[Freeswitch-users] Yet another question about A500 + FS

2009-08-23 Thread Vassil Panayotov 
Hi,

I managed to get our A500 running with FreeSWITCH 1.0.4 stable using wanpipe
3.4.4 drivers. But now I have another problem...
I want to originate calls through event socket, and I only want to receive
ANSWERED(+OK) reply when the user actually answers.

Now the situation is:


originate openzap/1/a/123456 023
2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:
CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456]
Ci=[00]
2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel
OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer
OpenZAP/1:1/123456!
API CALL [originate(openzap/1/a/123456 023)] output:
+OK f8fca2be-8fa7-11de-9076-511e29dfc082

2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer
OpenZAP/1:1/123456 to xml[...@default]
freeswi...@emo-voip 2009-08-23 08:44:06.743475 [INFO]
mod_dialplan_xml.c:315 Processing FreeSWITCH-023 in context default
2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
[OpenZAP/1:1/123456] has been answered
2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup
OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT (N):
CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086 Session 2
(OpenZAP/1:1/123456) Ended
2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close Channel
OpenZAP/1:1/123456 [CS_DESTROY]


Extension 023 is an IVR. As you can see FreeSWITCH answers the call
(2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
[OpenZAP/1:1/123456] has been answered) 20 seconds before user actually pick
up the phone (2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX
EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5).

So Sangoma drivers/daemons report the events correctly.
How can I set FreeSWITCH to answer after receiving RX EVENT (N):
CALL_ANSWERED from the driver?

Thank you,
V. Panayotov
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Re: [Freeswitch-users] Inboud Call Queue

2009-08-23 Thread Diego Viola 
Hi guys,

I was wondering if some of you run FreeSWITCH on a call center
environment, I ask this because I plan to do that soon and I was
wondering how well mod_fifo works for queues, etc.

Thanks,

Diego

On Thu, May 7, 2009 at 6:08 AM, Saeed Ahmedsaeedahmad1...@gmail.com wrote:
 Thanks Seven I’ll try it very soon.



 

 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of seven
 Sent: Thursday, May 07, 2009 5:42 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Inboud Call Queue



 See this:



 http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo





 On May 6, 2009, at 10:15 PM, Saeed Ahmed wrote:

 Thanks Guys



 

 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] on
 Behalf Of Anthony Minessale
 Sent: Wednesday, May 06, 2009 3:57 PM
 To: freeswitch-us...@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Inboud Call Queue



 I worked on the patch and added it to trunk rev 13240


 On Wed, May 6, 2009 at 7:53 AM, dujinfang dujinf...@gmail.com wrote:

 The patch haven't been merged into trunk. It should be as easy as execute
 the following command in the FS source code root dir:



 patch  /tmp/the_patch_file_name.diff



 I will post an example on the wiki when I finished, hope be soon.



 On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote:


 Hi Seven,

 I am exactly looking for this functionality.

 Please let me know when you are finished with new queue manager app. I’ll
 try it in my call center.

 Regarding Patch: is it already part of SVN trunk? If not then could you help
 me how to install it, I have no programming background.

 Many Thanks.



 

 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] on
 Behalf Of seven
 Sent: Wednesday, May 06, 2009 4:17 AM
 To: freeswitch-us...@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Inboud Call Queue





 On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote:



 Hi Michael,

 Thanks for a quick reply.

 I would definitely create a test environment, but my question is that will
 it work in required way?

 I read that in Mod_fifo agent has to call in queue but I need that all
 incoming calls are automatically distributed between available agents or if
 all are busy then should go to voicemail.

 I'm working on a call center like queue scenario right now, I'm pretty sure
 it call automatically distributed to available agents, but the customer will
 stay in the queue if all agents are busy by default. You can bind a key to
 the channel and play a message repeatedly to guide the customer to voicemail
 by press a key.



 Also maybe you need this patch to make the fifo works as desired.



 http://jira.freeswitch.org/browse/MODAPP-272



 I would join IRC for further assistance.



 Thanks.



 

 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] on
 Behalf Of Michael Collins
 Sent: Tuesday, May 05, 2009 7:19 PM
 To: freeswitch-us...@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Inboud Call Queue





 On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed saeedahmad1...@gmail.com
 wrote:

 Hi All,

 In an inbound call center scenario is it possible that customers calls in
 and calls are distributed between online (who are registered on FS and in
 idle state) agents. I saw some on going discussion on list where it looks
 that currently it’s not possible but I am newbie so maybe I didn’t
 understand it well. If it’s possible then please give me a start point that
 how can I implement it.

 I would start here:
 http://wiki.freeswitch.org/wiki/Mod_fifo

 I strongly recommend that you set up a FreeSWITCH server and play around
 with it so that you can learn the pros and cons of using the FIFO queues. It
 would be best if you could set up a few phones and set them as FIFO agents
 and then have someone help you make test calls so that you can emulate your
 CC environment.

 Also, you might want to join us on IRC: #freeswitch on irc.freenode.net -
 there are several users who've had real world experience with mod_fifo and
 they might be in a good position to answer your questions real-time.

 -MC

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Re: [Freeswitch-users] Inboud Call Queue

2009-08-23 Thread Jason White 
Diego Viola diego.vi...@gmail.com wrote:
 I was wondering if some of you run FreeSWITCH on a call center
 environment, I ask this because I plan to do that soon and I was
 wondering how well mod_fifo works for queues, etc.

This was mentioned on the list once before, and it might be what you want:
http://wiki.opencsm.org/wiki/index.php/Spice_Telephony
(Spice Telephony).


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[Freeswitch-users] Starting Freeswitch using Intercom

2009-08-23 Thread Edmar Cruz 

Hi,

  I dont know how can I start Freeswitch using Intercom device, can you help
me on this? Is there an alternative software like X-Lite but only when I
press call on the Intercom device? 

Thanks, 
Edmar
-- 
View this message in context: 
http://www.nabble.com/Starting-Freeswitch-using-Intercom-tp25100943p25100943.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Inboud Call Queue

2009-08-23 Thread Diego Viola 
Looks nice, is anyone running that in production?

On Sun, Aug 23, 2009 at 3:08 AM, Jason White ja...@jasonjgw.net wrote:

 Diego Viola diego.vi...@gmail.com wrote:
  I was wondering if some of you run FreeSWITCH on a call center
  environment, I ask this because I plan to do that soon and I was
  wondering how well mod_fifo works for queues, etc.

 This was mentioned on the list once before, and it might be what you want:
 http://wiki.opencsm.org/wiki/index.php/Spice_Telephony
 (Spice Telephony).


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Re: [Freeswitch-users] MFC-R2 support for FreeSWITCH

2009-08-23 Thread Arnaldo de Moraes Pereira 
Thanks a lot, moy, this is great. I'll check to see if there's somewhere I
can test it.

On Sun, Aug 23, 2009 at 2:40 AM, Diego Viola diego.vi...@gmail.com wrote:

 Nice work, keep up the great work :).

 On Fri, Aug 21, 2009 at 6:29 PM, Moises Silvamoises.si...@gmail.com
 wrote:
  So, I finally took some days to put up OpenR2 working with OpenZAP, which
  means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
  support for. Including Mexico, Brazil, Argentina and others. The stack
 has
  been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers
 most
  countries that users may be interested in, support for new variants will
 be
  added on-demand only (in any case users can always tweak the advanced
  configuration file to create their own variants as a last resort).
  I created a wiki page to illustrate the basic
  setup: http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
  Now is time for testing. I just did minimal testing on my development
  environment, no serious testing, and I know that some stuff is not
 working
  at this point (I had some issues with variable length DNIS and ANI) which
  should be fixed soon.
  If anyone around happens to have an R2 link and wants to test R2 support
 in
  OpenZAP, I can give them a hand with the configuration and any issues you
  may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
  #openzap as moy.
 
  --
  Moises Silva
  Software Developer
  Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
 9T3
  Canada
  t. 1 905 474 1990 x 128 | e. m...@sangoma.com
 
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-- 
Arnaldo M Pereira
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[Freeswitch-users] FXO and analogue phones

2009-08-23 Thread Merul Patel 
I have a Freeswitch setup working on an Alix embedded platform in  
conjunction with a USB FXO device from Sangoma. My goal is to be able  
to either answer incoming calls on a softphone or on a POTS handset  
elsewhere in the building, and to also be able to make outgoing calls  
from either. For clarity, the analogue line has two physical  
extensions, one connected to the POTS and the other to the FXO.


I can make and receive calls fine, but have problems when the call is  
answered on the POTS handset.


Here is the dialplan I initially used  in /opt/freeswitch/conf/ 
dialplans/public/01_incoming.xml:


include
  extension name=public_did
condition field=${strftime(%w)} expression=^(\d)$
  !-- There seems to be a delay of 7 seconds from when FS starts  
dealing with the call and from when it starts ringing --

  action application=sleep data=23000/
  action application=set data=domain_name=$${domain}/
  action application=transfer data=1001 XML default/
/condition
  /extension
/include

It's pretty basic, and if the softphone is not registered or does not  
answer then the call goes to voicemail. However the call will always  
go to voicemail, and the voicemail application will begin to execute  
after the call has been answered on the POTS handset.


I've been trying to make the dialplan more useful, by having it ring  
the softphone immediately, and only transfer the call to the voicemail  
application if the line is still ringing. I'm in the UK, hence my  
choice of frequencies in the tone_detect application:


include
  extension name=public_did
condition field=${strftime(%w)} expression=^(\d)$
  !-- There seems to be a delay of 7 seconds from when FS starts  
dealing with the call and from when it starts ringing --

  action application=set data=call_timeout=23/
  action application=set data=continue_on_fail=true/
  action application=set data=hangup_after_bridge=true/
  action application=bridge data=sofia/internal/1001%$$ 
{domain}/

  action application=sleep data=23000/
  action application=tone_detect data=ring 400,450 r +5000  
set RING=true/

  action application=transfer data=public_answer_and_email/
/condition
  /extension

extension name=public_answer_and_email
condition field=RING expression=true
   action application=answer/
   action application=voicemail data=default $${domain} 1001/
/condition
  /extension
/include

Unfortunately, this is not working, and the logs are not yielding  
anything the is helpful to me. Is my use of the tone_detect  
application and the basic dialplan correct?


Merul

smime.p7s
Description: S/MIME cryptographic signature
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[Freeswitch-users] problem compiling esl for use with freepbx v3

2009-08-23 Thread Harondel J. Sibble 
Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, 
then went to install FreePBX v3, I've gotten all the prerequisities in the 
wizard fixed except for ESL

As per

http://wiki.freeswitch.org/wiki/Event_Socket_Library
http://wiki.freeswitch.org/wiki/Event_Socket

I go into my FS source dir

/home/sibbleh/freeswitch-1.0.4/libs/esl

Run make and then sudo make phpmod-install 

and I get 


$ sudo make phpmod-install
make MYLIB=../libesl.a SOLINK=-shared -Xlinker -x CFLAGS=-
I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb 
-I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=-
I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb 
-I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable 
CXX_CFLAGS= -C php
make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g 
-ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -
I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
I/usr/include/php5/Zend -I/usr/include/php5/ext -
I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -
Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
cc1plus: warnings being treated as errors
esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1047: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1073: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:1141: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1172: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1198: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1234: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1269: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1294: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**, zval*, 
int)':
esl_wrap.cpp:1346: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1403: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1441: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1478: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1508: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1538: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1571: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1611: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1644: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1674: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1704: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1744: error: format not a string

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

2009-08-23 Thread Peter P GMX 
Hello Anthony,

I set p...@30i,p...@30i and I can see in the logs that PCMA is used.
However ptime is set to 20 msec as shown in the Logs:

2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
s=call
c=IN IP4 217.xx.xx.xxx
t=0 0
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7079

2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404
(sofia/internal/02xx...@fs1.my.domain) State NEW
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[G722:9:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/internal/02xx...@fs1.my.domain PCMA/8000 20 ms 160 samples
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
payload to 101

Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is horrible.

Best regards
Peter

Anthony Minessale schrieb:
 try setting FS to 30ms too

 edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it
 looks like p...@30i

 from:

   X-PRE-PROCESS cmd=set
 data=global_codec_prefs=g7...@32000h,g7...@16000h,G722,PCMU,PCMA,GSM/
   X-PRE-PROCESS cmd=set data=outbound_codec_prefs=PCMU,PCMA,GSM/

 to:

   X-PRE-PROCESS cmd=set
 data=global_codec_prefs=g7...@32000h,g7...@16000h,G722,p...@30i,p...@30i,GSM/
   X-PRE-PROCESS cmd=set
 data=outbound_codec_prefs=p...@30i,p...@30i,GSM/


 On Fri, Aug 21, 2009 at 1:38 PM, Brian West br...@freeswitch.org
 mailto:br...@freeswitch.org wrote:

 You can ship me one whois bkw.org http://bkw.org, I can add it
 to my lab.

 /b

 On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:

 
  BTW: We can ship you a FritzBox if you need one for testing.


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[Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 
I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp capable 
endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone client, 
2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i (symbian 
s60) and an O2 Xda Flame (windows mobile 5).

All 3 endpoints are registered with FS using the default extensions of 1000-
1003

With global_setvar zrtp_secure_media=true the zrtp negotiation between end 
points happens but the SAS never matches,below is console output for a call 
between 2 of the endpoints 


2009-08-23 14:10:17.643073 [NOTICE] mod_sofia.c:1509 Pre-Answer 
sofia/internal/1...@10.12.13.45!
2009-08-23 14:10:21.257568 [NOTICE] sofia.c:3794 Channel 
[sofia/internal/sip:1...@10.12.13.166:5062] has been answered
2009-08-23 14:10:21.275521 [NOTICE] switch_ivr_originate.c:2015 Channel 
[sofia/internal/1...@10.12.13.45] has been answered
2009-08-23 14:10:22.232053 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
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..
2009-08-23 14:11:34.496118 [NOTICE] sofia.c:322 Hangup 
sofia/internal/sip:1...@10.12.13.166:5062 [CS_EXCHANGE_MEDIA] 
[NORMAL_CLEARING]
2009-08-23 14:11:34.512100 [NOTICE] switch_ivr_bridge.c:1016 Hangup 
sofia/internal/1...@10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1086 Session 16 
(sofia/internal/sip:1...@10.12.13.166:5062) Ended
2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/sip:1...@10.12.13.166:5062 [CS_DESTROY]
2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1086 Session 15 
(sofia/internal/1...@10.12.13.45) Ended
2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1...@10.12.13.45 [CS_DESTROY]

Of note, with the endpoints registered through the Ekiga sip server, the sas 
DOES match on both ends.

With global_setvar zrtp_secure_media=false, the endpoints can't detect a zrtp 
peer.  

Reading the list archives hasn't enlightened me.

I see this comment from 2008

http://www.nabble.com/Freeswitch-and-Twinkle-and-ZRTP-
td18518140.html#a18518343

On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote:

 it should in bypass_media or proxy_media modes.  in the other modes we
 are in the media path and would not know how to handle the encrypted
 packets.

 Mike

Is this still relevant? Or is there some other setting not covered here

http://wiki.freeswitch.org/wiki/ZRTP

to make this work properly? I ask firstly about  this in the context of a 
peer 2 peer zrtp communication between the endpoints, then secondly in the 
case of FS acting as a trusted middleman as in section 2 here

http://www.zfoneproject.com/docs/asterisk/man/html/u_guide.html#passthrough

Lastly how does one implement the security enrollment as noted above with FS
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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Brian West 
This is because you didn't install the zrtpagent.lua script and dial  
zrtp on your keypad to enroll the FS box as a trusted man in the  
middle... which btw will only work with the unreleased zfone3.

/b

On Aug 23, 2009, at 4:37 PM, Harondel J. Sibble wrote:

 I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp  
 capable
 endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone  
 client,
 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i  
 (symbian
 s60) and an O2 Xda Flame (windows mobile 5).

 All 3 endpoints are registered with FS using the default extensions  
 of 1000-
 1003

 With global_setvar zrtp_secure_media=true the zrtp negotiation  
 between end
 points happens but the SAS never matches,below is console output for  
 a call
 between 2 of the endpoints


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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 
Brian, okay, that answers the case with FS acting as a trusted man in the 
middle, but what about in the peer to peer case? Shouldn't FS just be passing 
the ztrp traffic through to the endpoints? Or am I misunderstanding how it's 
supposed to work?

Secondly where would I find info about zrtpagent.lua? Doing a search for that 
term on the wiki returns no results, ditto for a search of the nabble list 
archives (other than your response today of course). Also ditto for a search 
of the box running FS.

On 23 Aug 2009 at 17:09, Brian West wrote:

 This is because you didn't install the zrtpagent.lua script and dial  
 zrtp on your keypad to enroll the FS box as a trusted man in the  
 middle... which btw will only work with the unreleased zfone3.
 
 /b

-- 
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Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
h...@pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Brian West 

On Aug 23, 2009, at 5:39 PM, Harondel J. Sibble wrote:

 Brian, okay, that answers the case with FS acting as a trusted man  
 in the
 middle, but what about in the peer to peer case? Shouldn't FS just  
 be passing
 the ztrp traffic through to the endpoints? Or am I misunderstanding  
 how it's
 supposed to work?

Nope.  FreeSWITCH is a b2bua so the traffic is decrypted... and  
relayed and encrypted again to the far side.  Which is why you have to  
use the trusted man in the middle stuff but you can't use it fully  
unless you have zfone3 beta or release.

 Secondly where would I find info about zrtpagent.lua? Doing a search  
 for that
 term on the wiki returns no results, ditto for a search of the  
 nabble list
 archives (other than your response today of course). Also ditto for  
 a search
 of the box running FS.

scripts/lua/ (in src tree)

/b


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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 


On 23 Aug 2009 at 17:48, Brian West wrote:

 Nope.  FreeSWITCH is a b2bua so the traffic is decrypted... and  

Hadn't heard that term before

http://en.wikipedia.org/wiki/Back-to-back_user_agent

that clears it up.  Any plans to offer straight proxy/passthru?

 relayed and encrypted again to the far side.  Which is why you have to  
 use the trusted man in the middle stuff but you can't use it fully  
 unless you have zfone3 beta or release.

Which explains the mismatched sas on each end. Okay got it.
 
 scripts/lua/ (in src tree)

ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add that into 
the mix and cross my fingers that zfone3 gets released soon along with it's 
inclusion into the softphones I have on my smartphone devices.
-- 
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Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
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Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

2009-08-23 Thread Raymond Chandler 
Could you load freeswitch with a couple hundred calls then run the  
test again.. and do the same to asterisk and see how the numbers stack  
up then? I'm just curious to see what happens at that point.


-Ray

On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:


Hi Everyone,

I'm working on a PBX project for the Sheevaplug ARM based computer,  
with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
So far I've found a big difference between Freeswitch and Asterisk  
performance times.
This is a comparison of the time it takes them to perform different  
actions:


startup Freeswitch: 3 min.
startup Asterisk:   2 sec.

call extension Freeswitch:  6 sec.
call extension Asterisk:0 sec.

shutdown Freeswitch:6.5 sec
shutdown Asterisk:  0 sec.

reload config Freeswitch:   1 sec.
reload config Asterisk: 1 sec.

Both were built from sources natively (no cross-compiling), and they  
use the default startup configurations.
I have managed to lower the Freeswitch times by disabling most of  
the modules and recompiling, but it is still far away from Asterisk  
(i.e. FS startup time 2.5 min).


1. Is there any way to further improve Freeswitch performance for  
the ARM architecture?
2. Can this be related to the lack of a FPU (the Sheevalug emulates  
the floating point operations).
3. On the startup I see this error repeated many times: [ERR]  
switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
related?


Thanks,
Rogelio Perez

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[Freeswitch-users] Couple of questions

2009-08-23 Thread Matt Riddell 
Hi,

I don't see how I can read some responses to command using esl.

I.E. esl_send_recv(handle, api show calls count\n\n);

and

printf(Header Test %s\n, esl_event_get_header(event, API-Command));
printf(Body Test %s\n, esl_event_get_body(event));

the header details are returned.

The body is null.

Also, I can originate a call and set the account code for it, but how do 
I get a list of calls with their account codes?

Do I get a list of calls then go through them one by one and get the 
variables for those calls by uuid?

Does anyone have any documentation for the esl api?

Even if I could read some comments from a usage of it would be useful.

-- 
Cheers,

Matt Riddell
Director
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Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

2009-08-23 Thread SP 
Don't forget to press tab at the asterisk console!  :)

On Sun, Aug 23, 2009 at 18:23, Raymond Chandler
intralan...@freeswitch.orgwrote:

 Could you load freeswitch with a couple hundred calls then run the test
 again.. and do the same to asterisk and see how the numbers stack up then?
 I'm just curious to see what happens at that point.
 -Ray

 On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:

 Hi Everyone,

 I'm working on a PBX project for the 
 Sheevaplughttp://www.google.com/url?sa=tsource=webct=rescd=1url=http%253A%252F%252Fwww.marvell.com%252Fproducts%252Fembedded_processors%252Fdeveloper%252Fkirkwood%252Fsheevaplug.jspei=EOOOSo6ILcyOtgeNruTOBAusg=AFQjCNFREhfy_erj5irBWk8XFUjkQOP-awsig2=pAHYlI15IbZ5Kcw0n-nIvA
  ARM
 based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
 So far I've found a big difference between Freeswitch and Asterisk
 performance times.
 This is a comparison of the time it takes them to perform different
 actions:

 startup Freeswitch: 3 min.
 startup Asterisk: 2 sec.

 call extension Freeswitch: 6 sec.
 call extension Asterisk: 0 sec.

 shutdown Freeswitch: 6.5 sec

 shutdown Asterisk: 0 sec.


 reload config Freeswitch: 1 sec.
 reload config Asterisk: 1 sec.


 Both were built from sources natively (no cross-compiling), and they use
 the default startup configurations.
 I have managed to lower the Freeswitch times by disabling most of the
 modules and recompiling, but it is still far away from Asterisk (i.e. FS
 startup time 2.5 min).

 1. Is there any way to further improve Freeswitch performance for the ARM
 architecture?
 2. Can this be related to the lack of a FPU (the Sheevalug emulates the
 floating point operations).
 3. On the startup I see this error repeated many times: [ERR]
 switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be related?

 Thanks,
 Rogelio Perez

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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Brian West 
I can confirm that it works 100% correct passing the SAS across the  
bridge correctly once you trust the switch in the middle.

/b

On Aug 23, 2009, at 6:16 PM, Harondel J. Sibble wrote:


 ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add  
 that into
 the mix and cross my fingers that zfone3 gets released soon along  
 with it's
 inclusion into the softphones I have on my smartphone devices.


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Re: [Freeswitch-users] Couple of questions

2009-08-23 Thread Brian West 

On Aug 23, 2009, at 6:27 PM, Matt Riddell wrote:

 Hi,

 I don't see how I can read some responses to command using esl.

 I.E. esl_send_recv(handle, api show calls count\n\n);

 and

 printf(Header Test %s\n, esl_event_get_header(event, API- 
 Command));
 printf(Body Test %s\n, esl_event_get_body(event));

 the header details are returned.

 The body is null.

I'm not too sure about using ESL in C, I have used it pretty much  
exclusively in perl.

 Also, I can originate a call and set the account code for it, but  
 how do
 I get a list of calls with their account codes?

originate {account_code=1234}sofia/profile/tar...@ip 

You can get the list of the channels via show channels or bridged  
calls with show calls

 From there you have the UUID's you can call uuid_dump on them to get  
all the variables.

 Do I get a list of calls then go through them one by one and get the
 variables for those calls by uuid?

You could do this or setup a listener to get the events as they happen  
and keep the info you need.

 Does anyone have any documentation for the esl api?

http://docs.freeswitch.org/ (this should help, its under files list  
see esl.h)

 Even if I could read some comments from a usage of it would be useful.

I just find it interesting you're doing this with C.

 -- 
 Cheers,

 Matt Riddell
 Director
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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 


On 23 Aug 2009 at 16:16, Harondel J. Sibble wrote:

 ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add that into
 the mix and cross my fingers that zfone3 gets released soon along with it's
 inclusion into the softphones I have on my smartphone devices.

Well good news for the Tiviphone client

Dear Harondel J. Sibble,

We support zfone3 starting from Tiviphone release 2.0.6 it is available for
Symbian try upgrading.

-- 
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Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Brian West 
Wish they would send me one for my E63 for testing... only been  
working with zfone 3 so far.

/b

On Aug 23, 2009, at 6:50 PM, Harondel J. Sibble wrote:

 Well good news for the Tiviphone client


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Re: [Freeswitch-users] Couple of questions

2009-08-23 Thread Matt Riddell 
On 24/08/09 11:50 AM, Brian West wrote:

 Even if I could read some comments from a usage of it would be useful.

 I just find it interesting you're doing this with C.

:)  The rest of the application is in C, so it makes sense to use 
FreeSwitch's esl in C.

Thanks for your help man will let you know if I have any other questions :)

-- 
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Director
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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 
They have a trial version of the full client available from their website, I 
think it's only for 10 days though. I suspect if you approached them, they'd 
probably give you a full client for permanent use for interoperability 
testing.

http://www.tivi.com/en/download/credit_paypal.php

Opps, my bad, it's only for 3 days :-(

You can bid on the price you want to pay if you are into buying a copy.

Just got the word back from them, the 2.0.6 client is available for Windows 
Mobile but not officially released since it has some issues with video 
encryption. Since I'm not using video yet, I am going to upgrade the windows 
mobile phones with this version.

So I popped the zrtp_agent.lua script into

/usr/local/freeswitch/scripts/

added following line to 

/usr/local/freeswitch/conf/autoload_configs/lua.conf.xml

   !--param name=startup-script value=zrtp_agent.lua/--

under this section

!--
The following options identifies a lua script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.

And restarted FS, when I try to connect and type zrtp on the keypad it 
doesn't work and I see

2009-08-23 17:06:46.272135 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1...@10.12.13.45 [03265312-9042-11de-8c5d-d333d780ffc7]
2009-08-23 17:06:46.288730 [INFO] mod_dialplan_xml.c:315 Processing 1001-
zrtp in context default
2009-08-23 17:06:46.301899 [NOTICE] switch_ivr.c:1349 Transfer 
sofia/internal/1...@10.12.13.45 to enum[z...@default]
2009-08-23 17:06:46.495944 [INFO] switch_core_state_machine.c:136 No Route, 
Aborting
2009-08-23 17:06:46.495944 [NOTICE] switch_core_state_machine.c:137 Hangup 
sofia/internal/1...@10.12.13.45 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2009-08-23 17:06:46.503261 [NOTICE] switch_core_session.c:1086 Session 3 
(sofia/internal/1...@10.12.13.45) Ended
2009-08-23 17:06:46.503261 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1...@10.12.13.45 [CS_DESTROY]

Were you being literal when you said to type in zrtp on the phones keypad? I 
assume you meant dial the phone number equivalent to that.

I also assume this is not working because 
1) I should put the zrtp_agent.lua script in one of the designated script 
dirs like 

!-- param name=module-directory value=/usr/lib/lua/5.1/?.so/ --
!-- param name=module-directory value=/usr/local/lib/lua/5.1/?.so/ 
--

or 

2) I need to wait until zfone3 is released and recompile FS with the new tar 
file from the zfoneproject site

or

3) both of the above apply?


On 23 Aug 2009 at 18:53, Brian West wrote:

 Wish they would send me one for my E63 for testing... only been  
 working with zfone 3 so far.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
h...@pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)


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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Brian West 


On Aug 23, 2009, at 7:20 PM, Harondel J. Sibble wrote:


added following line to

/usr/local/freeswitch/conf/autoload_configs/lua.conf.xml


WRONG.




  !--param name=startup-script value=zrtp_agent.lua/--



Don't touch this.


under this section

   !--
   The following options identifies a lua script that is launched
   at startup and may live forever in the background.
   You can define multiple lines, one for each script you
   need to run.

And restarted FS, when I try to connect and type zrtp on the keypad it
doesn't work and I see


Just put it into the scripts folder and run it from the dialplan see  
default configs.


!-- install zrtp_agent.lua into scripts (ZRTP == 9787) --
extension name=zrtp_enrollement
  condition field=destination_number expression=^9787$
action application=answer/
action application=lua data=zrtp_agent.lua/
  /condition
/extension


/b

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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 


On 23 Aug 2009 at 19:53, Brian West wrote:

 Just put it into the scripts folder and run it from the dialplan see  
 default configs.
 
  !-- install zrtp_agent.lua into scripts (ZRTP == 9787) --
  extension name=zrtp_enrollement
condition field=destination_number expression=^9787$
   action application=answer/
   action application=lua data=zrtp_agent.lua/
/condition
  /extension

Okay, where do I get the audio files?

2009-08-23 18:54:08.991629 [INFO] mod_dialplan_xml.c:315 Processing 1001-
9787 in context default
2009-08-23 18:54:09.10863 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1...@10.12.13.45] has been answered
2009-08-23 18:54:09.215288 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr  tp/zrtp-
status_securing.wav] [System error : No such file or directory.]
2009-08-23 18:54:12.233267 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr  tp/zrtp-
status_secure.wav] [System error : No such file or directory.]
2009-08-23 18:54:12.234565 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr  tp/zrtp-
enroll_welcome.wav] [System error : No such file or directory.]
2009-08-23 18:54:13.252933 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr  tp/zrtp-
enroll_confirmed.wav] [System error : No such file or directory.]
2009-08-23 18:54:15.293476 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
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..
2009-08-23 18:54:17.333543 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr  tp/zrtp-
thankyou_goodbye.wav] [System error : No such file or directory.]


my /usr/local/freeswitch/sounds is bereft of any files whatsoever, searching 
the full filesystem for callie only gets me

/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
16000.install
/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
32000.install
/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
8000.install





-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
h...@pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)


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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 
Ahh, I didn't quite clue-in that I had to run the additional installers as 
below when the main compile finished, I thought it was saying it had already 
done that.

Makes perfect sense in hindsite ;-)

+ FreeSWITCH install Complete --+
 + FreeSWITCH has been successfully installed.   +
 +   +
 +   Install sounds: +
 +   (uhd-sounds includes hd-sounds, sounds) +
 +   (hd-sounds includes sounds) +
 +   +
 +   make cd-sounds-install  +
 +   make cd-moh-install +
 +   +
 +   make uhd-sounds-install +
 +   make uhd-moh-install+
 +   +
 +   make hd-sounds-install  +
 +   make hd-moh-install +
 +   +
 +   make sounds-install +
 +   make moh-install+
 +   +
 +   Install non english sounds: +
 +   replace XX with language+
 +   (ru : Russian)  +
 +   +
 +   make cd-sounds-XX-install   +
 +   make uhd-sounds-XX-install  +
 +   make hd-sounds-XX-install   +
 +   make sounds-XX-install  +


On 23 Aug 2009 at 21:30, Brian West wrote:

 If you reinstall the latest sound files you'll have them.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
h...@pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)


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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 
Whoah.

2009-08-23 20:07:52.583524 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1...@192.168.73.45 [4ff9d452-905b-11de-8c5d-d333d780ffc7]
2009-08-23 20:07:52.740094 [INFO] mod_dialplan_xml.c:315 Processing 1001-
9787 in context default
2009-08-23 20:07:52.980164 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1...@192.168.73.45] has been answered
2009-08-23 20:07:54.62 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
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..
2009-08-23 20:08:16.912027 [NOTICE] sofia.c:322 Hangup 
sofia/internal/1...@192.168.73.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 20:08:22.54838 [NOTICE] switch_core_session.c:1086 Session 12 
(sofia/internal/1...@192.168.73.45) Ended
2009-08-23 20:08:22.54838 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1...@192.168.73.45 [CS_DESTROY]

I get audio now, but it's running really slooowly. I'd say about 
1/4 to 1/8 normal speech speed

I did the following

 make cd-sounds-install;make uhd-sounds-install;make uhd-moh-install

Probably relevant, box this is running on is a Celeron 1ghz with 512mb ram.

Weird, it starts off okay, but when this pops up

2009-08-23 20:21:10.93004 [WARNING] mod_sofia.c:810 We were told to use ptime 
20 but what they meant to say was 80
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..

the audio starts to crawl...
On 23 Aug 2009 at 21:30, Brian West wrote:

 If you reinstall the latest sound files you'll have them.
 
 /b
 
 On Aug 23, 2009, at 9:05 PM, Harondel J. Sibble wrote:
 
  /extension
 
  Okay, where do I get the audio files?
 
 
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Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
h...@pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)


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[Freeswitch-users] Screaming monkeys on ext 5000

2009-08-23 Thread Scott Torr 
Just a quick note, and I'm sure why, but screaming monkeys does not play
on the the default installation.

I have not looked into why, but thought I would just quickly let you
know.

Perhaps I have not done something?

regards,
sbt 

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Re: [Freeswitch-users] Screaming monkeys on ext 5000

2009-08-23 Thread Brian West 

On Aug 23, 2009, at 10:26 PM, Scott Torr wrote:

 Just a quick note, and I'm sure why, but screaming monkeys does not  
 play
 on the the default installation.

It requires internet connectivity.  It calls a remote system to play  
which is out of our control.

 I have not looked into why, but thought I would just quickly let you
 know.

Thanks,

 Perhaps I have not done something?

 regards,
 sbt

/b


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Re: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with same Gateway

2009-08-23 Thread Jim Burke 
If I understand your issue correctly, it sounds to me like FS is not
set to anchor the RTP media stream.  Experience suggests that most
SBC's do not like trying to loopback RTP traffic to themselves.

Check to see what IP address's are getting used in the
c=xxx.xxx.xxx.xxx for the INVITE and 200OK messages.  If you see the
SBC IP address in messages on both sides of the FS box this will be
your issue.

You could try setting bypass_media=false in your dialplan.



On Sat, Aug 22, 2009 at 4:41 PM, Matthew Fongmattdf...@gmail.com wrote:
 So there seems to be some sort of error when bridging directly like
 originate
 {ignore_early_media=true}sofia/gateway/.com/91415992 bridge(sofia/gateway/.com/91415465)
 BUT
 if I get FS to send media to leg A, and then bridge to leg B by using a lua
 script like
 session:streamFile(/usr/local/freeswitch/sounds/en/us/callie/hh/hh-welcome.wav);
 session:execute(bridge, sofia/gateway/epik.com/91415XXX);
 then the legs bridge together OK. This happens when trying to bridge two
 channels via the same Broadsoft SBC. Does this sound like a bug that should
 be submitted to JIRA?
 --matt
 http://www.hellohunter.com

 On Thu, Aug 20, 2009 at 10:21 AM, Matthew Fong mattdf...@gmail.com wrote:

 originate
 {ignore_early_media=true}sofia/gateway/epik.com/914159927717 bridge(sofia/gateway/epik.com/914154650027)

 is the string I was using from the console.

 On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene mrene_li...@avgs.ca wrote:

 Hi
 How are you bridging the calls in FS (which api call or C function are
 you using)?
 Mathieu Rene
 Avant-Garde Solutions Inc
 Office: + 1 (514) 664-1044 x100
 Cell: +1 (514) 664-1044 x200
 mr...@avgs.ca



 On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:

 I'm trying to get FreeSWITCH to bridge two channels together through the
 same external gateway, but I'm having issues hearing audio. Both legs if
 setup independently and forwarded to 5000 (test ivr) work fine for both
 inbound and outbound media, but when I try to bridge them together,
 everything seems fine in FreeSWITCH, but neither party can hear the other
 speak. I'm thinking the external gateway might be having some issues because
 I've been able to bridge 2 channels together through the same gateway on
 different providers, but thought I'd also try to seek some help here.
 FreeSWITCH should be handling the media for both channels, so I can't figure
 out why if Leg A and Leg B work independently, but not if they are bridged
 together. Is there a setting somewhere in FS that I'm missing?
 Below is a ngrep of the SIP interactions if it's useful. Thanks for the
 help.
 --matt

 interface: eth0 (172.24.200.0/255.255.255.0)
 filter: (ip or ip6) and ( port 5060 )
 U 2009/08/20 07:11:34.038686 216.81.56.198:5080 - 38.98.58.148:5060
 INVITE sip:914159927...@38.98.58.148 SIP/2.0.
 Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
 Max-Forwards: 70.
 From: FreeSWITCH sip:000...@216.81.56.198;tag=ZtFvjeFQmXvpp.
 To: sip:914159927...@38.98.58.148.
 Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
 CSeq: 119257811 INVITE.
 Contact: sip:gw+epik@216.81.56.198:5080;transport=udp.
 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
 NOTIFY, REFER, UPDATE, REGISTER, INFO.
 Supported: timer, precondition, path, replaces.
 Allow-Events: talk, refer.
 Content-Type: application/sdp.
 Content-Disposition: session.
 Content-Length: 293.
 Remote-Party-ID: FreeSWITCH
 sip:000...@216.81.56.198;party=calling;screen=yes;privacy=off.
 .
 v=0.
 o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
 s=FreeSWITCH.
 c=IN IP4 216.81.56.198.
 t=0 0.
 m=audio 24700 RTP/AVP 0 8 3 101 13.
 a=rtpmap:0 PCMU/8000.
 a=rtpmap:8 PCMA/8000.
 a=rtpmap:3 GSM/8000.
 a=rtpmap:101 telephone-event/8000.
 a=fmtp:101 0-16.
 a=rtpmap:13 CN/8000.
 a=ptime:20.

 U 2009/08/20 07:11:34.128331 38.98.58.148:5060 - 216.81.56.198:5080
 SIP/2.0 100 Trying.
 From: FreeSWITCH sip:000...@216.81.56.198;tag=ZtFvjeFQmXvpp.
 To: sip:914159927...@38.98.58.148;tag=F725.2C49.
 Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
 CSeq: 119257811 INVITE.
 Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
 Contact: sip:914159927...@38.98.58.148:5060.
 Content-Length: 0.
 .

 U 2009/08/20 07:11:34.338105 38.98.58.148:5060 - 216.81.56.198:5080
 SIP/2.0 183 Session Progress.
 From: FreeSWITCH sip:000...@216.81.56.198;tag=ZtFvjeFQmXvpp.
 To: sip:914159927...@38.98.58.148;tag=F725.2C49.
 Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
 CSeq: 119257811 INVITE.
 Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
 Contact: sip:914159927...@38.98.58.148:5060.
 Allow:
 INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
 Content-Type: application/sdp.
 Content-Length: 227.
 .
 v=0.
 o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
 s=M6 Call.
 c=IN IP4 38.98.58.148.
 t=0 0.
 m=audio 42554 RTP/AVP 0

Re: [Freeswitch-users] Call exits after 120 seconds with hangup cause

2009-08-23 Thread Jim Burke 
In your SIP profiles this could be set.  I beleive 120 is the default setting.

param name=session-timeout value=120

On Fri, Aug 21, 2009 at 11:35 PM, bakkoasannu...@gmail.com wrote:
 Do you have those lines in switch.conf file?

    !--RTP port range --
    param name=rtp-start-port value=1/
    param name=rtp-end-port value=32767/


 BR

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-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net

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Re: [Freeswitch-users] Couple of questions

2009-08-23 Thread João Mesquita 
Hey there, FsGui uses ESL a lot and I had to go through the code to document
it so here is a few hints inline ...

Don't hesitate to keep the questions coming. I will fill in whenever I can.

jmesquita

On Sun, Aug 23, 2009 at 8:50 PM, Brian West br...@freeswitch.org wrote:


 On Aug 23, 2009, at 6:27 PM, Matt Riddell wrote:

  Hi,
 
  I don't see how I can read some responses to command using esl.
 
  I.E. esl_send_recv(handle, api show calls count\n\n);
 
  and
 
  printf(Header Test %s\n, esl_event_get_header(event, API-
  Command));
  printf(Body Test %s\n, esl_event_get_body(event));
 
  the header details are returned.
 
  The body is null.


Body is null on every event that does not use headers to output information.
A good example would be console logs. I haven't seen too many default events
besides log that have body besides a application custom events.




 I'm not too sure about using ESL in C, I have used it pretty much
 exclusively in perl.

  Also, I can originate a call and set the account code for it, but
  how do
  I get a list of calls with their account codes?

 originate {account_code=1234}sofia/profile/tar...@ip 

 You can get the list of the channels via show channels or bridged
 calls with show calls

  From there you have the UUID's you can call uuid_dump on them to get
 all the variables.



  Do I get a list of calls then go through them one by one and get the
  variables for those calls by uuid?


All ESL does is output events to socket and expose the API commands. It does
not maintain any kind of list of calls or anything like that so it is up to
you to maintain that yourself if you don't want to parse API output every
time.




 You could do this or setup a listener to get the events as they happen
 and keep the info you need.

  Does anyone have any documentation for the esl api?

 http://docs.freeswitch.org/ (this should help, its under files list
 see esl.h)


I need to work a little bit more on that documentation as well.. I saw a few
conflicts with the core documentation too. Will get there once I have some
more time left.



  Even if I could read some comments from a usage of it would be useful.

 I just find it interesting you're doing this with C.

  --
  Cheers,
 
  Matt Riddell
  Director
  ___
 
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Re: [Freeswitch-users] Transporting SIP over TCP

2009-08-23 Thread Jim Burke 
X-Lite does support TCP, however you need to have NAPTR and SRV DNS
entries.  It used to support TLS, but this seems to have been removed
:(

sip.mydomain.net. IN NAPTR 0 0 s SIPS+D2T   _sips._tcp.sip.mydomain.net.
sip.mydomain.net. IN NAPTR 1 0 s SIP+D2T   _sip._tcp.sip.mydomain.net.
sip.mydomain.net. IN NAPTR 2 0 s SIP+D2U   _sip._udp.sip.mydomain.net.


_sip._udp.sip.mydomain.net.   43200 IN SRV 1 10 5060  sip.mydomain.net.
_sip._tcp.sip.mydomain.net.   43200 IN SRV 1 10 5060  sip.mydomain.net.
_sips._tcp.sip.mydomain.net.   43200 IN SRV 1 10 5061  sip.mydomain.net.


On Mon, Aug 17, 2009 at 10:18 PM, bakkoasannu...@gmail.com wrote:
 Y make my tests with eyebeam.

 I thing X-lite dont't support TCP transport.

 BR

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Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

2009-08-23 Thread Rogelio Perez 

Thanks Andrew and Anthony,

I created a ramdisk for the db and log directories using tmpfs and now  
I see better performance times:


startup:15.6 sec.
call extension: 0 sec.
shutdown:   7.5 sec
reload config:  0 sec.

I have noticed that during the startup there is a 12 sec. pause while  
checking for UPnP:


2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for  
PMP [general error]

2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP NAT  
detected!


Is there any way to lower this time? Maybe disabling the check?

Thanks,
Rogelio

On Aug 21, 2009, at 5:20 PM, Anthony Minessale wrote:


probably disk i/o.

Is it some kind of flash drive?

make a ramdisk and simlink in /usr/local/freeswitch/db and /usr/ 
local/freeswitch/log to it
the default configuration uses a lot of high level features that use  
the sqlite db on the disk.


We also offer commercial support where we could dig deeper into the  
problem if you can't figure it out

consult...@freeswitch.org



On Fri, Aug 21, 2009 at 2:15 PM, Rogelio Perez rogelio.pe...@gmail.com 
 wrote:

Hi Everyone,

I'm working on a PBX project for the Sheevaplug ARM based computer,  
with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
So far I've found a big difference between Freeswitch and Asterisk  
performance times.
This is a comparison of the time it takes them to perform different  
actions:


startup Freeswitch: 3 min.
startup Asterisk:   2 sec.

call extension Freeswitch:  6 sec.
call extension Asterisk:0 sec.

shutdown Freeswitch:6.5 sec
shutdown Asterisk:  0 sec.

reload config Freeswitch:   1 sec.
reload config Asterisk: 1 sec.

Both were built from sources natively (no cross-compiling), and they  
use the default startup configurations.
I have managed to lower the Freeswitch times by disabling most of  
the modules and recompiling, but it is still far away from Asterisk  
(i.e. FS startup time 2.5 min).


1. Is there any way to further improve Freeswitch performance for  
the ARM architecture?
2. Can this be related to the lack of a FPU (the Sheevalug emulates  
the floating point operations).
3. On the startup I see this error repeated many times: [ERR]  
switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
related?


Thanks,
Rogelio Perez


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Re: [Freeswitch-users] Transporting SIP over TCP

2009-08-23 Thread Jim Burke 
Well if you append ;transport=tcp on the bridge lines it will use TCP

IMHO this statement needs some clarification based on the context of
this thread.

If the destination is another PBX or Freeswitch box then this is ok as
FS will be initiating the TCP connection.

For terminating calls to registered User Agents (UA) the decision to
use TCP or UDP should be made using information collected when the UA
registered.  i.e if the UA registers using TCP, FS should use TCP to
send and receive messages, if the UA registers with TLS, then FS
should use TLS, same story for UDP.

On Tue, Aug 18, 2009 at 12:10 AM, Brian Westbr...@freeswitch.org wrote:
 Well if you append ;transport=tcp on the bridge lines it will use TCP .

 /b

 On Aug 17, 2009, at 9:06 AM, Tzury Bar Yochay wrote:

 FreeSWITCH works very well as a client :P
 I am currently porting it into iPhone and Symbian, I am almost
 done ;-)

 anyway, seriously now, can one point to a wiki page about this?
 How do I do that?
 I would need 3 server instances to place a call, right?


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Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

2009-08-23 Thread Brian West 
freeswitch -nonat

/b

On Aug 23, 2009, at 11:45 PM, Rogelio Perez wrote:

 2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for  
 PMP [general error]
 2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
 2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP  
 NAT detected!


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Re: [Freeswitch-users] Transporting SIP over TCP

2009-08-23 Thread Brian West 
It already does exactly this.

/b

On Aug 23, 2009, at 11:49 PM, Jim Burke wrote:

 For terminating calls to registered User Agents (UA) the decision to
 use TCP or UDP should be made using information collected when the UA
 registered.  i.e if the UA registers using TCP, FS should use TCP to
 send and receive messages, if the UA registers with TLS, then FS
 should use TLS, same story for UDP.


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Re: [Freeswitch-users] Transporting SIP over TCP

2009-08-23 Thread Jim Burke 
I always expected it did :)

My point was that you cannot put transport=TCP on a bridge statement
line to an internal registered client and expect it to use a protocol
that was not used at registration.

Hence the clarification based on the context of the thread :)

On Mon, Aug 24, 2009 at 2:53 PM, Brian Westbr...@freeswitch.org wrote:
 It already does exactly this.

 /b

 On Aug 23, 2009, at 11:49 PM, Jim Burke wrote:

 For terminating calls to registered User Agents (UA) the decision to
 use TCP or UDP should be made using information collected when the UA
 registered.  i.e if the UA registers using TCP, FS should use TCP to
 send and receive messages, if the UA registers with TLS, then FS
 should use TLS, same story for UDP.


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Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

2009-08-23 Thread Rogelio Perez 
Thanks Brian, now the startup time is 3 sec.

On Aug 24, 2009, at 1:52 AM, Brian West wrote:

 freeswitch -nonat

 /b

 On Aug 23, 2009, at 11:45 PM, Rogelio Perez wrote:

 2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for
 PMP [general error]
 2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
 2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP
 NAT detected!


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Re: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4

2009-08-23 Thread Harondel J. Sibble 


On 23 Aug 2009 at 20:22, Harondel J. Sibble wrote:

 Whoah.
 I get audio now, but it's running really slooowly. I'd say about
 1/4 to 1/8 normal speech speed

Hmmm, using one of my hardphones, specifically the Integrated Networks IN-
1002

2009-08-23 20:46:07.334596 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1...@10.12.13.45 [a7c10746-9060-11de-8c5d-d333d780ffc7]
2009-08-23 20:46:07.348834 [INFO] mod_dialplan_xml.c:315 Processing 1004-
5000 in context default
2009-08-23 20:46:07.364484 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1...@10.12.13.45] has been answered
2009-08-23 20:46:08.33173 [WARNING] mod_sofia.c:810 We were told to use ptime 
20 but what they meant to say was 10
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..
2009-08-23 20:47:30.512970 [NOTICE] switch_core_state_machine.c:179 Hangup 
sofia/internal/1...@10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 20:47:30.541157 [NOTICE] switch_core_session.c:1086 Session 20 
(sofia/internal/1...@10.12.13.45) Ended
2009-08-23 20:47:30.541157 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1...@10.12.13.45 [CS_DESTROY]

gets me much better results and no slowdown of audio, although there are some 
drop outs.  Running Ekiga on xp machine with zfone client works okay, but 
being it's an older verison of zfone, I'm guessing it doesn't support the 
enrollment packet from FS as the voice says I have to choose to enroll and 
then says goodbye ;-)

From looking through the list archives, I am guessing I need to talk to the 
Tivi developers, question is what should I say to them, there's nothing 
really exposed in the client, other than adjusting the codecs, 3 choices
GSM, uLaw and aLaw can be enabled or disabled.

Both phones are running over a wifi link and there is (randomly) a fair bit 
of interference affecting speeds.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
h...@pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)


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