[Freeswitch-users] Fwd: freeswitch support PCMU only?

[qian ma]

-- Forwarded message --
From: qian ma 
Date: 2009/7/1
Subject: freeswitch support PCMU only?
To: "Freeswitch-users@lists.freeswitch.org" <
Freeswitch-users@lists.freeswitch.org>


hi all
   freeswitch support PCMU only?
   i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml,
but freeswitch still support PCMU only,
   below is the trace:

   2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec
Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
to 101
2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
[telephone-event:101:8000:0]/[PCMU:0:8000:20]
2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup
sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal
sofia/maq/9...@58.212.219.104 [KILL]
2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal
sofia/maq/9...@58.212.219.104 [BREAK]
2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 (
sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 (
sofia/maq/9...@58.212.219.104) State HANGUP



  how to configure the freeswitch??
  support more codecs???

  thx!

m.q
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Re: [Freeswitch-users] freeswitch support PCMU only?

[seven]

absolutely not.

codec negotiate depending on your conf. do you have a sip trace?

On Jul 1, 2009, at 2:48 PM, qian ma wrote:


hi all
   freeswitch support PCMU only?
   i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml 
,   but freeswitch still support PCMU only,

   below is the trace:

   2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio  
Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf  
payload to 101
2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec  
Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20]
2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup sofia/maq/9...@58.212.219.104 
 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal sofia/maq/9...@58.212.219.104 
 [KILL]
2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send  
signal sofia/maq/9...@58.212.219.104 [BREAK]
2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 (sofia/maq/9...@58.212.219.104 
) Running State Change CS_HANGUP
2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 (sofia/maq/9...@58.212.219.104 
) State HANGUP




  how to configure the freeswitch??
  support more codecs???

  thx!
  m 
.q

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Re: [Freeswitch-users] Error in OpenZap

[Michael S Collins]

Look more closely at the output. It looks like mod_libpri.so didn't  
get installed properly. I think this is a bug in the ozmod_libpri  
build. For now just locate that missing .so file in your oz build  
environment and copy it to the freeswitch/mod directory and try again.


-MC

Sent from my iPhone

On Jun 30, 2009, at 11:29 PM, Baskar  wrote:


Hi,

i have changed the openzap.conf file but still i get the same error

[span wanpipe 1]
number => 1
trunk_type => e1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31

freeswi...@localhost.localdomain> load mod_libpri
API CALL [load(mod_libpri)] output:
-ERR [module load file routine returned an error]

2009-07-01 11:27:51 [CRIT] switch_loadable_module.c:871  
switch_loadable_module_load_file() Error Loading module /usr/local/ 
freeswitch/mod/mod_libpri.so
**/usr/local/freeswitch/mod/mod_libpri.so: cannot open shared object  
file: No such file or directory**

freeswi...@localhost.localdomain> load mod_openzap
2009-07-01 11:28:04 [NOTICE] zap_io.c:2626 zap_global_init() Modules  
configured: 1
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'name' / 'OpenZAP'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'number' / '1'

API CALL [load(mod_openzap)] output:
-ERR [module load file routine returned an error]

2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'trunk_type' / 'E1'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'b-channel' / '1:1-15'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'd-channel' / '1:16'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'b-channel' / '1:17-31'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'name' / 'OpenZAP'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'number' / '2'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'trunk_type' / 'E1'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'b-channel' / '2:1-15'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'd-channel' / '2:16'
2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param  
[] 'b-channel' / '2:17-31'
2009-07-01 11:28:04 [INFO] zap_io.c:2370 load_config() Configured 0  
channel(s)
2009-07-01 11:28:04 [ERR] zap_io.c:2633 zap_global_init() No modules  
configured!
2009-07-01 11:28:04 [ERR] mod_openzap.c:2401 mod_openzap_load()  
Error loading OpenZAP
2009-07-01 11:28:04 [CRIT] switch_loadable_module.c:871  
switch_loadable_module_load_file() Error Loading module /usr/local/ 
freeswitch/mod/mod_openzap.so

**Module load routine returned an error**


--
Thanks with Regards,
N.Baskar

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Re: [Freeswitch-users] freeswitch support PCMU only?

[Nandy Dagondon]

fs can support lots of codecs. you can find the ff variables defined in
vars.xml:

global_codec_prefs
outbound_codec_prefs

then look for "inbound_codec_negotiation" in
sip_profiles/internal.xml,sip_profiles/external.xml if you want your
codec_prefs to set priority or not.

-nandy

On Wed, Jul 1, 2009 at 2:48 PM, qian ma  wrote:

> hi all
>freeswitch support PCMU only?
>i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml,
> but freeswitch still support PCMU only,
>below is the trace:
>
>2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec
> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
> to 101
> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup
> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal
> sofia/maq/9...@58.212.219.104 [KILL]
> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 (
> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP
>
>
>
>   how to configure the freeswitch??
>   support more codecs???
>
>   thx!
>
> m.q
>
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>
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Re: [Freeswitch-users] freeswitch support PCMU only?

[qian ma]

thanks for your replies.

my var.xml:




below is the sip trace:
 recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
   
   INVITE sip:123...@58.212.219.104  SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.241:8422
;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
   Max-Forwards: 70
   Contact: 
   To: "123456">
   From: "9876"
>;tag=057de365
   Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
   CSeq: 1 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
   Content-Type: application/sdp
   User-Agent: eyeBeam release 1102u stamp 52345
   Content-Length: 237

   v=0
   o=- 6 2 IN IP4 192.168.1.241
   s=CounterPath eyeBeam 1.5
   c=IN IP4 192.168.1.241
   t=0 0
   m=audio 57862 RTP/AVP 8 101
   a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
   a=fmtp:101 0-15
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   
send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
   
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.1.241:8422
;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
   From: "9876"
>;tag=057de365
   To: "123456">
   Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
   Content-Length: 0

   
2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
sofia/maq/9...@58.212.219.104 entering state [received][100]
2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
v=0
o=- 6 2 IN IP4 192.168.1.241
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.241
t=0 0
m=audio 57862 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
[PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload to
101
2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
[telephone-event:101:8000:0]/[PCMU:0:8000:20]
2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
sofia/maq/9...@58.212.219.104 [KILL]
2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal
sofia/maq/9...@58.212.219.104 [BREAK]
2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
sofia/maq/9...@58.212.219.104) State HANGUP
2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
sofia/maq/9...@58.212.219.104 hanging up, cause: INCOMPATIBLE_DESTINATION
2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE with:
488
send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766:
   
   SIP/2.0 488 Not Acceptable Here
   Via: SIP/2.0/UDP 192.168.1.241:8422
;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
   From: "9876"
>;tag=057de365
   To: "123456" 
>;tag=28Q0QB73Bm35K
   Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
   Content-Length: 0

   
2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46
sofia/maq/9...@58.212.219.104 Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION
2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 (
sofia/maq/9...@58.212.219.104) State HANGUP going to sleep
2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 (
sofia/maq/9...@58.212.219.104) State Change CS_HANGUP -> CS_REPORTING
2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal
sofia/maq/9...@58.212.219.104 [BREAK]
2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 (
sofia/maq/9...@58.212.219.104) Running State Change CS_REPORTING
2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 (
sofia/maq/9...@58.212.219.104) State REPORTING
recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591:
   

Re: [Freeswitch-users] freeswitch support PCMU only?

[Nandy Dagondon]

seven,

of course, codec negotiation depends on the order of codecs in the
*_codec_prefs variables. but, the opposite end has also it's own codecs
prefs, too. fs can accept the other end's prefs
(inbound_codec_negotiation=generous) or imposes it's own prefs (=greedy).
you must include the codec in the *_codec_prefs to activate it. is this
correct?

-nandy

On Wed, Jul 1, 2009 at 3:19 PM, seven  wrote:

> absolutely not.
> codec negotiate depending on your conf. do you have a sip trace?
>
> On Jul 1, 2009, at 2:48 PM, qian ma wrote:
>
> hi all
>freeswitch support PCMU only?
>i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml,
> but freeswitch still support PCMU only,
>below is the trace:
>
>2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec
> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
> to 101
> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup
> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal
> sofia/maq/9...@58.212.219.104 [KILL]
> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 (
> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP
>
>
>
>   how to configure the freeswitch??
>   support more codecs???
>
>   thx!
>
> m.q
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>
>
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>
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Re: [Freeswitch-users] Any advances on T.38 support for FS?

[François Delawarde]

Is there any work planned for T.38 termination (in mod_fax)?

François.

On Tue, 2009-06-30 at 12:07 -0400, Michael Jerris wrote:
> We currently support t.38 passthrough only using proxy_media mode.  T. 
> 38 gateway is on the roadmap but not yet close to complete.
> 
> Mike
> 
> On Jun 30, 2009, at 5:15 AM, François Delawarde wrote:
> 
> > Many issues on Asterisk's T.38 (or probably just on T.38?)...
> >
> > Could it convince those relying on this "modern" version of a 50yo
> > technology to switch to and with FreeSwitch?
> 
> 
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Re: [Freeswitch-users] Could this be a bug in the SIP registry?

[Jason White]

Mitchel Constantin  wrote:
> 5. My phones now register using the correct domain name (i.e. weavver.com)
> instead of the IP address (205.134.225.20) as the domain.
> 6. Now the problem... My originate command no longer works using the new
> syntax: originate 
> sofia/internal/mythicalbox%weavver.comsofia/internal/johndoe%
> weavver.com
> 
> The phones do show up as registered when I type "sofia status profile
> internal":

What happens if you use the following syntax?

originate user/ph...@domain extension

e.g.
originate user/1...@example.com 3000
to connext u...@example.com to extension 3000.

My other advice would be to read the FreeSWITCH log files carefully. Also, use
the sofia_contact command to find out how the registered users will be called
when the syntax mentioned above is used. Make sure that everything will go
where you want it.


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Re: [Freeswitch-users] Any advances on T.38 support for FS?

[Jason White]

François Delawarde  wrote:
> Is there any work planned for T.38 termination (in mod_fax)?

Yes, as discussed on the mailing list recently.

If you're volunteering to help, I'm sure the FreeSWITCH developers would
appreciate contributions of code.


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Re: [Freeswitch-users] Could this be a bug in the SIP registry?

[Jason White]

Jason White  wrote:
> originate user/1...@example.com 3000
> to connext u...@example.com to extension 3000.

That should read "to connect 1...@example.com to extension 3000".


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Re: [Freeswitch-users] Cant register a pointer. What wrong?

[Alexey Lubimov]

grep -ir 111 *

default/bad.xml: 
default/bad.xml: 
default/bad.xml: 
default.xml: 

default.xml:















Michael Jerris пишет:
> you have a pointer somewhere in your directory for that user, hard to  
> see without seeing the whole config, but grep for 111 and see what  
> else you find.
>
> Mike
>
> On Jun 30, 2009, at 10:21 AM, Alexey Lubimov wrote:
>
>   
>> I sofia_reg.c:1765  have two user records - good #110 and bad #111.
>>
>> bad.xml:
>>
>> 
>>  
>>
>>  
>>  
>>
>>
>>  > value="domestic,international,local"/>
>>  
>>  
>>  
>>  
>>  > value="$${outbound_caller_name}"/>
>>  > value="$${outbound_caller_id}"/>
>>  
>>
>>  
>>
>>
>> good.xml:
>>
>> 
>>  
>>
>>  
>>  
>>
>>
>>  > value="domestic,international,local"/>
>>  
>>  
>>  
>>  
>>  > value="$${outbound_caller_name}"/>
>>  > value="$${outbound_caller_id}"/>
>>  
>>
>>  
>> 
>>
>> Good user 110 work without any problem. But user "Bad" user 111 can't
>> register to freeswitch.
>>
>> In log I can see only one message -  2009-06-30 16:31:36.590970
>> [WARNING] sofia_reg.c:1765 Cant register a pointer.
>>
>> good user exists:
>> freeswi...@internal> user_exists id 110 neolant.ru
>> true
>>
>> and bad user exists!
>> freeswi...@internal> user_exists id 111 neolant.ru
>> true
>>
>> good user don't have attr type:
>>
>> freeswi...@internal> user_data  1...@neolant.ru attr type
>> -ERR no reply
>>
>> but bad user have attr type! :
>>
>> freeswi...@internal> user_data  1...@neolant.ru attr type
>> pointer
>>
>>
>> Good user have password:
>>
>> freeswi...@internal> user_data  1...@neolant.ru param password
>> 123456
>>
>> But bad user no have param pasword!
>>
>> freeswi...@internal> user_data  1...@neolant.ru param password
>> -ERR no reply
>>
>>
>> What's wrong in these configuration? How I can debug and resolve these
>> problems?
>> 
>
>
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Re: [Freeswitch-users] freeswitch support PCMU only?

[Nandy Dagondon]

you FS doesn't accept PCMU. try to add "PCMU" on both variables.

On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:

> thanks for your replies.
>
my var.xml:
> 
> 
>
>
> below is the sip trace:
>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>
>INVITE sip:123...@58.212.219.104  SIP/2.0
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>Max-Forwards: 70
>Contact: 
>To: "123456">
>From: "9876"
> >;tag=057de365
>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>CSeq: 1 INVITE
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
>Content-Type: application/sdp
>User-Agent: eyeBeam release 1102u stamp 52345
>Content-Length: 237
>
>v=0
>o=- 6 2 IN IP4 192.168.1.241
>s=CounterPath eyeBeam 1.5
>c=IN IP4 192.168.1.241
>t=0 0
>m=audio 57862 RTP/AVP 8 101
>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>a=fmtp:101 0-15
>a=rtpmap:101 telephone-event/8000
>a=sendrecv
>
> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>From: "9876"
> >;tag=057de365
>To: "123456">
>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>CSeq: 1 INVITE
>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>Content-Length: 0
>
>
> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
> sofia/maq/9...@58.212.219.104 entering state [received][100]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
> v=0
> o=- 6 2 IN IP4 192.168.1.241
> s=CounterPath eyeBeam 1.5
> c=IN IP4 192.168.1.241
> t=0 0
> m=audio 57862 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
> to 101
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
> sofia/maq/9...@58.212.219.104 [KILL]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP
> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
> sofia/maq/9...@58.212.219.104 hanging up, cause: INCOMPATIBLE_DESTINATION
> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE
> with: 488
> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766:
>
>SIP/2.0 488 Not Acceptable Here
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>From: "9876"
> >;tag=057de365
>To: "123456" 
> >;tag=28Q0QB73Bm35K
>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>CSeq: 1 INVITE
>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>Accept: application/sdp
>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
>Supported: timer, precondition, path, replaces
>Allow-Events: talk, refer
>Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
>Content-Length: 0
>
>
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46
> sofia/maq/9...@58.212.219.104 Standard HANGUP, cause:
> INCOMPATIBLE_DESTINATION
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP going to sleep
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 (
> sofia/maq/9...@58.212.219.104) State Change CS_HANGUP -> CS_REPORTING
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state

[Freeswitch-users] How to know my gateway registering is successed??

[Brad Tuan]

As title ,Does FS keep the status of gateways??
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Re: [Freeswitch-users] Cant register a pointer. What wrong?

[Gonzalo Servat]

On Wed, Jul 1, 2009 at 6:56 PM, Alexey Lubimov  wrote:

>
> grep -ir 111 *
>
> default/bad.xml: 
> default/bad.xml: 
> default/bad.xml: 
> default.xml: 
>

[..snip..]

This is probably a long shot, but for bad user you didn't close off the
 (I'm assuming you just didn't paste it but it's closed off, but
you never know...)

- Gonzalo
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Re: [Freeswitch-users] How to know my gateway registering is successed??

[Jason White]

Brad Tuan  wrote:
> As title ,Does FS keep the status of gateways??

sofia status gateway 


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Re: [Freeswitch-users] Cant register a pointer. What wrong?

[Alexey Lubimov]

No, tag  is exists.





















Gonzalo Servat пишет:
> On Wed, Jul 1, 2009 at 6:56 PM, Alexey Lubimov  > wrote:
>
>
> grep -ir 111 *
>
> default/bad.xml: 
> default/bad.xml: 
> default/bad.xml:  value="111"/>
> default.xml: 
>
>
> [..snip..]
>
> This is probably a long shot, but for bad user you didn't close off 
> the  (I'm assuming you just didn't paste it but it's closed 
> off, but you never know...)
>
> - Gonzalo
>
> 
>
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Re: [Freeswitch-users] Cant register a pointer. What wrong?

[Alexey Lubimov]

Thank You, Gonzalo!

freeswi...@internal> reloadxml
+OK [Success]

2009-07-01 13:40:37.205835 [ERR] switch_xml.c:1282 Couldnt open 
/opt/freeswitch/conf/directory/default/bad.xml (Permission denied)

ls -l
-rw-r- 1 root   root   756 2009-06-30 16:39 bad.xml
-rw-r- 1 freeswitch daemon 761 2009-06-19 12:02 good.xml

After chown freeswitch:daemon, problem was resolved.



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Re: [Freeswitch-users] How to know my gateway registering is successed??

[Peter P GMX]

or simply
sofia status
for all gateways

Jason White schrieb:
> Brad Tuan  wrote:
>   
>> As title ,Does FS keep the status of gateways??
>> 
>
> sofia status gateway 
>
>
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Re: [Freeswitch-users] Testing Freeswitch performance led to strange behavior

[Apostolos Pantsiopoulos]

I am writing this to let you know that this behavior
persists in the 1.0.4pre9.

Could the calls/sec issue be due to the single threaded nature of Sofia?
Because I am getting the feeling that the number of simultaneous 
channels doesn't really burdens FS, but many Calls/sec does.



Apostolos Pantsiopoulos wrote:
> Anthony Minessale wrote:
>> FS uses async rtp timers so you may want to set rtp-timer-name=none in 
>> the profile param to simulate asterisk conditions.
> 
> I tried that - although I am not using rtp in my scenario - with the 
> same results.
> 
>> Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit 
>> single cpu box because that was what was popular when it was designed 
>> and the chance for race conditions is minimal because there is only 1 
>> cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic 
>> difference.
> 
> Yes I know that this machine is not well suited for today's test needs.
> But the issue occurs in every machine as long as it is pushed near (but 
> not quite near) to its limits. I have the same odd durations using a 64 
> bit low end server. In this case I could achieve a better call/sec rate
> than that of the crappy PC but around 50-60 calls/sec the same problem
> showed up. I also used a Mosso Cloud Server (quad core - 64-bit) and the 
> same thing happened at a higher rate.
> 
> 
>> I will be happy to investigate this issue a bit if you'd like but i do 
>> not have any box like you describe so if I can't find anything
>> you may have to lend us your lab.
> 
> I would appreciate it if you did. After all there this might be a 
> problem that has not surfaced yet but someday will as more and more
> production boxes start using FS. So it would be better to investigate it 
> now.
> I don't think lending you access to my old P4 PC would help you very much :)
> If you have access to a normal 2-4 core system you can easily start 
> raising the sipp parameters until it starts happening. However if you 
> really think it is appropriate to use my test machines I'd be happy to 
> grant access to our low-end Opteron machine (just send me a personal 
> email). I cannot grant you access to larger systems because they are 
> used in production.
> 
> I used the embedded sipp scenarios :
> 
> on the UAS side :
> 
> sipp -i  -mi  -ci  -mp 8000 -sn uas
> 
> on the UAC side :
> 
> sipp :5060 -s 44050505-i  -mi  -ci  -r 70 
> -d 5000 -l 500  -m 2000 -sn uac
> 
> The dialplan :
> 
> 
> 
> 
> 
> 
>   
>   
>   
>   data="absolute_codec_string=PCMU"/>
> 
>data="sofia/gateway/sipp01/$1"/>
>   
>   
> 
> 
> 
> 
> If you need anything else from the config just notify me.
> 
> In order to verify that at some point the calls start having a
> duration larger than the scenario's 5secs you can tcpdump on the sipp 
> machine or turn on the cdrs logging (I know that it degrades 
> performance, but as I said it is not a matter of when exactly it
> starts happening, it is a matter that it DOES start happening).
> 
> 
>>
>> On Thu, Jun 4, 2009 at 12:47 PM, r...@kinetix.gr 
>>  mailto:r...@kinetix.gr>> wrote:
>>
>> Michael Collins wrote:
>>  >
>>  >
>>  > The dialplan :
>>  >
>>  > 
>>  > 
>>  > 
>>  >
>>  > 
>>  >  
>>  >  >  > expression="^.*$">
>>  >
>>  >
>>  > You forgot the parens around .*
>>  > It should be expression="^(.*)$" if you plan to use $1 later in the
>>  > extension...
>>  >
>>  >
>>  >
>>  >  
>>  > >  > data="absolute_codec_string=PCMA"/>
>>  >  >  > data="sofia/gateway/sipp01/$1"/>
>>  >
>>  > ... like here ^^^
>>  > :)
>>  > -MC
>>
>> You are right! Although, I don't think that would change the outcome of
>> my test :)
>>  >
>>  >
>>  >
>>  >  
>>  >  
>>  > 
>>  >
>>  > 
>>  >
>>  >
>>  >
>> 
>>  >
>>  > ___
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>> 
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>>  >
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>>  >
>>
>>
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Re: [Freeswitch-users] Any advances on T.38 support for FS?

[François Delawarde]

Great news! For what I could read, the most famous DSP programmer
worldwide (Steve) seems to be helping out for mod_fax.

I guess I should register to freeswitch-dev to monitor this closely.

Thanks,
François.




On Wed, 2009-07-01 at 18:38 +1000, Jason White wrote:
> François Delawarde  wrote:
> > Is there any work planned for T.38 termination (in mod_fax)?
> 
> Yes, as discussed on the mailing list recently.
> 
> If you're volunteering to help, I'm sure the FreeSWITCH developers would
> appreciate contributions of code.
> 
> 
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Re: [Freeswitch-users] How to know my gateway registering is successed??

[Jason White]

Peter P GMX  wrote:
> or simply
> sofia status
> for all gateways

and, from the shell,
fs_cli -x help > helpfile
fs_cli -x sofia help >> helpfile
and any others you need so as to obtain synopses of all the commands that you
might need.


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[Freeswitch-users] Freeswitch memory usage is too high

[Muhammad Danish Moosa]

Hi

Freeswitch is being used in a scenario where two endpoints are running
traffic with bypass media mode. Performance is good and all things are
smooth.

But as the time goes after starting freeswitch, it starts consuming almost
whole of memory. Note , freeswitch is being started with -core option, is it
related?

If this 99% memory consumption is any red alert, as we can see calls are
still connecting fine and all is going as usual.


-- 
Muhammad Danish Moosa
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Re: [Freeswitch-users] freeswitch support PCMU only?

[qian ma]

i want the fs accept the PCMA not PCMU.
i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
work. FS only accept PCMU.
why??




2009/7/1 Nandy Dagondon 

> you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>
>
> On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:
>
>> thanks for your replies.
>>
>  my var.xml:
>> 
>> 
>>
>>
>> below is the sip trace:
>>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>>
>> 
>>INVITE sip:123...@58.212.219.104  SIP/2.0
>>Via: SIP/2.0/UDP 192.168.1.241:8422
>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>>Max-Forwards: 70
>>Contact: 
>>To: "123456">
>>From: "9876"
>> >;tag=057de365
>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>CSeq: 1 INVITE
>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>> SUBSCRIBE, INFO
>>Content-Type: application/sdp
>>User-Agent: eyeBeam release 1102u stamp 52345
>>Content-Length: 237
>>
>>v=0
>>o=- 6 2 IN IP4 192.168.1.241
>>s=CounterPath eyeBeam 1.5
>>c=IN IP4 192.168.1.241
>>t=0 0
>>m=audio 57862 RTP/AVP 8 101
>>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>a=fmtp:101 0-15
>>a=rtpmap:101 telephone-event/8000
>>a=sendrecv
>>
>> 
>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>>
>> 
>>SIP/2.0 100 Trying
>>Via: SIP/2.0/UDP 192.168.1.241:8422
>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>From: "9876"
>> >;tag=057de365
>>To: "123456">
>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>CSeq: 1 INVITE
>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>Content-Length: 0
>>
>>
>> 
>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
>> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
>> sofia/maq/9...@58.212.219.104 entering state [received][100]
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
>> v=0
>> o=- 6 2 IN IP4 192.168.1.241
>> s=CounterPath eyeBeam 1.5
>> c=IN IP4 192.168.1.241
>> t=0 0
>> m=audio 57862 RTP/AVP 8 101
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
>> [PCMA:8:8000:0]/[PCMU:0:8000:20]
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
>> to 101
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
>> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
>> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
>> sofia/maq/9...@58.212.219.104 [KILL]
>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal
>> sofia/maq/9...@58.212.219.104 [BREAK]
>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
>> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
>> sofia/maq/9...@58.212.219.104) State HANGUP
>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
>> sofia/maq/9...@58.212.219.104 hanging up, cause: INCOMPATIBLE_DESTINATION
>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE
>> with: 488
>> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766:
>>
>> 
>>SIP/2.0 488 Not Acceptable Here
>>Via: SIP/2.0/UDP 192.168.1.241:8422
>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>From: "9876"
>> >;tag=057de365
>>To: "123456" 
>> >;tag=28Q0QB73Bm35K
>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>CSeq: 1 INVITE
>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>Accept: application/sdp
>>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO
>>Supported: timer, precondition, path, replaces
>>Allow-Events: talk, refer
>>Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
>>Content-Length: 0
>>
>>
>> 
>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46
>> sofia/maq/9...@58.212.219.104 Standard HANGUP, cause:
>> INCOMPATIBLE_DESTINATION
>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 (
>> sofia/maq/9...@58.212.219.104) State HANGUP going to sleep
>> 2009-07-01

Re: [Freeswitch-users] Freeswitch memory usage is too high

[Raymond Chandler]

On 07/01/2009 08:29 AM, Muhammad Danish Moosa wrote:
> Hi
>
> Freeswitch is being used in a scenario where two endpoints are running 
> traffic with bypass media mode. Performance is good and all things are 
> smooth.
>
> But as the time goes after starting freeswitch, it starts consuming 
> almost whole of memory.
How much is the "whole"?  You should see the memory usage level off, it 
won't keep growing forever.

-Ray

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Re: [Freeswitch-users] freeswitch support PCMU only?

[Nandy Dagondon]

check the value of "inbound_codec_negotiation" in  the sip_profiles/*.xml
files. is it "generous" or "greedy"? you should also check if the endpoint
is offering PCMU.


On Wed, Jul 1, 2009 at 8:27 PM, qian ma  wrote:

> i want the fs accept the PCMA not PCMU.
> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
> work. FS only accept PCMU.
> why??
>
>
>
>
> 2009/7/1 Nandy Dagondon 
>
> you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>>
>>
>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:
>>
>>> thanks for your replies.
>>>
>>  my var.xml:
>>> 
>>> 
>>>
>>>
>>> below is the sip trace:
>>>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>>>
>>> 
>>>INVITE sip:123...@58.212.219.104 SIP/2.0
>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>>>Max-Forwards: 70
>>>Contact: 
>>>To: "123456">
>>>From: "9876"
>>> >;tag=057de365
>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>CSeq: 1 INVITE
>>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>> SUBSCRIBE, INFO
>>>Content-Type: application/sdp
>>>User-Agent: eyeBeam release 1102u stamp 52345
>>>Content-Length: 237
>>>
>>>v=0
>>>o=- 6 2 IN IP4 192.168.1.241
>>>s=CounterPath eyeBeam 1.5
>>>c=IN IP4 192.168.1.241
>>>t=0 0
>>>m=audio 57862 RTP/AVP 8 101
>>>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>>a=fmtp:101 0-15
>>>a=rtpmap:101 telephone-event/8000
>>>a=sendrecv
>>>
>>> 
>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>>>
>>> 
>>>SIP/2.0 100 Trying
>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>>From: "9876"
>>> >;tag=057de365
>>>To: "123456">
>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>CSeq: 1 INVITE
>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>>Content-Length: 0
>>>
>>>
>>> 
>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
>>> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
>>> sofia/maq/9...@58.212.219.104 entering state [received][100]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
>>> v=0
>>> o=- 6 2 IN IP4 192.168.1.241
>>> s=CounterPath eyeBeam 1.5
>>> c=IN IP4 192.168.1.241
>>> t=0 0
>>> m=audio 57862 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
>>> [PCMA:8:8000:0]/[PCMU:0:8000:20]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload
>>> to 101
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
>>> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
>>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
>>> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
>>> sofia/maq/9...@58.212.219.104 [KILL]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal
>>> sofia/maq/9...@58.212.219.104 [BREAK]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
>>> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
>>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
>>> sofia/maq/9...@58.212.219.104) State HANGUP
>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
>>> sofia/maq/9...@58.212.219.104 hanging up, cause:
>>> INCOMPATIBLE_DESTINATION
>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE
>>> with: 488
>>> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766:
>>>
>>> 
>>>SIP/2.0 488 Not Acceptable Here
>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>>From: "9876"
>>> >;tag=057de365
>>>To: "123456" 
>>> >;tag=28Q0QB73Bm35K
>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>CSeq: 1 INVITE
>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>>Accept: application/sdp
>>>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO
>>>Supported: timer, precondition, path, replaces
>>>Allow-Events: talk, refer
>>>Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
>>>Content-Length: 0
>>>
>>>
>>> 

Re: [Freeswitch-users] freeswitch support PCMU only?

[Nandy Dagondon]

sorry. i mean check the x-lite client if PCMA is enabled?


On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon  wrote:

> check the value of "inbound_codec_negotiation" in  the sip_profiles/*.xml
> files. is it "generous" or "greedy"? you should also check if the endpoint
> is offering PCMU.
>
>
>
> On Wed, Jul 1, 2009 at 8:27 PM, qian ma  wrote:
>
>> i want the fs accept the PCMA not PCMU.
>> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
>> work. FS only accept PCMU.
>> why??
>>
>>
>>
>>
>> 2009/7/1 Nandy Dagondon 
>>
>> you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>>>
>>>
>>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:
>>>
 thanks for your replies.

>>>  my var.xml:
 
 


 below is the sip trace:
  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:

 
INVITE sip:123...@58.212.219.104 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.241:8422
 ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
Max-Forwards: 70
Contact: 
To: "123456"
 >
From: "9876"
 >;tag=057de365
Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 237

v=0
o=- 6 2 IN IP4 192.168.1.241
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.241
t=0 0
m=audio 57862 RTP/AVP 8 101
a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

 
 send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:

 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.241:8422
 ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
From: "9876"
 >;tag=057de365
To: "123456"
 >
Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
Content-Length: 0


 
 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
 sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
 sofia/maq/9...@58.212.219.104 entering state [received][100]
 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
 v=0
 o=- 6 2 IN IP4 192.168.1.241
 s=CounterPath eyeBeam 1.5
 c=IN IP4 192.168.1.241
 t=0 0
 m=audio 57862 RTP/AVP 8 101
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
 a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
 [PCMA:8:8000:0]/[PCMU:0:8000:20]
 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf
 payload to 101
 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
 [telephone-event:101:8000:0]/[PCMU:0:8000:20]
 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
 sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
 sofia/maq/9...@58.212.219.104 [KILL]
 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal
 sofia/maq/9...@58.212.219.104 [BREAK]
 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
 sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
 sofia/maq/9...@58.212.219.104) State HANGUP
 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
 sofia/maq/9...@58.212.219.104 hanging up, cause:
 INCOMPATIBLE_DESTINATION
 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE
 with: 488
 send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766:

 
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.241:8422
 ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
From: "9876"
 >;tag=057de365
To: "123456" 
 >;tag=28Q0QB73Bm35K
Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE

Re: [Freeswitch-users] freeswitch support PCMU only?

[qian ma]

inbound_codec_negotiation is generous
and the xlite PCMU is enabled.

my var.xml.conf:





2009/7/1 Nandy Dagondon 

> sorry. i mean check the x-lite client if PCMA is enabled?
>
>
> On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon  wrote:
>
>> check the value of "inbound_codec_negotiation" in  the sip_profiles/*.xml
>> files. is it "generous" or "greedy"? you should also check if the endpoint
>> is offering PCMU.
>>
>>
>>
>> On Wed, Jul 1, 2009 at 8:27 PM, qian ma  wrote:
>>
>>> i want the fs accept the PCMA not PCMU.
>>> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
>>> work. FS only accept PCMU.
>>> why??
>>>
>>>
>>>
>>>
>>> 2009/7/1 Nandy Dagondon 
>>>
>>> you FS doesn't accept PCMU. try to add "PCMU" on both variables.


 On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:

> thanks for your replies.
>
  my var.xml:
> 
> 
>
>
> below is the sip trace:
>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>
> 
>INVITE sip:123...@58.212.219.104 SIP/2.0
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>Max-Forwards: 70
>Contact: 
>To: "123456"
> >
>From: "9876"
> >;tag=057de365
>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>CSeq: 1 INVITE
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
>Content-Type: application/sdp
>User-Agent: eyeBeam release 1102u stamp 52345
>Content-Length: 237
>
>v=0
>o=- 6 2 IN IP4 192.168.1.241
>s=CounterPath eyeBeam 1.5
>c=IN IP4 192.168.1.241
>t=0 0
>m=audio 57862 RTP/AVP 8 101
>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>a=fmtp:101 0-15
>a=rtpmap:101 telephone-event/8000
>a=sendrecv
>
> 
> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>
> 
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>From: "9876"
> >;tag=057de365
>To: "123456"
> >
>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>CSeq: 1 INVITE
>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>Content-Length: 0
>
>
> 
> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
> sofia/maq/9...@58.212.219.104 entering state [received][100]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
> v=0
> o=- 6 2 IN IP4 192.168.1.241
> s=CounterPath eyeBeam 1.5
> c=IN IP4 192.168.1.241
> t=0 0
> m=audio 57862 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf
> payload to 101
> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare
> [telephone-event:101:8000:0]/[PCMU:0:8000:20]
> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
> sofia/maq/9...@58.212.219.104 [KILL]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal
> sofia/maq/9...@58.212.219.104 [BREAK]
> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
> sofia/maq/9...@58.212.219.104) State HANGUP
> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
> sofia/maq/9...@58.212.219.104 hanging up, cause:
> INCOMPATIBLE_DESTINATION
> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE
> with: 488
> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766:
>
> 
>SIP/2.0 488 Not Acceptable Here
>Via: SIP/2.0/UDP 192.168.1.241:8422
> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>From: "9876"
> >;tag=057de365
>To: "123456" 
> >;tag=28Q

[Freeswitch-users] Javascript session Recording

[Baskar]

*Hi,

I have configured outbound call through JavaScript it is working fine but i
want the conversation to be recorded .

Javascript:

sessionA = new Session("{ignore_early_media=true,
origination_uuid="+argv[0]+"}sofia/default/sip:"+argv[0]+"@
192.168.1.135:5066");
sessionB = new Session("sofia/internal/"+ argv[1] +"%192.168.1.77");
rtn = sessionA .recordFile("/tmp/"+ argv[0] +".wav", "", "", 4, 500,
3);
bridge(sessionA, sessionB);


i have 2 legs one is sessionA and sessionB

if i record the sessionA  leg i get only the sessionA voice recorded. How
can i merge both the call (**sessionA and sessionB)**  into single file.

can any one assist me to resolve this problem.

-- 
Thanks with Regards,
N.Baskar

*
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[Freeswitch-users] Fwd: [UniMRCP] PocketSphinx Plugin Available

[Brian West]

I'm about to wire Arsen the donated money for his work.  Remember if  
you haven't sent me what you have said you would send in please paypal br...@freeswitch.org 
 so I can wire it to Arsen.


Thanks,
Brian



Begin forwarded message:

I would like to announce the availability of PocketSphinx ASR plugin  
for UniMRCP server.


PocketSphinx UniMRCP server can be used with an MRCP compliant  
client, which supports JSGF grammar.

Currently supported ASR features are as follows:

Methods:
DEFINE-GRAMMAR
RECOGNIZE
GET-RESULT
START-INPUT-TIMERS
STOP
Events:
START-OF-INPUT
RECOGNITION-COMPLETE
Headers:
Noinput-Timeout
Recognition-Timeout
Completion-Cause
Completion-Reason
Save-Waveform
Grammar: JSGF

For the instructions on how to build and configure PocketSphinx with  
UniMRCP refer to

http://code.google.com/p/unimrcp/wiki/PocketSphinxPlugin

Please note, everything is working now, nevertheless this is basic  
availability only.

I have mostly tested the integrated solution in the following setup
SipPhone -> FreeSWITCH/UniMRCPClient -> UniMRCPServer/PocketSphinx

However it requires further testing in different environments from  
different speakers, e.t.c.

In other words, your feedback is welcome.

Thanks,
--
Arsen Chaloyan
The author of UniMRCP
http://www.unimrcp.org


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Re: [Freeswitch-users] Any advances on T.38 support for FS?

[Michael Jerris]

There was a bit of work towards it but no one has worked on it lately

On Jul 1, 2009, at 4:24 AM, François Delawarde  wrote:

> Is there any work planned for T.38 termination (in mod_fax)?
>
> François.
>
> On Tue, 2009-06-30 at 12:07 -0400, Michael Jerris wrote:
>> We currently support t.38 passthrough only using proxy_media mode.   
>> T.
>> 38 gateway is on the roadmap but not yet close to complete.
>>
>> Mike
>>
>> On Jun 30, 2009, at 5:15 AM, François Delawarde wrote:
>>
>>> Many issues on Asterisk's T.38 (or probably just on T.38?)...
>>>
>>> Could it convince those relying on this "modern" version of a 50yo
>>> technology to switch to and with FreeSwitch?
>>
>>
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Re: [Freeswitch-users] Freeswitch memory usage is too high

[Michael Jerris]

How much memory is it using?  Can you use memstat to see where the  
memory is allocated.

Mike

On Jul 1, 2009, at 8:29 AM, Muhammad Danish Moosa  
 wrote:

> Hi
>
> Freeswitch is being used in a scenario where two endpoints are  
> running traffic with bypass media mode. Performance is good and all  
> things are smooth.
>
> But as the time goes after starting freeswitch, it starts consuming  
> almost whole of memory. Note , freeswitch is being started with - 
> core option, is it related?
>
> If this 99% memory consumption is any red alert, as we can see calls  
> are still connecting fine and all is going as usual.
>
>
> -- 
> Muhammad Danish Moosa
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Re: [Freeswitch-users] Freeswitch memory usage is too high

[Brian West]

You also have a jira http://jira.freeswitch.org/browse/MODAPP-298, It  
looks like you're using lua sql and the backtrace you attached to the  
jira was cut off right before the data I needed to see... can you  
follow up on that ASAP?

It looks like a crash in libmysql from the last line but again I can't  
see the rest of it.

/b

On Jul 1, 2009, at 7:29 AM, Muhammad Danish Moosa wrote:

> Hi
>
> Freeswitch is being used in a scenario where two endpoints are  
> running traffic with bypass media mode. Performance is good and all  
> things are smooth.
>
> But as the time goes after starting freeswitch, it starts consuming  
> almost whole of memory. Note , freeswitch is being started with - 
> core option, is it related?
>
> If this 99% memory consumption is any red alert, as we can see calls  
> are still connecting fine and all is going as usual.
>
>
> -- 
> Muhammad Danish Moosa
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[Freeswitch-users] How to check if a channel variable is present or NULL

[Cavalera Claudio Luigi]

Hello,
I'm trying to implement this kind of logic in the dialplan

if the channel variable sip_refer_to matches regexp
than do action
else if sip_refer_to exists (not NULL) but does not match regexp
than do anti-action
else if sip_refer_to does not exist as a channel_variable (NULL ??)
than do another action

is there a way to check if a channel variable exists or not?
Maybe a special regexp to match?

Thanks,
Claudio


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Re: [Freeswitch-users] freeswitch support PCMU only?

[Nandy Dagondon]

is PCMA enabled in X-Lite, too?

On Wed, Jul 1, 2009 at 9:25 PM, qian ma  wrote:

>
> inbound_codec_negotiation is generous
> and the xlite PCMU is enabled.
>
> my var.xml.conf:
> 
> 
>
>
>
> 2009/7/1 Nandy Dagondon 
>
>> sorry. i mean check the x-lite client if PCMA is enabled?
>>
>>
>> On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon  wrote:
>>
>>> check the value of "inbound_codec_negotiation" in  the sip_profiles/*.xml
>>> files. is it "generous" or "greedy"? you should also check if the endpoint
>>> is offering PCMU.
>>>
>>>
>>>
>>> On Wed, Jul 1, 2009 at 8:27 PM, qian ma  wrote:
>>>
 i want the fs accept the PCMA not PCMU.
 i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
 work. FS only accept PCMU.
 why??




 2009/7/1 Nandy Dagondon 

 you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>
>
> On Wed, Jul 1, 2009 at 3:44 PM, qian ma  wrote:
>
>> thanks for your replies.
>>
>  my var.xml:
>> 
>> 
>>
>>
>> below is the sip trace:
>>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>>
>> 
>>INVITE sip:123...@58.212.219.104 SIP/2.0
>>Via: SIP/2.0/UDP 192.168.1.241:8422
>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>>Max-Forwards: 70
>>Contact: 
>>To: "123456"
>> >
>>From: "9876"
>> >;tag=057de365
>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>CSeq: 1 INVITE
>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>> SUBSCRIBE, INFO
>>Content-Type: application/sdp
>>User-Agent: eyeBeam release 1102u stamp 52345
>>Content-Length: 237
>>
>>v=0
>>o=- 6 2 IN IP4 192.168.1.241
>>s=CounterPath eyeBeam 1.5
>>c=IN IP4 192.168.1.241
>>t=0 0
>>m=audio 57862 RTP/AVP 8 101
>>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>a=fmtp:101 0-15
>>a=rtpmap:101 telephone-event/8000
>>a=sendrecv
>>
>> 
>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>>
>> 
>>SIP/2.0 100 Trying
>>Via: SIP/2.0/UDP 192.168.1.241:8422
>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>From: "9876"
>> >;tag=057de365
>>To: "123456"
>> >
>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>CSeq: 1 INVITE
>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>Content-Length: 0
>>
>>
>> 
>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
>> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
>> sofia/maq/9...@58.212.219.104 entering state [received][100]
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
>> v=0
>> o=- 6 2 IN IP4 192.168.1.241
>> s=CounterPath eyeBeam 1.5
>> c=IN IP4 192.168.1.241
>> t=0 0
>> m=audio 57862 RTP/AVP 8 101
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec
>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf
>> payload to 101
>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec
>> Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20]
>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
>> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
>> sofia/maq/9...@58.212.219.104 [KILL]
>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send
>> signal sofia/maq/9...@58.212.219.104 [BREAK]
>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
>> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
>> sofia/maq/9...@58.212.219.104) State HANGUP
>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
>> sofia/maq/9...@58.212.219.104 hanging up, cause:
>> INCOMPATIBLE_DESTINATION
>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE
>> with: 488
>> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766:
>>
>> 
>>SIP/2.0 488 Not Acceptable 

Re: [Freeswitch-users] Freeswitch memory usage is too high

[Matthew Fong]

bkw,

you said "Downgrading. I suspect its an issue with your lua sql module
not linking to the thread safe client." in the Jira ticket. I'm
curious how one would go about doing this. I use luasql (the default
ubuntu apt-get install) and have a similar memory problem. I suppose I
would need to compile luasql with some sort of flag?

--matt

On Wed, Jul 1, 2009 at 7:18 AM, Brian West wrote:
> You also have a jira http://jira.freeswitch.org/browse/MODAPP-298, It
> looks like you're using lua sql and the backtrace you attached to the
> jira was cut off right before the data I needed to see... can you
> follow up on that ASAP?
>
> It looks like a crash in libmysql from the last line but again I can't
> see the rest of it.
>
> /b
>
> On Jul 1, 2009, at 7:29 AM, Muhammad Danish Moosa wrote:
>
>> Hi
>>
>> Freeswitch is being used in a scenario where two endpoints are
>> running traffic with bypass media mode. Performance is good and all
>> things are smooth.
>>
>> But as the time goes after starting freeswitch, it starts consuming
>> almost whole of memory. Note , freeswitch is being started with -
>> core option, is it related?
>>
>> If this 99% memory consumption is any red alert, as we can see calls
>> are still connecting fine and all is going as usual.
>>
>>
>> --
>> Muhammad Danish Moosa
>> ___
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>> http://www.freeswitch.org
>
>
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Re: [Freeswitch-users] Freeswitch memory usage is too high

[Chris Fowler]

Hi Ray,

This was a problem some time ago (couple of months ago).  Are you running the 
latest build?

Chris.


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Raymond 
Chandler
Sent: Wednesday, July 01, 2009 6:11 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch memory usage is too high

On 07/01/2009 08:29 AM, Muhammad Danish Moosa wrote:
> Hi
>
> Freeswitch is being used in a scenario where two endpoints are running 
> traffic with bypass media mode. Performance is good and all things are 
> smooth.
>
> But as the time goes after starting freeswitch, it starts consuming 
> almost whole of memory.
How much is the "whole"?  You should see the memory usage level off, it 
won't keep growing forever.

-Ray

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Re: [Freeswitch-users] How to check if a channel variable is present or NULL

[Mathieu Rene]

it will match ^$ if the var isnt defined

Math

On 1-Jul-09, at 12:25 PM, Cavalera Claudio Luigi wrote:

> Hello,
> I'm trying to implement this kind of logic in the dialplan
>
> if the channel variable sip_refer_to matches regexp
> than do action
> else if sip_refer_to exists (not NULL) but does not match regexp
> than do anti-action
> else if sip_refer_to does not exist as a channel_variable (NULL ??)
> than do another action
>
> is there a way to check if a channel variable exists or not?
> Maybe a special regexp to match?
>
> Thanks,
> Claudio
>
>
> Internet Email Confidentiality Footer
> -
> La presente comunicazione, con le informazioni in essa contenute e  
> ogni documento o file allegato, e' rivolta unicamente alla/e persona/ 
> e cui e' indirizzata ed alle altre da questa autorizzata/e a  
> riceverla. Se non siete i destinatari/autorizzati siete avvisati che  
> qualsiasi azione, copia, comunicazione, divulgazione o simili basate  
> sul contenuto di tali informazioni e' vietata e potrebbe essere  
> contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia  
> di protezione dei dati personali). Se avete ricevuto questa  
> comunicazione per errore, vi preghiamo di darne immediata notizia al  
> mittente e di distruggere il messaggio originale e ogni file  
> allegato senza farne copia alcuna o riprodurne in alcun modo il  
> contenuto.
>
> This e-mail and its attachments are intended for the addressee(s)  
> only and are confidential and/or may contain legally privileged  
> information. If you have received this message by mistake or are not  
> one of the addressees above, you may take no action based on it, and  
> you may not copy or show it to anyone; please reply to this e-mail  
> and point out the error which has occurred.
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>
>
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Re: [Freeswitch-users] Could this be a bug in the SIP registry?

[mitcheloc]

Jason,

Thanks for the reply. I tried the commands as suggested:

freeswi...@internal> originate user/mythical...@weavver.com 3000
-ERR SUBSCRIBER_ABSENT

2009-07-01 09:43:16 [ERR] switch_xml.c:1555 switch_xml_locate() Error[[error
near line 1]: root tag missing]
freeswi...@internal> 2009-07-01 09:43:16 [WARNING] mod_dptools.c:2364
user_outgoing_channel() Can't find user [mythical...@weavver.com]
2009-07-01 09:43:16 [ERR] switch_ivr_originate.c:1494 switch_ivr_originate()
Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-07-01 09:43:16 [DEBUG] switch_ivr_originate.c:2101
switch_ivr_originate() Originate Resulted in Error Cause: 20
[SUBSCRIBER_ABSENT]

Trying the sofia_contact function:

freeswi...@internal> expand echo ${sofia_contact(profile/
mythical...@weavver.com)}
error/facility_not_subscribed

Here is some output to show that parts of FreeSWITCH do think that the phone
is registered:

freeswi...@internal> sofia status profile internal
=
Nameinternal
Domain Name N/A
DBName  sofia_reg_internal
Pres Hosts
DialplanXML
Context public
Challenge Realm auto_from
RTP-IP  205.134.225.20
SIP-IP  205.134.225.20
URL sip:mod_so...@205.134.225.20:5060
BIND-URLsip:mod_so...@205.134.225.20:5060
HOLD-MUSIC  local_stream://moh
OUTBOUND-PROXY  N/A
CODECS  g7...@32000h,g7...@16000h,G722,PCMU,PCMA,GSM
TEL-EVENT   101
DTMF-MODE   rfc2833
CNG 13
SESSION-TO  0
MAX-DIALOG  0
NOMEDIA false
LATE-NEGfalse
PROXY-MEDIA false
AGGRESSIVENAT   true
STUN-ENABLEDtrue
STUN-AUTO-DISABLE   false
CALLS-IN9
FAILED-CALLS-IN 3
CALLS-OUT   8
FAILED-CALLS-OUT18

Registrations:
=

Call-ID:ZDg4NDU3MjI2ODVlZmZiNGYzZDYzNmRkOTYxMmNhMDY.
User:   mythical...@weavver.com
Contact:"mythicalbox" 
Agent:  eyeBeam release 1102u stamp 52344
Status: Registered(TCP-NAT)(unknown) EXP(2009-07-01 11:33:18)
Host:   duck.weavver.com
IP: 64.183.110.250
Port:   8443
Auth-User:  mythicalbox
Auth-Realm: weavver.com

=


FreeSWITCH and my Softphone (eyeBeam) shows me as registered when polling
for registrations but when trying to connect the call FreeSWITCH is not
seeing it as registered.

Any more ideas?


Thanks again!

On Wed, Jul 1, 2009 at 1:45 AM, Jason White  wrote:

> Jason White  wrote:
> > originate user/1...@example.com 3000
> > to connext u...@example.com to extension 3000.
>
> That should read "to connect 1...@example.com to extension 3000".
>
>
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Re: [Freeswitch-users] Could this be a bug in the SIP registry?

[Michael Collins]

On Wed, Jul 1, 2009 at 10:04 AM, mitcheloc  wrote:

> Jason,
>
> Thanks for the reply. I tried the commands as suggested:
>
> freeswi...@internal> originate user/mythical...@weavver.com 3000
> -ERR SUBSCRIBER_ABSENT
>

I suspect the following line is a clue:

>
> 2009-07-01 09:43:16 [ERR] switch_xml.c:1555 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
>

Can you confirm the XML that is getting read?
-MC
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Re: [Freeswitch-users] Could this be a bug in the SIP registry?

[Brian West]

What does 'sofia status' say?

expand echo ${sofia_contact(internal/mythical...@weavver.com)}  <--  
notice I put the profile name instead of the word "profile"


/b



On Jul 1, 2009, at 12:04 PM, mitcheloc wrote:


Jason,


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Re: [Freeswitch-users] Could this be a bug in the SIP registry?

[mitcheloc]

Brian,
Oh yay! Good catch.. it gave me output this time, and I could make a call
using it:

sofia/internal/sip:mythical...@64.183.110.250:9136
;rinstance=24c9b78f5fc6c759;transport=TCP;fs_nat=yes;fs_path=sip%3Amythicalbox%4064.183.110.250%3A9136%3Brinstance%3D24c9b78f5fc6c759%3Btransport%3DTCP

That is definitely not what I'd been trying!

Here is sofia status in case you still want it:

freeswi...@internal> sofia status
 Name  Type   Data
   State
=
 internal   profile
sip:mod_so...@205.134.225.20:5060RUNNING (0)
 external   profile
sip:mod_so...@205.134.225.20:5080RUNNING (0)
  example.com   gateway
sip:joeu...@example.com NOREG
internal-ipv6   profile   sip:mod_so...@[::1]:5060
   RUNNING (0)
  default alias   internal
   ALIASED
  nat alias   external
   ALIASED
 outbound alias   external
   ALIASED
   205.134.225.20 alias   external
   ALIASED
=
3 profiles 4 aliases


Thank you!!

On Wed, Jul 1, 2009 at 10:14 AM, Brian West  wrote:

> What does 'sofia status' say?
> expand echo ${sofia_contact(internal/mythical...@weavver.com)}  <-- notice
> I put the profile name instead of the word "profile"
>
> /b
>
>
>
> On Jul 1, 2009, at 12:04 PM, mitcheloc wrote:
>
> Jason,
>
>
>
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[Freeswitch-users] Freeswitch memory usage is too high

[Muhammad Danish Moosa]

Hi Brian

My customer is now using lua odbc ( not lua mysql anymore) and problem
mentioned in jira is resolved now.

*
http://www.mail-archive.com/freeswitch-...@lists.freeswitch.org/msg01352.html
*
*
*This seems to answer my question, rite?

BTW , starting FS with following

ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 99
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 244
ulimit -l unlimited
ulimit -a


Date: Wed, 1 Jul 2009 09:18:57 -0500
From: Brian West 
Subject: Re: [Freeswitch-users] Freeswitch memory usage is too high
To: freeswitch-users@lists.freeswitch.org
Message-ID: 
Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes

You also have a jira http://jira.freeswitch.org/browse/MODAPP-298, It
looks like you're using lua sql and the backtrace you attached to the
jira was cut off right before the data I needed to see... can you
follow up on that ASAP?

It looks like a crash in libmysql from the last line but again I can't
see the rest of it.

/b

On Jul 1, 2009, at 7:29 AM, Muhammad Danish Moosa wrote:

> Hi
>
> Freeswitch is being used in a scenario where two endpoints are
> running traffic with bypass media mode. Performance is good and all
> things are smooth.
>
> But as the time goes after starting freeswitch, it starts consuming
> almost whole of memory. Note , freeswitch is being started with -
> core option, is it related?
>
> If this 99% memory consumption is any red alert, as we can see calls
> are still connecting fine and all is going as usual.
>
>
> --
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Re: [Freeswitch-users] Freeswitch memory usage is too high

[Brian West]

If the problem is resolved please follow up on the jira.

/b

On Jul 1, 2009, at 12:42 PM, Muhammad Danish Moosa wrote:

> My customer is now using lua odbc ( not lua mysql anymore) and  
> problem mentioned in jira is resolved now.
>


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Re: [Freeswitch-users] freeswitch support PCMU only?

[qian ma]

yes,PCMA enabled in x-lite.

doesn't work.

FS accept PCMU only.

2009/7/2 Nandy Dagondon 

> is PCMA enabled in X-Lite, too?
>
>
> On Wed, Jul 1, 2009 at 9:25 PM, qian ma  wrote:
>
>>
>> inbound_codec_negotiation is generous
>> and the xlite PCMU is enabled.
>>
>> my var.xml.conf:
>>  
>> 
>>
>>
>>
>> 2009/7/1 Nandy Dagondon 
>>
>>> sorry. i mean check the x-lite client if PCMA is enabled?
>>>
>>>
>>> On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon  wrote:
>>>
 check the value of "inbound_codec_negotiation" in  the
 sip_profiles/*.xml files. is it "generous" or "greedy"? you should also
 check if the endpoint is offering PCMU.



 On Wed, Jul 1, 2009 at 8:27 PM, qian ma  wrote:

> i want the fs accept the PCMA not PCMU.
> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
> work. FS only accept PCMU.
> why??
>
>
>
>
> 2009/7/1 Nandy Dagondon 
>
> you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>>
>>
>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote:
>>
>>> thanks for your replies.
>>>
>>  my var.xml:
>>> 
>>> 
>>>
>>>
>>> below is the sip trace:
>>>  recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>>>
>>> 
>>>INVITE sip:123...@58.212.219.104 SIP/2.0
>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport
>>>Max-Forwards: 70
>>>Contact: 
>>>To: "123456"
>>> >
>>>From: "9876"
>>> >;tag=057de365
>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>CSeq: 1 INVITE
>>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>> SUBSCRIBE, INFO
>>>Content-Type: application/sdp
>>>User-Agent: eyeBeam release 1102u stamp 52345
>>>Content-Length: 237
>>>
>>>v=0
>>>o=- 6 2 IN IP4 192.168.1.241
>>>s=CounterPath eyeBeam 1.5
>>>c=IN IP4 192.168.1.241
>>>t=0 0
>>>m=audio 57862 RTP/AVP 8 101
>>>a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>>a=fmtp:101 0-15
>>>a=rtpmap:101 telephone-event/8000
>>>a=sendrecv
>>>
>>> 
>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505:
>>>
>>> 
>>>SIP/2.0 100 Trying
>>>Via: SIP/2.0/UDP 192.168.1.241:8422
>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104
>>>From: "9876"
>>> >;tag=057de365
>>>To: "123456"
>>> >
>>>Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY.
>>>CSeq: 1 INVITE
>>>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M
>>>Content-Length: 0
>>>
>>>
>>> 
>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel
>>> sofia/maq/9...@58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel
>>> sofia/maq/9...@58.212.219.104 entering state [received][100]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP:
>>> v=0
>>> o=- 6 2 IN IP4 192.168.1.241
>>> s=CounterPath eyeBeam 1.5
>>> c=IN IP4 192.168.1.241
>>> t=0 0
>>> m=audio 57862 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec
>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf
>>> payload to 101
>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec
>>> Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20]
>>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup
>>> sofia/maq/9...@58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal
>>> sofia/maq/9...@58.212.219.104 [KILL]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send
>>> signal sofia/maq/9...@58.212.219.104 [BREAK]
>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 (
>>> sofia/maq/9...@58.212.219.104) Running State Change CS_HANGUP
>>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 (
>>> sofia/maq/9...@58.212.219.104) State HANGUP
>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel
>>> sofia/maq/9...@58.212.219.104 hanging up, cause:
>>> INCOMPATIBLE_DESTINATION
>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Res

Re: [Freeswitch-users] freeswitch support PCMU only?

[Brian West]

Thats bull... I just did PCMA all morning testing!  Your config is  
wrong.

/b

On Jul 1, 2009, at 1:33 PM, qian ma wrote:

> yes,PCMA enabled in x-lite.
>
> doesn't work.
>
> FS accept PCMU only.


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[Freeswitch-users] sound_prefix is changed

[Diego Toro]

hi all, 
I am working with release Pre9, I have a problem now with say module, the 
sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for 
default_language to 'es'.
 
I checked c code on switch_ivr_say the value of sound_prefix is changed always
 
Diego


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Re: [Freeswitch-users] freeswitch support PCMU only?

[Nik Middleton]

I'm ONLY use PCMA, so I would agree with Brian

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 01 July 2009 20:35
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] freeswitch support PCMU only?

Thats bull... I just did PCMA all morning testing!  Your config is  
wrong.

/b

On Jul 1, 2009, at 1:33 PM, qian ma wrote:

> yes,PCMA enabled in x-lite.
>
> doesn't work.
>
> FS accept PCMU only.


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[Freeswitch-users] SIP re-invite / bypass_media

[Phillip Jones]

Hi there,

I was wondering whether it is possible to have FreeSwitch go into
bypass_media mode on demand?

For instance, leg a bridges to leg b - leg b is invited to accept the call
by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute
the media) after the one is pressed.

Currently I am issuing the following from my js script that prompts for the
1:

session.apiExecute("uuid_media",session.uuid);

Not working however.

Any help to get me going would be appreciated.

Thanks

Phillip Jones.
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Re: [Freeswitch-users] sound_prefix is changed

[Anthony Minessale]

what is the problem exactly? what do you want it to change to?


On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro  wrote:

> hi all,
> I am working with release Pre9, I have a problem now with say module, the
> sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for
> default_language to 'es'.
>
> I checked c code on switch_ivr_say the value of sound_prefix is changed
> always
>
> Diego
>
>
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>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com 
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sip:8...@conference.freeswitch.org 
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[Freeswitch-users] G723 timer problem

[Muhammad Shahzad]

Hi,

I am using FS svn revision 14046 and trying to send call from SIP Dialer to
a SIP gateway using G723 in passthrough mode. Everything works perfect and
destination rings but then call drops with following error on FS CLI,


2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to use
ptime 30 but what they meant to say was 60
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who
knows what will happen..
2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec G723
Exists but not at the desired implementation. 8000hz 60ms


Is there any work around for this or i have downgrade my server back to
Asterisk. :'-(

Thank you.


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
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MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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Re: [Freeswitch-users] G723 timer problem

[Brian West]

You have two choices... set codec neg. to scrooge or get a provider  
that doesn't lie about the ptime in their SDP.


/b

On Jul 1, 2009, at 4:04 PM, Muhammad Shahzad wrote:


Hi,

I am using FS svn revision 14046 and trying to send call from SIP  
Dialer to a SIP gateway using G723 in passthrough mode. Everything  
works perfect and destination rings but then call drops with  
following error on FS CLI,



2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to  
use ptime 30 but what they meant to say was 60
This issue has so far been identified to happen on the following  
broken platforms/devices:

Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so  
broken who knows what will happen..
2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec  
G723 Exists but not at the desired implementation. 8000hz 60ms



Is there any work around for this or i have downgrade my server back  
to Asterisk. :'-(


Thank you.


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Re: [Freeswitch-users] sound_prefix is changed

[Diego Toro]

Hello, the problem is that sound_prefix value is ignored and it's changed to 
SWITCH_GLOBAL_dirs.base_dir/sounds/language  (with language=en), so audio files 
are not found .

Diego
 

--- On Wed, 7/1/09, Anthony Minessale  wrote:


From: Anthony Minessale 
Subject: Re: [Freeswitch-users] sound_prefix is changed
To: freeswitch-users@lists.freeswitch.org
Date: Wednesday, July 1, 2009, 3:36 PM


what is the problem exactly? what do you want it to change to?



On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro  wrote:






hi all, 
I am working with release Pre9, I have a problem now with say module, the 
sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for 
default_language to 'es'.
 
I checked c code on switch_ivr_say the value of sound_prefix is changed always
 
Diego

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-- 
Anthony Minessale II

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

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-Inline Attachment Follows-


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Re: [Freeswitch-users] G723 timer problem

[Brian West]

I'm sorry about my response... I had overlooked that I only did one  
30ms implementation in mod_g723_1.c, Anthony added some more to the  
list so it might actually work correctly.


Thanks,
Brian

On Jul 1, 2009, at 4:04 PM, Muhammad Shahzad wrote:


Hi,

I am using FS svn revision 14046 and trying to send call from SIP  
Dialer to a SIP gateway using G723 in passthrough mode. Everything  
works perfect and destination rings but then call drops with  
following error on FS CLI,



2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to  
use ptime 30 but what they meant to say was 60
This issue has so far been identified to happen on the following  
broken platforms/devices:

Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so  
broken who knows what will happen..
2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec  
G723 Exists but not at the desired implementation. 8000hz 60ms



Is there any work around for this or i have downgrade my server back  
to Asterisk. :'-(


Thank you.


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Re: [Freeswitch-users] sound_prefix is changed

[Anthony Minessale]

did you put them in that spot? the say app needs them to be in that spot.


On Wed, Jul 1, 2009 at 4:10 PM, Diego Toro  wrote:

> Hello, the problem is that sound_prefix value is ignored and it's changed
> to SWITCH_GLOBAL_dirs.base_dir/sounds/language  (with language=en), so audio
> files are not found .
> Diego
>
>
> --- On *Wed, 7/1/09, Anthony Minessale *wrote:
>
>
> From: Anthony Minessale 
> Subject: Re: [Freeswitch-users] sound_prefix is changed
> To: freeswitch-users@lists.freeswitch.org
> Date: Wednesday, July 1, 2009, 3:36 PM
>
>
> what is the problem exactly? what do you want it to change to?
>
>
> On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro 
> http://us.mc335.mail.yahoo.com/mc/compose?to=dft...@yahoo.com>
> > wrote:
>
>>   hi all,
>> I am working with release Pre9, I have a problem now with say module, the
>> sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for
>> default_language to 'es'.
>>
>> I checked c code on switch_ivr_say the value of sound_prefix is changed
>> always
>>
>> Diego
>>
>>
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>
>
> --
> Anthony Minessale II
>
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>
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Re: [Freeswitch-users] sound_prefix is changed

[Diego Toro]

The audio files are for instance on ...us/callie/digits/8000, but the last 
1.0.4 pre9 version has changed, it ignores the var sound_prefix setting to 
SWITCH_GLOBAL_dirs.base_dir/sounds/en the path to audio files.
 
 That change is on the switch_ivr_say function
 
Diego.

--- On Wed, 7/1/09, Anthony Minessale  wrote:


From: Anthony Minessale 
Subject: Re: [Freeswitch-users] sound_prefix is changed
To: freeswitch-users@lists.freeswitch.org
Date: Wednesday, July 1, 2009, 4:31 PM


did you put them in that spot? the say app needs them to be in that spot.



On Wed, Jul 1, 2009 at 4:10 PM, Diego Toro  wrote:






Hello, the problem is that sound_prefix value is ignored and it's changed to 
SWITCH_GLOBAL_dirs.base_dir/sounds/language  (with language=en), so audio files 
are not found .

Diego
 

--- On Wed, 7/1/09, Anthony Minessale  wrote:


From: Anthony Minessale 
Subject: Re: [Freeswitch-users] sound_prefix is changed
To: freeswitch-users@lists.freeswitch.org
Date: Wednesday, July 1, 2009, 3:36 PM





what is the problem exactly? what do you want it to change to?



On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro  wrote:






hi all, 
I am working with release Pre9, I have a problem now with say module, the 
sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for 
default_language to 'es'.
 
I checked c code on switch_ivr_say the value of sound_prefix is changed always
 
Diego

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Anthony Minessale II

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Re: [Freeswitch-users] SIP re-invite / bypass_media

[Anthony Minessale]

try
apiExecute("uuid_media", "off " + session.uuid);



On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones  wrote:

> Hi there,
>
> I was wondering whether it is possible to have FreeSwitch go into
> bypass_media mode on demand?
>
> For instance, leg a bridges to leg b - leg b is invited to accept the call
> by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute
> the media) after the one is pressed.
>
> Currently I am issuing the following from my js script that prompts for the
> 1:
>
> session.apiExecute("uuid_media",session.uuid);
>
> Not working however.
>
> Any help to get me going would be appreciated.
>
> Thanks
>
> Phillip Jones.
>
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Re: [Freeswitch-users] PCMU fallback for T.38

[Mike Fedyk]

On Fri, Mar 20, 2009 at 7:46 PM, Steve Underwood  wrote:

> Gabriel Kuri wrote:
> > once the FAX tone is detected on the PSTN side, FS receives a T.38
> > re-INVITE from the provider and FS sends back a 488/Not Acceptable
> > (proxy_media=false). at that point the provider than attempts fallback
> > to PCMU with another reINVITE ...
> >
>
> This part is interesting, and the subject of a discussion we had
> recently. A number of systems try that second re-invite after a 488, but
> the SIP specs say the call is pretty much dead after the 488 message is
> exchanged. Are they just hoping that maybe the other end will be
> non-compliant enough to keep the call alive, and recover its media mode,
> or haven't they read the specs?
>
> Steve


I am interested in this later document.  From what I can see there is
rfc3261 and at least one other RFC (and one draft that I have found after
about 30 minutes of searching) that support that a 488 response can be
recovered from when it is a response to a reinvite (ie, the dialog is
already in place and there is something to fall back to).

Where does it say that a 488 has to end a dialog?  From what I understand
there are not any 4xx codes that by themselves terminate a dialog (except
where it terminates the last leg of a call -- much like unlink() in unix).

draft-ietf-sipping-realtimefax-01 says:

> 6.2. Unsuccessful T.38 fax scenario -
>
> - 488/606 rsp & G.711 fallback
>
>
>This section represents an unsuccessful SIP T.38 fax call:  when the
>emitting gateway does not support T.38 fax relay, it SHOULD respond
>with either a ��488 Not Acceptable Here�� response or a ��606 Not
>
>Acceptable�� response to indicate that some aspects of the session
>description are not acceptable.  The terminating gateway SHOULD
>react by proposing a fallback to G.711 fax pass-through with special
>
>codec characteristics -
>  -silence suppression OFF.  The message details
>in this section make use of the generic SDP attribute silenceSupp
>defined in RFC3108
>
>
 rfc3261 section 3 says:

> During the session, either Alice or Bob may decide to change the
>
>characteristics of the media session.  This is accomplished by
>sending a re-INVITE containing a new media description.  This re-
>INVITE references the existing dialog so that the other party knows
>that it is to modify an existing session instead of establishing a
>
>new session.  The other party sends a 200 (OK) to accept the change.
>The requestor responds to the 200 (OK) with an ACK.  If the other
>party does not accept the change, he sends an error response such as
>
>488 (Not Acceptable Here), which also receives an ACK.  However, the
>failure of the re-INVITE does not cause the existing call to fail -
>the session continues using the previously negotiated
>characteristics.  Full details on session modification are in Section
>
>14.
>
> section 14.1 says:

>If a UA receives a non-2xx final response to a re-INVITE, the session
>
>parameters MUST remain unchanged, as if no re-INVITE had been issued.
>Note that, as stated in Section 12.2.1.2, if the non-2xx final
>response is a 481 (Call/Transaction Does Not Exist), or a 408
>(Request Timeout), or no response at all is received for the re-
>
>INVITE (that is, a timeout is returned by the INVITE client
>transaction), the UAC will terminate the dialog.
>
>
 rfc4497 says:

> 8.5.  Request to Change Media Characteristics
>
>If after a call has been successfully established the gateway
>receives a SIP INVITE request to change the media characteristics of
>the call in a way that would be incompatible with the bearer
>capability in use within the PISN, the gateway SHALL send back a SIP
>488 (Not Acceptable Here) response and SHALL NOT change the media
>characteristics of the existing call.
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[Freeswitch-users] Language Handling: call for assistance

[Michael Collins]

Hello all!

There's been some discussion lately on how to handle multiple languages,
specifically with the *say* application. We would like some input from the
community on how to handle multiple languages and sound files. Anthony notes
that the say application needs to build the path to the sound files by using
the ${sound_prefix} and ${lang} variables. Some have asked about countries
or language variants, like European Portugese vs. Brazilian Portugese. These
are good questions.

>From the community we need input. If you have experience with multiple
languages in a telephony environment then please give us your suggestions.
How would you like to see the say application handle various languages and
dialects? Please give us your helpful suggestions.

Thanks,
Michael
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[Freeswitch-users] How to know my gateway registering is successed??

[Brad Tuan]

Another question, Where does FS keep these information??

In *.db or somewhere??
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Re: [Freeswitch-users] How to know my gateway registering is successed??

[Michael Collins]

Are you looking for something more than what "sofia status" at the CLI
shows?
-MC

On Wed, Jul 1, 2009 at 6:05 PM, Brad Tuan  wrote:

> Another question, Where does FS keep these information??
>
> In *.db or somewhere??
>
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Re: [Freeswitch-users] How to know my gateway registering is successed??

[Jason White]

Brad Tuan  wrote:
> Another question, Where does FS keep these information??
> 
> In *.db or somewhere??

It's a hash table in memory. See sofia_reg_find_gateway__ in sofia_reg.c for
the code that performs the hash table lookup and returns a pointer to the
structure with all of the fields in it.



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[Freeswitch-users] How to remove the IP from the SIP caller id number

[Mitchel Constantin]

Hello,
I'm working on configuring my FreeSWITCH and would like to set the caller id
number like this in dialplan/default.xml:




I wonder if this is a problem with eyeBeam.. When the call is received the
CID is like this:

John Doe
john...@weavver.com@205.134.225.20

205.134.225.20 is the EXT IP of the switch

I'd like to remove the 205.134.225.20 from the Caller ID if possible. From
what I understand the CID without the IP would still point to the correct
server.


TIA,

-- 
Mitchel Constantin
Weavver, Inc.
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Re: [Freeswitch-users] How to remove the IP from the SIP caller id number

[Jason White]

Mitchel Constantin  wrote:
> I'm working on configuring my FreeSWITCH and would like to set the caller id
> number like this in dialplan/default.xml:
> 
> 
> 
> 
> I wonder if this is a problem with eyeBeam.. When the call is received the
> CID is like this:
> 
> John Doe
> john...@weavver.com@205.134.225.20
> 
> 205.134.225.20 is the EXT IP of the switch

I suspect the other end (whatever device you are calling from FreeSWITCH) is
adding the IP address to the caller id. However, I am no SIP expert and may be
wrong, but you can confirm this by doing a SIP trace on the device that
receives the call (or on its local network via packet capture) to discover
what FreeSWITCH is sending out.


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