 
  fscore_pb has been updated.   Next time put in the pastebin url.
   This
  one was 10911.   Ok, a ...
 
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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

2009-11-01 Thread Anthony Minessale
if you set the variable to:

originate foo bar baz

bgapi originate foo bar baz

bgapi is itself an api command that takes the argument string and runs it in
a separate thread and returns instantly.
This will stop it from hanging at that point.




On Sun, Nov 1, 2009 at 1:53 PM, Dome Charoenyost d...@tel.co.th wrote:

 2009/11/2 Anthony Minessale anthony.miness...@gmail.com:
  #4 makes no sense to me.
 This solution call miss callback. some contry cost for toll free
 number is so high. but call to user cheaper. so when user want to
 use international call. they call to my DID and my system callback to them.

  Are you just trying to create a call that the channel does not
 participate
  in?
  Then for sure you want to use bgapi to cause the originate to  happen in
 the
  background.
 Give me example how bgapi work when user hangup call



 Dome C.

 
  Also your gcore report has to be taken while you have the stuck channels
 to
  see why they are stuck.
  I can promise you this problem should be filed under misuse/abuse.
 
 
  On Sat, Oct 31, 2009 at 8:45 PM, Dome Charoenyost d...@tel.co.th
 wrote:
 
  How to use bgapi in my flow.
  1. user call did
  2. FS send ringing and check balance, LCR from DB (by mod_odbc_quey)
  3. Hangup (by use or timeout)
  4. FS callback to user and bridge to IVR
 
  I'm not sure  bgapi can do after channel hangup.
 
  Dome C.
 
 
  2009/11/1 Anthony Minessale anthony.miness...@gmail.com:
   Use bgapi originate ...
  
   On Oct 31, 2009 9:09 PM, Dome Charoenyost d...@tel.co.th wrote:
  
   Yes. i user api_hangup_hook for do callback.
   action application=set data=api_hangup_hook=originate
  
  
 {origination_caller_id_number=${caller_id_number},callback_uuid=${uuid},ignore_early_media=true}${route_1}|${route_2}|${route_3}
   call_ivr${caller_id_number} XML public /
  
   may be need originate_timeout
  
   Dome C.
  
  
   2009/11/1 Rupa Schomaker r...@rupa.com:
  
   fscore_pb has been updated.   Next time put in the pastebin url.
This
   one was 10911.   Ok, a ...
  
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Re: [Freeswitch-users] SIP provider with extern rtp server

2009-11-01 Thread Anthony Minessale
Session-Expires: -1;refresher=uas

nta: 200 OK has fatal syntax errors

This is a know-bug in asterisk.

see: https://issues.asterisk.org/view.php?id=15621



On Sun, Nov 1, 2009 at 4:40 AM, Ivan C Myrvold i...@myrvold.org wrote:

 No one have any idea why this is not working? I have combed through
 the log, but couldn't find any clue there.
 Incoming calls from my sip provider is working perfect, but for
 outgoing calls it looks like Freeswitch is not letting the incoming
 rtp to the local sip phone.

 Ivan

 On 30. okt. 2009, at 21:26, Ivan C Myrvold wrote:

  Yes, now I got a more detailed trace. Thank you for helping me with
  this.
 
  A new pastebin at http://pastebin.freeswitch.org/10905
 
  Ivan
 
  Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:
 
  fsctl loglevel debug
  console loglevel debug
  sofia profile internal siptrace on
  sofia profile external siptrace on
  sofia loglevel all 9
  ^
 
  Then run your call, then do this:
 
  sofia loglevel all 0
  sofia profile external siptrace off
  sofia profile internal siptrace off
  fsctl loglevel warning
  console loglevel warning
 
  On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold i...@myrvold.org
  wrote:
  I have already set debug to 9, on both profiles.
 
  Ivan
 
 
  Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:
 
  See that 200 OK that keeps coming in over and over and over and
  over
  again? That's because they never received your ACK. If you can
  turn on
  sofia loglevel to 9 and then watch where you send the ACK, you will
  probably have your answer to why the other system did not receive
  it.
  If you're still not sure what's going on, post another pastebin
  with
  sofia loglevel set to 9.
 
 
  On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold i...@myrvold.org
  wrote:
  Oh, what happened to it?
  Anyway, here is a new pb:
  http://pastebin.freeswitch.org/10867
  Ivan
  Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
 
 
  On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org
  wrote:
 
  Here is a debug log from a call from an internal phone out to an
  external (my iPhone with nbr 91316356):
  http://pastebin.freeswitch.org/108578
 
  Ivan
 
  Uh... you wanna try that PB number again?
  -MC
 
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[Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session

2009-11-01 Thread Seven Du
Just suspicious would be possible that happened on sqlite stage? I
manually deleted the channels from sqlite and nothing bad happend.
just FYI.

-- Forwarded message --
From: Seven Du dujinf...@gmail.com
Date: Sun, 1 Nov 2009 10:24:32 +0800
Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session
To:

Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even
unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound
and outbound. I must be very careful if I upgrade to trunk and turn on
rwlock debug because it's on production and the the problem not happening
that much so would be hard to trace. But I will find time to test and report
back.

FYI, I also noticed that in some zombile channels, couple of INVITEs sent
but never got a response. However, it sent out an ACK at last.

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[Freeswitch-users] Get error 415 Unsupported Media Type when receiving call from softswitch

2009-11-01 Thread Lei Tang
Hi all, I get a  415 Unsupported Media Type when FS receiving call from a
softswitch. I captured some packets, It seems that the softswitch use SIP-I
protocol, does FS can handle SIP-I message?

===here is the invite messagefrom softswitch
INVITE sip:xx...@:5060;user=phone SIP/2.0
Contact: sip:x...@:5061
MIME-version: 1.0
Content-Type: multipart/mixed;boundary=Alcatel-boundary
To: sip:x...@:5060;user=phone
From: sip:x...@x:5061;user=phone;tag=73D332463135364195291201
P-Asserted-Identity: sip:x...@x:5061;user=phone
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE
Supported: 100rel,timer,replaces,diversion
Expires: 155
Session-Expires: 1800
Min-SE: 90
Call-ID: 01fd0348728140023...@sip-3
Max-Forwards: 70
CSeq: 1 INVITE
Timestamp: 10645
Via: SIP/2.0/UDP x:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E
Content-Length: 542

--Alcatel-boundary
Content-Type: application/sdp

v=0
o=- 5 8 IN IP4 xx
s=SDP Data
c=IN IP4 x
t=0 0
m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 TELEPHONE-EVENT/8000
a=fmtp:110 mode=3
a=rtpmap:111 speex/8000/1
a=rtpmap:3 GSM/8000
a=ptime:30
a=fmtp:111 mode=2
a=rtpmap:112 speex/8000/1

--Alcatel-boundary
Content-Type: application/ISUP;version=N/A

.. 
...0.D...
...V..0..1...n..p...
--Alcatel-boundary--

=response message from FS
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP x:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E
From: x sip:xx...@:5061;user=phone;tag=73D332463135364195291201
To: sip:xx...@x:5060;user=phone;tag=m6XS44BKF4pSS
Call-ID: 01fd0348728140023...@sip-3
CSeq: 1 INVITE
Timestamp: 10645 0.00
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
Accept: application/sdp
Accept-Encoding:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Length: 0
-- 
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Re: [Freeswitch-users] Get error 415 Unsupported Media Type when receiving call from softswitch

2009-11-01 Thread Lei Tang
FYI, Here is the log when I set sofia loglevel all 9
==
tport_wakeup_pri(00DFE3E8): events IN
tport_recv_event(00DFE3E8)
tport(00DFE3E8) msg 01B2E0C0 from (udp/MyIP:5060) has 1315 bytes, veclen
= 1
tport(00DFE3E8): msg 01B2E0C0 (1315 bytes) from udp/SSIP:5060/sip next=000
0
nta: received INVITE sip:a...@myip:5060;user=phone SIP/2.0 (CSeq
1)
nta: canonizing sip:a...@myip:5060 with contact
nta: INVITE (1) going to a default leg
nta: timer shortened to 200 ms
tport_tsend(00DFE3E8) tpn = UDP/SSIP:5061
tport_resolve addrinfo = SSIP:5061
tport(00DFE3E8): not found by name UDP/SSIP:5061
tport_vsend(00DFE3E8): 652 bytes of 652 to udp/SSIP:5061
tport_vsend returned 652
nta: sent 415 Unsupported Media Type for INVITE (1)
tport_wakeup_pri(00DFE3E8): events IN
tport_recv_event(00DFE3E8)
tport(00DFE3E8) msg 0192C510 from (udp/MyIP:5060) has 388 bytes, veclen =
 1
tport(00DFE3E8): msg 0192C510 (388 bytes) from udp/SSIP:5060/sip next=

nta: received ACK sip:a...@myip:5060;user=phone SIP/2.0 (CSeq 1)
nta: ACK (1) is going to INVITE (1)
nta: timer set next to 937 ms
nta: timer J fired, terminate 200 response
incoming_reclaim_all(, , 02E2FEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
nta: timer set next to 3859 ms
nta: timer I fired, terminate 415 response
incoming_reclaim_all(, , 02E2FEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set

2009/11/2 Lei Tang lei.tl...@gmail.com

 Hi all, I get a  415 Unsupported Media Type when FS receiving call from a
 softswitch. I captured some packets, It seems that the softswitch use SIP-I
 protocol, does FS can handle SIP-I message?

 ===here is the invite messagefrom softswitch
 INVITE sip:xx...@:5060;user=phone SIP/2.0
 Contact: sip:x...@:5061
 MIME-version: 1.0
 Content-Type: multipart/mixed;boundary=Alcatel-boundary
 To: sip:x...@:5060;user=phone
 From: sip:x...@x:5061;user=phone;tag=73D332463135364195291201
 P-Asserted-Identity: sip:x...@x:5061;user=phone
 Allow:
 INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE
 Supported: 100rel,timer,replaces,diversion
 Expires: 155
 Session-Expires: 1800
 Min-SE: 90
 Call-ID: 01fd0348728140023...@sip-3
 Max-Forwards: 70
 CSeq: 1 INVITE
 Timestamp: 10645
 Via: SIP/2.0/UDP x:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E
 Content-Length: 542

 --Alcatel-boundary
 Content-Type: application/sdp

 v=0
 o=- 5 8 IN IP4 xx
 s=SDP Data
 c=IN IP4 x
 t=0 0
 m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97
 a=rtpmap:110 speex/8000/1
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:96 TELEPHONE-EVENT/8000
 a=fmtp:110 mode=3
 a=rtpmap:111 speex/8000/1
 a=rtpmap:3 GSM/8000
 a=ptime:30
 a=fmtp:111 mode=2
 a=rtpmap:112 speex/8000/1

 --Alcatel-boundary
 Content-Type: application/ISUP;version=N/A

 .. 
 ...0.D...
 ...V..0..1...n..p...
 --Alcatel-boundary--

 =response message from FS
 SIP/2.0 415 Unsupported Media Type
 Via: SIP/2.0/UDP x:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E
 From: x sip:xx...@:5061;user=phone;tag=73D332463135364195291201
 To: sip:xx...@x:5060;user=phone;tag=m6XS44BKF4pSS
 Call-ID: 01fd0348728140023...@sip-3
 CSeq: 1 INVITE
 Timestamp: 10645 0.00
 User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
 Accept: application/sdp
 Accept-Encoding:
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
 NOTIFY, REFER, UPDATE, REGISTER, INFO
 Supported: timer, precondition, path, replaces
 Allow-Events: talk, refer
 Content-Length: 0
 --
 Lei.Tang
 lei.tl...@gmail.com




-- 
Lei.Tang
lei.tl...@gmail.com
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Re: [Freeswitch-users] Rejecting a call from JavaScript

2009-11-01 Thread Yehavi Bourvine
Thanks! It works!

  __Yehavi:

 2009/11/1 Anthony Minessale anthony.miness...@gmail.com

 try session.execute(hangup, user_busy);


   On Sun, Nov 1, 2009 at 8:24 AM, Yehavi Bourvine 
 yehavi.bourv...@gmail.com wrote:

   Hello,

   We would like to handle an incoming call to a busy phone according
 to user's prefference:  Some want waiting call, some want to just reject the
 call, and others want to send the call to voicemail.

   We have a small JavaScript which tests the status of the destination and
 the user's will and tries to act accordingly. Our problem is how to send
 busy. I tried session.hangup(USER_BUSY) but it always sends temporary
 unavailable which causes the orignator to think that the destination is out
 of order.

 What is the correct way to do so?

Thanks! __Yehavi:

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 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session

2009-11-01 Thread Anthony Minessale
We already concluded its your unacceptabe use of originate in hangup hook
right?

On Nov 1, 2009 7:45 PM, Seven Du dujinf...@gmail.com wrote:

Just suspicious would be possible that happened on sqlite stage? I
manually deleted the channels from sqlite and nothing bad happend.
just FYI.

-- Forwarded message --
From: Seven Du dujinf...@gmail.com
Date: Sun, 1 Nov 2009 10:24:32 +0800
Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session
To:

Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even
unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound
and outbound. I must be very careful if I upgrade to trunk and turn on
rwlock debug because it's on production and the the problem not happening
that much so would be hard to trace. But I will find time to test and report
back.

FYI, I also noticed that in some zombile channels, couple of INVITEs sent
but never got a response. However, it sent out an ACK at last.

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