[Freeswitch-users] Load Balacing

[Juan Manuel Vicente]

Hello:

I am using two boxes, with same domain, to register about 1000 users working
together a DNS box doing round robin to resolve two differentes Ips one to
each box. This two boxes are using the same database (through unixodbc).

When I use the funcion sofia_contact or sofia profile status both boxes show
the status of the user. Does not matter where the resgistration was did.

But when I try to call the user this call fails, the call only works if the
call is sent through the same box where the user is registered.

Any ideas where i need to looking for ?

Thx in Advance.
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[Freeswitch-users] SDP Passthrough, INVITE messages.

[Juan Manuel Vicente]

Hi,

I'm trying to use the Freeswitch as a proxy (I know that is not designed for
that, but I really need to do it in this way), here is my config:

Endpoint 1- FS A--FS B-FS A-Endpoint 2

 * Both Endpoints are registered in FS A how is acting as a proxy and
registrar.
 * FS B  only sends back the Invite to FS A in order to reach Endpoint 2.
 * Both FS have a public IP
 * FS A Only handles SIP messages
 * FS B Handles RTP (Also SIP)

My objetive is to keep the signaling in FS A and the RTP in FS B so
basically FS A will work as a registrar.

So far I've been able to succesfully do it if both endpoint are not nated,
how ever I do need to do it in a Natted sceneario too, for what I have been
sniffing the problem is that in the INVITE, the SDP is sending the internal
IP instead of the external.

I've tried to change the switch_r_sdp and switch_l_sdp but I'm not quite
sure if I'm doing the correct config of the switch (late_codec_negotiation)

If anyone could give a tip or a sample of how can I change the INVITE
messages I will appreciate.

Thanks in advance
Regards
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Re: [Freeswitch-users] xml_curl dynamically register users problem

[Juan Manuel Vicente]

Yes, I saw that after send that email.
This is the corrected one

?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=directory
domain name=200.49.25.11
params
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain},transfer
_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@
${dialed_domain})}/
/params
groups
 group name=default
users
user id=5000
params
param name=password value=5000/
param name=vm-password value=5000/
/params
variables
   variable name=sip-force-contact
value=NDLB-connectile-dysfunction/
 variable name=user_context value=default/
 /variables
/user
/users
/group
/groups
/domain
/section
/document

And the dial plan:

document type=freeswitch/xml
section name=dialplan description=default
context name=default
extension name=llamando
condition field=destination_number expresion=^5000$
action application=bridge data=user/5000/
/condition
/extension
/context
/section
/document

I tried with:

action application=bridge data=user/5000/
action application=bridge data=user/5...@200.49.25.11/
action application=bridge data=user/5000%200.49.25.11/

The console is the same:

2009-04-22 18:00:28 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
1000-5000 in context default
2009-04-22 18:00:28 [ERR] switch_xml.c:1532 switch_xml_locate() Error[[error
near line 1]: root tag missing]
2009-04-22 18:00:28 [WARNING] mod_dptools.c:2171 user_outgoing_channel()
Can't find user [5...@200.49.25.11]
2009-04-22 18:00:28 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate()
Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-04-22 18:00:28 [INFO] mod_dptools.c:1909 audio_bridge_function()
Originate Failed.  Cause: SUBSCRIBER_ABSENT
2009-04-22 18:00:28 [NOTICE] mod_dptools.c:1936 audio_bridge_function()
Hangup sofia/internal/1...@200.49.25.11 [CS_EXECUTE] [SUBSCRIBER_ABSENT]
2009-04-22 18:00:28 [NOTICE] switch_core_session.c:957
switch_core_session_thread() Session 17 (sofia/internal/1...@200.49.25.11)
Ended
2009-04-22 18:00:28 [NOTICE] switch_core_session.c:959
switch_core_session_thread() Close Channel
sofia/internal/1...@200.49.25.11[cs_hangup]
2009-04-22 18:00:52 [NOTICE] switch_channel.c:565 switch_channel_set_name()
New Channel 
sofia/internal/1...@200.49.25.11[a9e8b0d2-2f80-11de-90e9-8b6680def56e]

also i was trying with the dialplan hardcode in
/usr/local/freeswitch/conf/dialplan/default.xml.

Thanks in advance.

On Wed, Apr 22, 2009 at 3:03 PM, Michael Jerris m...@jerris.com wrote:

 $${domain} (or any other preprocessor vars) will not be expanded on an
 xml_curl return.

 Mike

 On Apr 22, 2009, at 12:31 PM, JuanMa wrote:

  Peter Thanks for your reply:
 
  I did what you said but FS still with the same problem, cant found the
  user.
 
  This is my reply to registration request
 
  ?xml version=1.0 encoding=UTF-8 standalone=no?
  document type=freeswitch/xml
section name=directory
domain name=$${domain}
params
param name=dial-string
 value={presence_id=${dialed_us...@$
  {dialed_domain},transfer_fallback_extension=${dialed_user}}$
  {sofia_contact(${dialed_us...@${dialed_domain})}/
/params
groups
group name=default
users
user id=5000
params
param
 name=password value=5000/
param
 name=vm-password value=5000/
/params
variables
variable
 name=sip-force-contact value=NDLB-
  connectile-dysfunction/
variable
 name=user_context value=default/
/variables
/user
/users
/group
/groups
/domain
/section
  /document
 
  I have made some changes but the result is the same, the registration
  is ok.
 
  My dial plan reply, call from user 1000 to user 5000 registered
  through xml_curl
 
  document type=freeswitch/xml
section name=dialplan description=default
context name=default
extension name=llamando
condition field=destination_number
 expresion=^5000$
action application=bridge
 data=user/5000/
/condition
/extension
/context
/section
  /document
 
  console returns: