[Freeswitch-users] Flushing the Event buffer in Perl Event Socket

2009-10-30 Thread velusamy velu
Dear All,
  I receiving the events in while loop by using recvEventTimed method in
ESL.pm. I have to flush that Event buffer after some particular time. How
can I do it?

Thanks,
Velusamy
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Re: [Freeswitch-users] PBX vs. SIP proxy?

2009-10-30 Thread Fred-145

Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
the registration process, while a PBX offers services like ringing phones,
setting up calls, handling voice mail, etc.
-- 
View this message in context: 
http://old.nabble.com/PBX-vs.-SIP-proxy--tp25920788p26128938.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket

2009-10-30 Thread Anthony Minessale
read them in a timed loop of some small number of MS until you get a timeout
meaning you have flushed them all.


On Fri, Oct 30, 2009 at 1:57 AM, velusamy velu velu.techni...@gmail.comwrote:

 Dear All,
   I receiving the events in while loop by using recvEventTimed method
 in ESL.pm. I have to flush that Event buffer after some particular time. How
 can I do it?

 Thanks,
 Velusamy

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Re: [Freeswitch-users] Setting up Conference with Moderator

2009-10-30 Thread Ujjval Karihaloo
It's strange... a tcpdump tells me that there is no DTMF from my provider when 
using IVR, but when I call into a TN that goes directly into the Conference 
App, I see DTMF from the provider.



-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 7:23 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

I've never had any problem with that.  Is your logging at debug level  
so you can see the RECV DTFM in the log/fs_cli?  Are you calling from  
a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider  
isn't passing them through.

Make sure your logging is turned up then try something simpler, like  
calling the echo application, and see if DTFM comes through.

Rob

On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:

 Rob:

 For some reason, I don't see the DTMF appear on the fs_CLI when  
 using the below configurationso it basically timesout.

 UK



 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org 
 [mailto:freeswitch-users-boun...@lists.freeswitch.org 
 ] On Behalf Of Ujjval Karihaloo
 Sent: Monday, October 26, 2009 9:21 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

 Thx a lot Rob, reading the wiki your way or using IVR seems correct..
 ===
 The wiki also says that the wait-mod might be  used in conjunction
 with an IVR where the moderators are authenticated with an extra pass-
 code, which is what I did.  I guess that's why I didn't understand
 the point of the +pin.
 ==

 I will try it out.

 Again thx a lot for your help. Will keep everyone posted.

 
 From: freeswitch-users-boun...@lists.freeswitch.org 
 [freeswitch-users-boun...@lists.freeswitch.org 
 ] On Behalf Of Rob Forman [rob4manh...@gmail.com]
 Sent: Friday, October 23, 2009 12:22 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

 I just re-tested with the pin in my dial plan:

   action application=conference data=conference 123...@default
 +flags{}+1234 /

 And it doesn't challenge me for the pin.  I just drop right in.  I
 figured this is how it was intended, since the wiki says the pin is
 set initially and only challenged in later attempts [by future  
 callers]:

 The first time a conference name (confname) is used, it will be
 created on demand, and the pin will be set to what ever is specified
 at that time: the pin in the data string if specified, or if not, the
 pin setting in the conference profile, and if that is also
 unspecified, then there is no pin protection. Any later attempt to
 join the conference must specify the same pin number, if one existed
 when it was created. 


 The wiki also says that the wait-mod might be  used in conjunction
 with an IVR where the moderators are authenticated with an extra pass-
 code, which is what I did.  I guess that's why I didn't understand
 the point of the +pin.

 I'm sure there's a scenario where its used and useful, the wiki just
 doesn't explain it.

 Rob

 On Oct 23, 2009, at 12:43 PM, Brian West wrote:

 Well first off you're not defining a pine here...

 confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
 number]

 That might be why its not asking for a pin.

 /b

 On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:

 entry action=menu-exec-app digits=1 param=conference
 123...@default+flags{} /


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Re: [Freeswitch-users] Setting up Conference with Moderator

2009-10-30 Thread Rob Forman
Hm, strange.  I haven't seen that before.  Can you pastebin your logs  
at debug level?

On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:

 It's strange... a tcpdump tells me that there is no DTMF from my  
 provider when using IVR, but when I call into a TN that goes  
 directly into the Conference App, I see DTMF from the provider.



 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org 
 [mailto:freeswitch-users-boun...@lists.freeswitch.org 
 ] On Behalf Of Rob Forman
 Sent: Friday, October 30, 2009 7:23 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

 I've never had any problem with that.  Is your logging at debug level
 so you can see the RECV DTFM in the log/fs_cli?  Are you calling from
 a SIP phone on the pbx, or via a PSTN provider?  Maybe your provider
 isn't passing them through.

 Make sure your logging is turned up then try something simpler, like
 calling the echo application, and see if DTFM comes through.

 Rob

 On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:

 Rob:

 For some reason, I don't see the DTMF appear on the fs_CLI when
 using the below configurationso it basically timesout.

 UK



 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org 
 [mailto:freeswitch-users-boun...@lists.freeswitch.org
 ] On Behalf Of Ujjval Karihaloo
 Sent: Monday, October 26, 2009 9:21 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

 Thx a lot Rob, reading the wiki your way or using IVR seems correct..
 ===
 The wiki also says that the wait-mod might be  used in conjunction
 with an IVR where the moderators are authenticated with an extra  
 pass-
 code, which is what I did.  I guess that's why I didn't understand
 the point of the +pin.
 ==

 I will try it out.

 Again thx a lot for your help. Will keep everyone posted.

 
 From: freeswitch-users-boun...@lists.freeswitch.org 
 [freeswitch-users-boun...@lists.freeswitch.org
 ] On Behalf Of Rob Forman [rob4manh...@gmail.com]
 Sent: Friday, October 23, 2009 12:22 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

 I just re-tested with the pin in my dial plan:

  action application=conference data=conference 123...@default
 +flags{}+1234 /

 And it doesn't challenge me for the pin.  I just drop right in.  I
 figured this is how it was intended, since the wiki says the pin is
 set initially and only challenged in later attempts [by future
 callers]:

 The first time a conference name (confname) is used, it will be
 created on demand, and the pin will be set to what ever is specified
 at that time: the pin in the data string if specified, or if not, the
 pin setting in the conference profile, and if that is also
 unspecified, then there is no pin protection. Any later attempt to
 join the conference must specify the same pin number, if one existed
 when it was created. 


 The wiki also says that the wait-mod might be  used in conjunction
 with an IVR where the moderators are authenticated with an extra  
 pass-
 code, which is what I did.  I guess that's why I didn't understand
 the point of the +pin.

 I'm sure there's a scenario where its used and useful, the wiki just
 doesn't explain it.

 Rob

 On Oct 23, 2009, at 12:43 PM, Brian West wrote:

 Well first off you're not defining a pine here...

 confn...@profilename+flags{mute|deaf|waste|moderator}+[conference  
 pin
 number]

 That might be why its not asking for a pin.

 /b

 On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:

entry action=menu-exec-app digits=1 param=conference
 123...@default+flags{} /


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[Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call

2009-10-30 Thread Brian Stafford
Hi all

I did a 'make update' to 15289 and I found the auto feature added to 
mod_valet_parking so I figured I'd try it out.  Everything works fine 
except that the extension number for the parked call is reported on the 
wrong leg of the call.  It'd be good to sort this out as this is a 
useful feature for us.

Regards
Brian Stafford

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[Freeswitch-users] FreeSWITCH Weekly Conference Call Starting Shortly, Please Start Calling In

2009-10-30 Thread Michael Collins
Hey everyone! The weekly meeting will start in a few minutes. Here's the
agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30

Please feel free to add other subjects. If you have something to share with
the community that would be great. Also, if you have questions for the dev
team by all means bring them too.

-Michael
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Re: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call

2009-10-30 Thread Brian West
You have to be doing it wrong then.

Can you show us your dialplan you should have two extensions one for  
the lot range and one to attended transfer someone into the lot.

/b




On Oct 30, 2009, at 10:47 AM, Brian Stafford wrote:

 Hi all

 I did a 'make update' to 15289 and I found the auto feature added to
 mod_valet_parking so I figured I'd try it out.  Everything works fine
 except that the extension number for the parked call is reported on  
 the
 wrong leg of the call.  It'd be good to sort this out as this is a
 useful feature for us.

 Regards
 Brian Stafford


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[Freeswitch-users] Freeswitch in signaling path only

2009-10-30 Thread DJB
I am wondering why I cannot do as condition#2.

For Lua in dialplan, when I have the followings:


--WORKING--
(Condition#1)
.
.
session:execute(set,bypass_media=true)
session:execute(set,hangup_after_bridge=true)
session:execute(set,continue_on_fail=true)
.
.
 session:execute(bridge,sofia/external/ .. called_num .. 
@1.1.1.1|sofia/external/ .. called_num .. @1.1.1.2)
.
.

--NOT WORKING--
(Condition#2)
Note:  FS tries to be in media path and send re-invite.
.
.
session:execute(set,bypass_media=true)
session:execute(set,hangup_after_bridge=true)
session:execute(set,continue_on_fail=true)
.
.
 session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.1)
 session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.2)
.
.

Thank you,
Dorn B.


  

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Re: [Freeswitch-users] mod_valet_parking: auto reports on wrong leg of call

2009-10-30 Thread Brian Stafford
Brian West wrote:
 You have to be doing it wrong then.

 Can you show us your dialplan you should have two extensions one for  
 the lot range and one to attended transfer someone into the lot.

 /b
   
The relevant excerpt from the dialplan is

extension name=valet_unpark
condition field=destination_number expression=^(41[0-9])$
action application=answer/
action application=valet_park data=valet_lot $1/
/condition
/extension

extension name=valet_park
condition field=destination_number expression=^(420)$
action application=answer/
action application=valet_park data=valet_lot auto in 410 419/
/condition
/extension

x410-419 are the slots and 420 parks a call. Parking by picking one of 
410-419 works fine and subsequently dialling them from another works 
fine, I added x420 for the auto feature.

Regards
Brian

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

2009-10-30 Thread Ivan C Myrvold
I have already set debug to 9, on both profiles.

Ivan


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

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


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


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

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

 Ivan

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

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 -- 
 Eliot Gable

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

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

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

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Re: [Freeswitch-users] Freeswitch in signaling path only

2009-10-30 Thread Chris Burns
Do you have a debug level message from switch_ivr_originate.c in your log?
Channel is already up, delaying proxy mode 'till both legs are answered.

Set bypass_media b4 each bridge. It is unsetting on you and setting 
bypass_media_after_bridge because you already answered the channel running 
the lua script.

On October 30, 2009 12:03:29 pm DJB wrote:
 I am wondering why I cannot do as condition#2.

 For Lua in dialplan, when I have the followings:


 --WORKING--
 (Condition#1)
 .
 .
 session:execute(set,bypass_media=true)
 session:execute(set,hangup_after_bridge=true)
 session:execute(set,continue_on_fail=true)
 .
 .
  session:execute(bridge,sofia/external/ .. called_num ..
 @1.1.1.1|sofia/external/ .. called_num .. @1.1.1.2) .
 .

 --NOT WORKING--
 (Condition#2)
 Note:  FS tries to be in media path and send re-invite.
 .
 .
 session:execute(set,bypass_media=true)
 session:execute(set,hangup_after_bridge=true)
 session:execute(set,continue_on_fail=true)
 .
 .
  session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.1)
  session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.2)
 .
 .

 Thank you,
 Dorn B.




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

2009-10-30 Thread Eliot Gable
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^

Then run your call, then do this:

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

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

 Ivan


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

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


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


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

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

 Ivan

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

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 --
 Eliot Gable

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

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

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

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I decided the words were too conservative for me. We're not borrowing
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Re: [Freeswitch-users] Gateway registration fails: IP used for realm in place of configured realm value

2009-10-30 Thread Robert Wilczyński
Hi Brian,

Thanks for the explanation. This seems to have solved the issue.

Robert.

 -- Forwarded message --
 From: Brian West br...@freeswitch.org
 To: freeswitch-us...@lists.freeswitch.org
 Date: Thu, 29 Oct 2009 13:31:26 -0500
 Subject: Re: [Freeswitch-users] Gateway registration fails: IP used for realm 
 in place of configured realm value
 It's because you need to set the from-user and from-domain because FreeSWITCH 
 don't care about the realm because WE provide you the realm in our 
 challenge... you only provide realm for broken providers such as BroadVoice.

 remove realm from your setting.  Add from-user and from-domain

 /b

 On Oct 29, 2009, at 4:46 AM, Robert Wilczyński wrote:

 Hi,

 We have two FS instances production (prod) and gateway (gw)

 when prod starts it tries to register a gateway:

 include
    gateway name=fs_gateway
        param name=realm value=24ccdomain/
        param name=proxy value=$${local_ip_v4}:15060 /
        param name=username value=24cc/
        param name=password value=24ccpass/
        param name=expire-seconds value=60/
        param name=register value=true/
    /gateway
 /include

 and fails:

 2009-10-29 10:19:45.449220 [NOTICE] sofia_glue.c:3534 Reload XML [Success]
 2009-10-29 10:19:45.450221 [INFO] mod_enum.c:808 ENUM Reloaded
 2009-10-29 10:19:45.450221 [NOTICE] sofia_reg.c:2117 Added gateway
 'fs_gateway' to profile 'external'
 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias
 [outbound] for profile [external]
 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [nat]
 for profile [external]
 2009-10-29 10:19:45.451221 [NOTICE] sofia.c:3108 Started Profile
 external [sofia_reg_external]
 2009-10-29 10:19:45.452221 [INFO] switch_time.c:661 Timezone reloaded
 530 definitions
 2009-10-29 10:19:45.628743 [NOTICE] sofia_reg.c:333 Registering fs_gateway
 2009-10-29 10:19:45.996790 [NOTICE] sofia.c:1202 Waiting for worker thread
 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:1729 Adding Alias
 [192.168.1.20] for profile [internal]
 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3093 Adding Alias
 [default] for profile [internal]
 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3108 Started Profile
 internal [sofia_reg_internal]
 2009-10-29 10:19:53.354724 [ERR] sofia_reg.c:1425 fs_gateway
 Registration Failed with status Forbidden [403]. failure #1
 2009-10-29 10:19:53.440735 [WARNING] sofia_reg.c:364 fs_gateway Failed
 Registration, setting retry to 60 seconds.
 2009-10-29 10:20:53.108812 [NOTICE] sofia_reg.c:333 Registering fs_gateway
 2009-10-29 10:20:53.116813 [ERR] sofia_reg.c:1425 fs_gateway
 Registration Failed with status Forbidden [403]. failure #2
 2009-10-29 10:20:53.208825 [WARNING] sofia_reg.c:364 fs_gateway Failed
 Registration, setting retry to 90 seconds.

 gw is configured like that:

 include
    domain name=24ccdomain
        params
            param name=dial-string
 value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
        /params

        groups
            group name=default
                users
                    user id=24cc
                        params
                            param name=password value=24ccpass/
                        /params
                        variables
                            variable name=user_context value=default/
                            variable name=effective_caller_id_name
 value=24 CallCenter/
                        /variables
                    /user
                /users
            /group
        /groups
    /domain
 /include

 and outputs:

 2009-10-29 10:11:29.165700 [WARNING] sofia_reg.c:1737 Can't find user
 [2...@192.168.1.20]
 You must define a domain called '192.168.1.20' in your directory and
 add a user with the id=24cc attribute
 and you must configure your device to use the proper domain in it's
 authentication credentials.

 on each registration attempt from prod.

 This worked perfectly fine with one of previous revisions but started
 failing on 15257. Using siptrace we were able to narrow down the issue
 to prod sending an IP address as realm instead of gateway configured
 realm (as it did in previous revision):

 Authorization: Digest username=24cc, realm=192.168.1.20,
 nonce=4d82d624-c884-6340-8f7f-a4575f2583bd,
 cnonce=Es23bj8MEi29T7UQ80i7sw, algorithm=MD5,
 uri=sip:192.168.1.20:15060,
 response=afea161876aad137657bf1f923d18659, qop=auth, nc=0004

 Is there any configuration setting we are missing that would bring
 back previous behavior?

 --
 Best regards,
 Robert Wilczynski

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Re: [Freeswitch-users] Freeswitch in signaling path only

2009-10-30 Thread DJB
Now i have as follows, but it's still the same result.  By the way, I am 
running:  FreeSWITCH Version 1.0.4 (exported)
.
.
.
session:execute(set,hangup_after_bridge=true)
session:execute(set,continue_on_fail=true)
session:execute(set,originate_timeout=2)
session:execute(set,originate_retries=3)
session:execute(set,progress_timeout=15)
.
.
.
while row do
 local gw_ip_address = row.gw_ip_address
 local cust_name = row.cust_name
 session:execute(set, accountcode= ..cust_name .. )
 session:execute(set,bypass_media=true)
 session:execute(bridge,sofia/external/ .. called_num .. @XX.XX.XX.XX.146)
 session:execute(set,bypass_media=true)
 session:execute(bridge,sofia/external/ .. called_num .. @XX.XX.XX.XX.105)
 -- Block for testing -- session:execute(bridge,sofia/external/ .. 
called_num .. @ .. gw_ip_address ..)
 row = cur:fetch (row, a)
end

Here is the debug for switch_ivr_originate.c:

2009-10-30 11:09:52.877832 [DEBUG] switch_ivr_originate.c:63 
(sofia/external/6463924...@xx.xx.xx.146) State Change CS_ROUTING - C
S_CONSUME_MEDIA
2009-10-30 11:09:53.17811 [DEBUG] switch_ivr_originate.c:2061 Originate 
Resulted in Success: [sofia/external/6463924...@xx.xx.xx.1
46]
2009-10-30 11:09:54.285453 [DEBUG] switch_ivr_originate.c:63 
(sofia/external/6463924...@xx.xx.xx.105) State Change CS_ROUTING - C
S_CONSUME_MEDIA
2009-10-30 11:09:54.422426 [DEBUG] switch_ivr_originate.c:2061 Originate 
Resulted in Success: [sofia/external/6463924...@xx.xx.xx.
105]
2009-10-30 11:09:55.694761 [DEBUG] switch_ivr_originate.c:63 
(sofia/external/6463924...@xx.xx.xx.146) State Change CS_ROUTING - C
S_CONSUME_MEDIA
2009-10-30 11:09:55.836036 [DEBUG] switch_ivr_originate.c:2061 Originate 
Resulted in Success: [sofia/external/6463924...@xx.xx.xx.
146]
2009-10-30 11:09:57.107697 [DEBUG] switch_ivr_originate.c:63 
(sofia/external/6463924...@xx.xx.xx.105) State Change CS_ROUTING - C
S_CONSUME_MEDIA
2009-10-30 11:09:57.254664 [DEBUG] switch_ivr_originate.c:2061 Originate 
Resulted in Success: [sofia/external/6463924...@xx.xx.xx.
105]
2009-10-30 11:12:03.129097 [DEBUG] switch_ivr_originate.c:63 
(sofia/external/6463924...@xx.xx.xx.146) State Change CS_ROUTING - C
S_CONSUME_MEDIA
2009-10-30 11:12:03.273055 [DEBUG] switch_ivr_originate.c:2061 Originate 
Resulted in Success: [sofia/external/6463924...@xx.xx.xx.
146]
2009-10-30 11:12:04.546410 [DEBUG] switch_ivr_originate.c:63 
(sofia/external/6463924...@xx.xx.xx.105) State Change CS_ROUTING - C
S_CONSUME_MEDIA
2009-10-30 11:12:04.682661 [DEBUG] switch_ivr_originate.c:2061 Originate 
Resulted in Success: [sofia/external/6463924...@xx.xx.xx.
105]
2009-10-30 11:12:15.781701 [DEBUG] switch_ivr_originate.c:2138 Originate 
Resulted in Error Cause: 16 [NORMAL_CLEARING]
2009-10-30 11:12:33.349162 [DEBUG] switch_ivr_originate.c:63 
(sofia/external/6463924...@xx.xx.xx.146) State Change CS_ROUTING - C
S_CONSUME_MEDIA
2009-10-30 11:12:33.470989 [DEBUG] switch_ivr_originate.c:2061 Originate 
Resulted in Success: [sofia/external/6463924...@xx.xx.xx.
146]
2009-10-30 11:12:34.724641 [DEBUG] switch_ivr_originate.c:2138 Originate 
Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
2009-10-30 11:12:34.730634 [DEBUG] switch_ivr_originate.c:2138 Originate 
Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
2009-10-30 11:12:34.750637 [DEBUG] switch_ivr_originate.c:2138 Originate 
Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]


FIRST ROUTE: XX.XX.XX.146 
and I tried to failed the first route and it gave 500 back, then it goes to the 
next one.
SECOND ROUTE:  XX.XX.XX.105


Thank you,
Dorn B.



- Original Message 
From: Chris Burns ch...@cloudtel.com
To: freeswitch-users@lists.freeswitch.org
Sent: Fri, October 30, 2009 10:00:01 AM
Subject: Re: [Freeswitch-users] Freeswitch in signaling path only

Do you have a debug level message from switch_ivr_originate.c in your log?
Channel is already up, delaying proxy mode 'till both legs are answered.

Set bypass_media b4 each bridge. It is unsetting on you and setting 
bypass_media_after_bridge because you already answered the channel running 
the lua script.

On October 30, 2009 12:03:29 pm DJB wrote:
 I am wondering why I cannot do as condition#2.

 For Lua in dialplan, when I have the followings:


 --WORKING--
 (Condition#1)
 .
 .
 session:execute(set,bypass_media=true)
 session:execute(set,hangup_after_bridge=true)
 session:execute(set,continue_on_fail=true)
 .
 .
  session:execute(bridge,sofia/external/ .. called_num ..
 @1.1.1.1|sofia/external/ .. called_num .. @1.1.1.2) .
 .

 --NOT WORKING--
 (Condition#2)
 Note:  FS tries to be in media path and send re-invite.
 .
 .
 session:execute(set,bypass_media=true)
 session:execute(set,hangup_after_bridge=true)
 session:execute(set,continue_on_fail=true)
 .
 .
  session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.1)
  session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.2)
 .
 .

 Thank you,
 Dorn B.




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[Freeswitch-users] Control Panel

2009-10-30 Thread Pedro Prado

Hi,

Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or other?

Thanks,
Pedro Prado
  
_
Você sabia que com o Hotmail você tem espaço ilimitado para guardar seus 
e-mails? Começe a usar já!
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Re: [Freeswitch-users] Control Panel

2009-10-30 Thread Traun Leyden
From watching the mailing list, I've seen:

95% are using straight XML combined with their home-cooked solutions based
on mod_xml_curl, which makes it pretty easy to write your own custom GUI for
freeswitch.

Out of the rest, the most are probably using pfsense, freepbx, and a small
handful of people using wikipbx (based on forum activity, guess would be
50ish).  There's also TCAPI (php).

Domain-based multi-tenancy is on the horizon for wikipbx, which has been
chugging along slowly, but has not died.  BroadTone (broadtone.net) has
kindly donated some test servers and a web developer, so kudos goes out to
them.

2009/10/30 Pedro Prado pedropr...@msn.com

  Hi,

 Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or
 other?

 Thanks,
 Pedro Prado

 --
 Navegue com segurança com o Novo Internet Explorer 8. Baixe agora, é
 gratis!http://brasil.microsoft.com.br/IE8/mergulhe/?utm_source=MSN%3BHotmailutm_medium=Taglineutm_campaign=IE8

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Re: [Freeswitch-users] Control Panel

2009-10-30 Thread Michael Collins
2009/10/30 Pedro Prado pedropr...@msn.com

  Hi,

 Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or
 other?

These are all at various stages of production/usability:
Most of the people here just use emacs/vim/etc. and then use the built-in
webserver for simple stuff. (see http://wiki.freeswitch.org/wiki/Webapi)

For the others:
WikiPBX is an enigma. I haven't heard of anything new in quite some time. I
don't recommend it.
FreePBX V3 - very promising, in early development, don't expect anything
usable right away
FusionPBX - Farther along, simpler than FreePBX, closest to being usable in
production
FSGUI - Donated to contrib by jmesquita; simple, work in progress,
interesting to play with

CudaTel - It's the bomb! ;)

-MC
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Re: [Freeswitch-users] PBX vs. SIP proxy?

2009-10-30 Thread Michael Collins
On Fri, Oct 30, 2009 at 5:06 AM, Fred-145 codecompl...@free.fr wrote:


 Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
 the registration process, while a PBX offers services like ringing phones,
 setting up calls, handling voice mail, etc.


Of course, as usual, the lines are getting blurred. Look at opensips.org and
see all the crazy things that a good SIP proxy can actually do.
-MC
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Re: [Freeswitch-users] PBX vs. SIP proxy?

2009-10-30 Thread Fred-145

Thanks for the link. But then, since OpenSIPs doesn't run on Windows, I'll
stick to FreeSwitch :)
-- 
View this message in context: 
http://old.nabble.com/PBX-vs.-SIP-proxy--tp25920788p26136363.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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

2009-10-30 Thread Ivan C Myrvold
Yes, now I got a more detailed trace. Thank you for helping me with  
this.

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

Ivan

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

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

 Then run your call, then do this:

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

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

 Ivan


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

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


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


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

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

 Ivan

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

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 --
 Eliot Gable

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

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

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

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 -- 
 Eliot Gable

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

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

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

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Re: [Freeswitch-users] Some documentation thoughts

2009-10-30 Thread Michael Collins
thanks for the heads up! I'll ask someone in the know to fill that out.
-MC

On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola diego.vi...@gmail.com wrote:

 This needs to be converted to documentation form also.


 http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used


 On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola diego.vi...@gmail.comwrote:

 Thanks, our wiki is getting a lot better :).

 Diego

 On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins m...@freeswitch.orgwrote:

 Diego, when I get some time I will help convert these from IRC to wiki.
 Thanks,
 MC


 On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola diego.vi...@gmail.comwrote:

 This needs to be converted to documentation form too.

 http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations

 Diego


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Re: [Freeswitch-users] Some documentation thoughts

2009-10-30 Thread Diego Viola
No problem, when I have some time I need to merge these two pages also.

http://wiki.freeswitch.org/wiki/Multi-tenant
http://wiki.freeswitch.org/wiki/Multiple_Companies

=D

Diego


On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins m...@freeswitch.org wrote:

 thanks for the heads up! I'll ask someone in the know to fill that out.
 -MC


 On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola diego.vi...@gmail.comwrote:

 This needs to be converted to documentation form also.


 http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used


 On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola diego.vi...@gmail.comwrote:

 Thanks, our wiki is getting a lot better :).

 Diego

 On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins m...@freeswitch.orgwrote:

 Diego, when I get some time I will help convert these from IRC to wiki.
 Thanks,
 MC


 On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola diego.vi...@gmail.comwrote:

 This needs to be converted to documentation form too.

 http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations

 Diego


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[Freeswitch-users] mod_t38gateway

2009-10-30 Thread Tihomir Culjaga
does anybody know how does it work and how to use it in a dialplan?


freeswi...@nemesis
freeswi...@nemesis
freeswi...@nemesis load mod_t38gateway
API CALL [load(mod_t38gateway)] output:
+OK

2009-10-30 22:44:38.204268 [NOTICE] mod_t38gateway.c:147 T.38 gateway
enabled
2009-10-30 22:44:38.204268 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_t38gateway]
2009-10-30 22:44:38.204268 [NOTICE] switch_loadable_module.c:248 Adding
Application 't38gateway'
2009-10-30 22:44:38.205374 [NOTICE] switch_loadable_module.c:270 Adding API
Function 't38gateway'
freeswi...@nemesis
freeswi...@nemesis
freeswi...@nemesis
freeswi...@nemesis t38gateway
API CALL [t38gateway()] output:
-USAGE: uuid command

freeswi...@nemesis

T.
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[Freeswitch-users] no REINVITE on Blind Transfer with bypass_media

2009-10-30 Thread Humberto Quintana

Hi everybody,

I'd like some help with this situation that is 'haunting' me :-) 

My scenario is as follows:

inbound-bypass-media is set in the profile because we dont want FS handling the 
media.

1. A calls B
2. FS sends to B the A's SDP
3. B answers
4. FS sends to A the B's SDP
5. Media going directly between A and B
6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
7. FS accepts(202) the REFER and sends the NOTIFY
7a. B and FS send the BYE
8. FS sends an INVITE  to C with A's SDP
9. C answers
10. FS doesn't send a reINVITE to A to let it know about C's SDP


Is that the expected FS behavior or is this a bug?


I'd appreciate any hints. Thanks,


Humberto


  
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Re: [Freeswitch-users] Some documentation thoughts

2009-10-30 Thread Diego Viola
Another one to convert:

http://wiki.freeswitch.org/wiki/User:Agx#CAMP_ON

On Fri, Oct 30, 2009 at 9:36 PM, Diego Viola diego.vi...@gmail.com wrote:

 No problem, when I have some time I need to merge these two pages also.

 http://wiki.freeswitch.org/wiki/Multi-tenant
 http://wiki.freeswitch.org/wiki/Multiple_Companies

 =D

 Diego



 On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins m...@freeswitch.orgwrote:

 thanks for the heads up! I'll ask someone in the know to fill that out.
 -MC


 On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola diego.vi...@gmail.comwrote:

 This needs to be converted to documentation form also.


 http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used


 On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola diego.vi...@gmail.comwrote:

 Thanks, our wiki is getting a lot better :).

 Diego

 On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins 
 m...@freeswitch.orgwrote:

 Diego, when I get some time I will help convert these from IRC to wiki.
 Thanks,
 MC


 On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola diego.vi...@gmail.comwrote:

 This needs to be converted to documentation form too.

 http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations

 Diego


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Re: [Freeswitch-users] Some documentation thoughts

2009-10-30 Thread Michael Collins
actually that's a user's personal page so you don't have to worry about that
one. :)
-MC

On Fri, Oct 30, 2009 at 3:09 PM, Diego Viola diego.vi...@gmail.com wrote:

 Another one to convert:

 http://wiki.freeswitch.org/wiki/User:Agx#CAMP_ON


 On Fri, Oct 30, 2009 at 9:36 PM, Diego Viola diego.vi...@gmail.comwrote:

 No problem, when I have some time I need to merge these two pages also.

 http://wiki.freeswitch.org/wiki/Multi-tenant
 http://wiki.freeswitch.org/wiki/Multiple_Companies

 =D

 Diego



 On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins m...@freeswitch.orgwrote:

 thanks for the heads up! I'll ask someone in the know to fill that out.
 -MC


 On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola diego.vi...@gmail.comwrote:

 This needs to be converted to documentation form also.


 http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used


 On Sat, Oct 17, 2009 at 6:12 AM, Diego Viola diego.vi...@gmail.comwrote:

 Thanks, our wiki is getting a lot better :).

 Diego

 On Sat, Oct 17, 2009 at 5:54 AM, Michael Collins 
 m...@freeswitch.orgwrote:

 Diego, when I get some time I will help convert these from IRC to
 wiki.
 Thanks,
 MC


 On Fri, Oct 16, 2009 at 8:50 PM, Diego Viola 
 diego.vi...@gmail.comwrote:

 This needs to be converted to documentation form too.

 http://wiki.freeswitch.org/wiki/Mod_vmd#Related_conversations

 Diego


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Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media

2009-10-30 Thread Brian West

First you forgot to mention what SVN rev you're on...

/b

On Oct 30, 2009, at 5:07 PM, Humberto Quintana wrote:


Hi everybody,

I'd like some help with this situation that is 'haunting' me :-)

My scenario is as follows:

inbound-bypass-media is set in the profile because we dont want FS  
handling the media.


1. A calls B
2. FS sends to B the A's SDP
3. B answers
4. FS sends to A the B's SDP
5. Media going directly between A and B
6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
7. FS accepts(202) the REFER and sends the NOTIFY
7a. B and FS send the BYE
8. FS sends an INVITE  to C with A's SDP
9. C answers
10. FS doesn't send a reINVITE to A to let it know about C's SDP


Is that the expected FS behavior or is this a bug?


I'd appreciate any hints. Thanks,


Humberto



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a deal on Windows 7 now  
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Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media

2009-10-30 Thread Anthony Minessale
profile options

param name=media-option value=resume-media-on-hold/
param name=media-option value=bypass-media-after-att-xfer/


On Fri, Oct 30, 2009 at 5:07 PM, Humberto Quintana hjqlo...@hotmail.comwrote:

  Hi everybody,

 I'd like some help with this situation that is 'haunting' me :-)

 My scenario is as follows:

 inbound-bypass-media is set in the profile because we dont want FS handling
 the media.

 1. A calls B
 2. FS sends to B the A's SDP
 3. B answers
 4. FS sends to A the B's SDP
 5. Media going directly between A and B
 6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
 7. FS accepts(202) the REFER and sends the NOTIFY
 7a. B and FS send the BYE
 8. FS sends an INVITE  to C with A's SDP
 9. C answers
 10. FS doesn't send a reINVITE to A to let it know about C's SDP


 Is that the expected FS behavior or is this a bug?


 I'd appreciate any hints. Thanks,


 Humberto



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 on Windows 7 now http://go.microsoft.com/?linkid=9691628

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