[Freeswitch-users] Difference between park and valet_park

2009-10-10 Thread velusamy velu
Dear All,
 Could you please any one explain the difference between parking and
valet parking?
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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-10-10 Thread Jason White
for those who are working on the Debian packages, the following ITP bug may be
of interest.
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=513606
which also refers to an older entry.

Thanks are due to all who are contributing to the packaging effort.


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Re: [Freeswitch-users] Difference between park and valet_park

2009-10-10 Thread Seven Du
search this list, just has been discussed.

2009/10/10 velusamy velu velu.techni...@gmail.com

 Dear All,
  Could you please any one explain the difference between parking and
 valet parking?

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[Freeswitch-users] Play music on hold after parking the call

2009-10-10 Thread velusamy velu
Dear All,
  I am using ESL.pm module to control the dial plan application. I want
to play some music while executing the some external scripts. I executed
park after then I executed the playback the music didn't play.

Could any one please explain how can I solve this problem without using
async mode in socket application?

Thanks,
Velusamy.
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[Freeswitch-users] how to match '#' in XML dialplan ?

2009-10-10 Thread daqiang wang
hello every one :
   I want to match the # in XML dialplan , how to do  ?
   example :
  1## . how to do ?
 I do this :
 condition field=destination_number expression=^1#(d+)#(d+)$/
but it's not work
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Re: [Freeswitch-users] directory variables not set correct with the latest trunk

2009-10-10 Thread Juan Backson
Hi,

My sip phone did authentication ( sending in SIP REGISTER etc ), but it is
still not showing up.  How come?

It was working before.

thanks,
juan

On Thu, Oct 8, 2009 at 11:21 PM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 they only set if you authenticate on sip or run the set_user app


 On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson juanback...@gmail.comwrote:

 Hi,

 My application fails to set the appropriate variables using directory xml
 after using the latest trunk as of yesterday.

 My curl looks like:



 document type=freeswitch/xml
   section name=directory
 domain name=192.168.1.102
   user id=22
 params
   param name=password value=123456/param
   param name=dial-string
 value=sofia/internal/sip:200...@192.168.1.11:29440
 ;rinstance=0245b2a59ddff837/param
 /params
 variables
   variable name=account_id value=20/variable
   variable name=vm-code value=1000/variable
 /variables
   /user
 /domain
   /section
 /document



 In the info app, I am not seeing account-id and vm-code anymore.

 How to fix that?

 juan

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

 AIM: anthm
 MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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 sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
 iax:gu...@conference.freeswitch.org/888
 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] Bad sound quality while eavesdropping

2009-10-10 Thread Maciej Aniserowicz

Hi,
Here are the messages with a:ptime parameter. All the calls are started by
commands sent through socket.
I'm not sure if this is all information you need, please let me know if
something is missing here and I'll post that.

1) starting connection with x-lite (number 2003, the eavesdropper):

   INVITE sip:2...@192.168.3.100:60188;rinstance=80b8f8d92af87cd2 SIP/2.0
   Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K
   Max-Forwards: 69
   From: MyApp sip:000...@192.168.3.159;tag=jpQ6D7D2jUXvF
   To: sip:2...@192.168.3.100:60188;rinstance=80b8f8d92af87cd2
   Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff
   CSeq: 121465610 INVITE
   Contact: sip:mod_so...@192.168.3.159:15060
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 447
   Remote-Party-ID: MyApp
sip:000...@192.168.3.159;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:115 G7221/32000
   a=fmtp:115 bitrate=48000
   a=rtpmap:107 G7221/16000
   a=fmtp:107 bitrate=32000
   a=rtpmap:9 G722/8000
   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


2) starting connection with cisco ip phone (number 2006, first leg of
eavesdropped session):

   INVITE sip:2...@192.168.2.106:5060;user=phone SIP/2.0
   Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p
   Max-Forwards: 69
   From: MyApp sip:000...@192.168.3.159;tag=Q3N2pe2K47ctS
   To: sip:2...@192.168.2.106:5060;user=phone
   Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff
   CSeq: 121465616 INVITE
   Contact: sip:mod_so...@192.168.3.159:15060
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 447
   Remote-Party-ID: MyApp
sip:000...@192.168.3.159;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:115 G7221/32000
   a=fmtp:115 bitrate=48000
   a=rtpmap:107 G7221/16000
   a=fmtp:107 bitrate=32000
   a=rtpmap:9 G722/8000
   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


3) starting connection with extension playing a file (number , second
leg of eavesdropped session):

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS
   From: FreeSWITCH sip:myu...@mydomain;transport=udp;tag=091j2Q0Fre8vp
   To: sip:9...@192.168.3.159:15060;tag=U7t5Xt51rB64Q
   Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8
   CSeq: 121465623 INVITE
   Contact: sip:mod_so...@192.168.3.159:15060;transport=udp
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 263
   
   v=0
   o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4 192.168.3.159
   s=FreeSWITCH
   c=IN IP4 192.168.3.159
   t=0 0
   m=audio 30086 RTP/AVP 0 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20




Anthony Minessale wrote:
 
 you probably have some device lying about ptime everywhere
 look at a sip trace an pay especially close attention to ptime:x param in
 sdp
 if you don't understand this just attach it here
 
 execute the following at the cli
 sofia profile internal siptrace on
 sofila loglevel debug
 
 
 
 On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz 
 maciej.aniserow...@gmail.com wrote:
 

 It's the same on the trunk (the last rev I used was not so old anyway).

 Codecs are the same on both legs:
 read codec/read rate: PCMU  8000
 

[Freeswitch-users] Anyone intairaisted?

2009-10-10 Thread Meftah Tayeb
dear all
anyone intairaisted in porting/integrating some billing engine/software 
to freeswitch?
we are building a new Community for that
no for billing only but billing is the primary focus
if anyone intairaisted please mail me to:
tayeb.mef...@gmail.com
i'm waiting for you to join!
the web site is not yet ready, is:
http://www.awesomevoip.org
we have also a hosted PBX to do conferencing... but we need a conference 
bridge
thanks!


__ Information from ESET NOD32 Antivirus, version of virus signature 
database 4494 (20091009) __

The message was checked by ESET NOD32 Antivirus.

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Re: [Freeswitch-users] how to match '#' in XML dialplan ?

2009-10-10 Thread Milena
escape character is '\'try
  condition field=destination_number expression=^1\#(d+)\#(d+)$/
2009/10/10 daqiang wang wangdq@gmail.com

 hello every one :
I want to match the # in XML dialplan , how to do  ?
example :
   1## . how to do ?
  I do this :
  condition field=destination_number expression=^1#(d+)#(d+)$/
 but it's not work

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Re: [Freeswitch-users] openzap Failure opening channel error

2009-10-10 Thread lakshmanan ganapathy
Hi,
I've tried with the following dialplan(After making the changes that are
recommended). But still I got INVALID_IE_CONTENTS error.

Another thing is, if I use prispan configuration, I'm able to make outgoing
calls.
But if I use the libpri span configuration I'm not able to make outgoing
calls, which says INVALID_IE_CONTENTS.

I've been struck with this problem for the past 1 week.
Any solution to this???


On Fri, Oct 9, 2009 at 6:06 PM, russell.mosem...@cune.org wrote:

 lakshmanan ganapathy lakindi...@gmail.com said:

  But still I'm facing problem with the outgoing call. It says
  INVALID_IE_CONTENTS.
  What might be the issue? Even I tried the following dialplan to call by
  using bridge.
  extension name=outgoing-fxo
condition field=destination_number
  expression=^(\d{10})$
action application=answer/
action application=bridge
  data=openzap/1/1/${dialed_ext}/

 Does answer need to be called here? I haven't used an fxo. So, I don't
 know. What value does $dialed_ext have? If you want to use the number
 matched in the condition, then it should be

 openzap/1/1/$1

 --
 Russell Mosemann



 
 Concordia University, Nebraska
 See http://www.cune.edu/ for the latest news and events!


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Re: [Freeswitch-users] Questions regarding to mod_nibble

2009-10-10 Thread João Mesquita
I am testing the latest version of nibblebill so let me see if I can help
you with your questions.

jm

On Fri, Oct 9, 2009 at 12:39 AM, Ahmed Munir ahmedmunir...@gmail.comwrote:

 I want to ask three questions related to mod_nibble bills, as I'm listing
 down below;

 1- Can we select/use dynamic tables for billing using nibble bill?

What do you mean for dynamic tables? Like LCR does that you can specify your
own SQL statement to be executed? If that's what you are asking, no, but it
would be a nice todo.


 2- Can we define more than two tables and attributes in
 nibblebill.conf.xml?

What else do you want to define and how do you imagine it to behave?


 3- As Nibble bill is use to deduct amount of user account, Can we deduct
 minutes instead of cash? Because my case is, if a user buy a package and I
 only want to deducts his/her minutes. How we can resolve it by nibble bill?
 / What other way we can resolve it?

When we say cash on the column, we are really saying just a number that is
being deducted, that's it. If you deduct 1 every 60 seconds, you will have
your cash converted to minutes, won't you?



 Kindly advise soon.

 --
 Regards,

 Ahmed Munir



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Re: [Freeswitch-users] Questions regarding to mod_nibble

2009-10-10 Thread Dome Charoenyost
2009/10/9 Ahmed Munir ahmedmunir...@gmail.com:
 I want to ask three questions related to mod_nibble bills, as I'm listing
 down below;

 1- Can we select/use dynamic tables for billing using nibble bill?
 2- Can we define more than two tables and attributes in nibblebill.conf.xml?
 3- As Nibble bill is use to deduct amount of user account, Can we deduct
 minutes instead of cash? Because my case is, if a user buy a package and I
 only want to deducts his/her minutes. How we can resolve it by nibble bill?
 / What other way we can resolve it?
My solution (Postgresql 8.4 with postgresql-prefix).
1. modify mod_nibblebill.c  change sqlupdate command to update
duration in my cdr table (insert blank data when start billing)
2. create trigger to deduct balance.


Dome C.

 Kindly advise soon.

 --
 Regards,

 Ahmed Munir



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Re: [Freeswitch-users] openzap Failure opening channel error

2009-10-10 Thread Michael Collins
Okay, please go to pastebin.freeswitch.org and paste your openzap.conf.xml
file. Also, paste the ENTIRE debug log from a call from start to finish.
Telling us that you see INVALID_IE_CONTENTS doesn't help if we don't know
what the information element contains. Finally, turn on PRI debugging and
make another test call and pastebin that debug log as well. The debug will
show details about the communications between your machine and the carrier.
Instructions for turning on debugging for libpri are found in the OpenZAP
wiki page in the same place where the libpri instructions are located.

Put the pastebin number in this email thread and then we'll go have a look.

Thanks,
MC

On Sat, Oct 10, 2009 at 6:47 AM, lakshmanan ganapathy
lakindi...@gmail.comwrote:

 Hi,
 I've tried with the following dialplan(After making the changes that are
 recommended). But still I got INVALID_IE_CONTENTS error.

 Another thing is, if I use prispan configuration, I'm able to make outgoing
 calls.
 But if I use the libpri span configuration I'm not able to make outgoing
 calls, which says INVALID_IE_CONTENTS.

 I've been struck with this problem for the past 1 week.
 Any solution to this???



 On Fri, Oct 9, 2009 at 6:06 PM, russell.mosem...@cune.org wrote:

 lakshmanan ganapathy lakindi...@gmail.com said:

  But still I'm facing problem with the outgoing call. It says
  INVALID_IE_CONTENTS.
  What might be the issue? Even I tried the following dialplan to call by
  using bridge.
  extension name=outgoing-fxo
condition field=destination_number
  expression=^(\d{10})$
action application=answer/
action application=bridge
  data=openzap/1/1/${dialed_ext}/

 Does answer need to be called here? I haven't used an fxo. So, I don't
 know. What value does $dialed_ext have? If you want to use the number
 matched in the condition, then it should be

 openzap/1/1/$1

 --
 Russell Mosemann



 
 Concordia University, Nebraska
 See http://www.cune.edu/ for the latest news and events!


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Re: [Freeswitch-users] Difference between park and valet_park

2009-10-10 Thread Michael Collins
On Fri, Oct 9, 2009 at 11:51 PM, velusamy velu velu.techni...@gmail.comwrote:

 Dear All,
  Could you please any one explain the difference between parking and
 valet parking?


http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park

Like the wiki says, valet_park puts calls into numbered parking stalls. A
call that is parked in a numbered stall is retrieved by calling (or
transferring) a second call into that numbered stall. That makes it easy to
pass calls around. The regular park app puts calls in a special park state
that is like being on hold, but the only way to get them out of park is to
uuid_bridge a second leg to the uuid of the parked call leg.

Read the wiki entry on valet_park and then read the one on park and you'll
see that they are similar apps but the valet_park app is a bit easier for
phone users to use for moving calls in and out of the park state.

-MC


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Re: [Freeswitch-users] how to match '#' in XML dialplan ?

2009-10-10 Thread Michael Collins
Some characters need a backslash to match in a regular expression. However,
# is not one of them. I think your regex is wrong:
condition field=destination_number expression=^1#(d+)#(d+)$/

It should probably be:
condition field=destination_number expression=^1#(\d+)#(\d+)$/

Note the backslashes in front of the d+ entries. \d means match a digit
whereas a bare d means make a lowercase d character.

Hope that helps.
-MC

P.S. - The * character does need to be escaped in regexes. See the
default.xml dialplan file for some obvious examples.

On Sat, Oct 10, 2009 at 6:24 AM, Milena testeado...@gmail.com wrote:

 escape character is '\'try
   condition field=destination_number expression=^1\#(d+)\#(d+)$/
 2009/10/10 daqiang wang wangdq@gmail.com

 hello every one :
I want to match the # in XML dialplan , how to do  ?
example :
   1## . how to do ?
  I do this :
  condition field=destination_number expression=^1#(d+)#(d+)$/
 but it's not work

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Re: [Freeswitch-users] Difference between park and valet_park

2009-10-10 Thread Anthony Minessale
park is an app you can never return from that just sits idle so you can
remote control the channel.
valet_park is an app to be used from the phones as a way to park and
retrieve calls.



On Sat, Oct 10, 2009 at 12:22 PM, Michael Collins m...@freeswitch.orgwrote:



 On Fri, Oct 9, 2009 at 11:51 PM, velusamy velu 
 velu.techni...@gmail.comwrote:

 Dear All,
  Could you please any one explain the difference between parking and
 valet parking?


 http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park

 Like the wiki says, valet_park puts calls into numbered parking stalls. A
 call that is parked in a numbered stall is retrieved by calling (or
 transferring) a second call into that numbered stall. That makes it easy to
 pass calls around. The regular park app puts calls in a special park state
 that is like being on hold, but the only way to get them out of park is to
 uuid_bridge a second leg to the uuid of the parked call leg.

 Read the wiki entry on valet_park and then read the one on park and you'll
 see that they are similar apps but the valet_park app is a bit easier for
 phone users to use for moving calls in and out of the park state.

 -MC


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Re: [Freeswitch-users] libpri_span vs prispans

2009-10-10 Thread Moises Silva
From libs/openzap/mod_openzap/mod_openzap.c and
libs/openzap/src/ozmod_libpri/ozmod_libpri.c

node=cpe|net
switch=ni1|ni2|dms100|lucent5|5ess|att4ess|4ess|euroisdn|gr303eoc|gr303tmc
dp=international|national|local|isdn|private|unknown
l1=alaw|ulaw -- this recently is set based on the trunk type, but old
openzap in E1 mode needs to explicitly set it to alaw
debug=q921_raw,q921_dump,q921_state,config,q931_dump,q931_state,q931_anomaly,apdu,acc
none or all
opts=suggest_channel,omit_display,omit_redirecting_number

The following values are commonly known.
context=
dialplan=

On Sat, Oct 10, 2009 at 12:39 AM, lakshmanan ganapathy lakindi...@gmail.com
 wrote:

 Where can I found entire configurations for a libpri spans. Right now I've
 only 4 things in my libpri configuration. They are Switchtype, Node,
 Contect, Dialplan, which is given as example in wiki.

 But I think libprispan has more configuration. Where can I find those
 things?



 On Fri, Oct 9, 2009 at 11:27 PM, Michael Collins m...@freeswitch.orgwrote:



 On Fri, Oct 9, 2009 at 5:57 AM, russell.mosem...@cune.org wrote:

 lakshmanan ganapathy lakindi...@gmail.com said:
  What difference it will make, if I use pri_span configuration and
 lib_pri
  span configuration in openzap.conf.xml.
  I'm preety much confused on the difference between this.

 Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
 libpri is more mature than openzap. In my case openzap was not able to
 handle everything invovling a T1 here, and using libpri fixed it.

 A libpri_span configuration is handled by libpri. A pri_span
 configuration is handled natively by openzap. Use the one that works for
 you.


 Right now the libpri stack is more mature than the ozmod_isdn stuff. See
 the OpenZAP page on the wiki for instructions on how to set up ozmod_libpri.
 -MC


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Re: [Freeswitch-users] directory variables not set correct with the latest trunk

2009-10-10 Thread Juan Backson
Hi,

I am still stuck in trying to get curl directory variables to show up in
channel even after the user has registered.

Is this something that gets changed in the latest trunk or is it just my
mis-configuration?

please help.

jb

On Sat, Oct 10, 2009 at 6:05 PM, Juan Backson juanback...@gmail.com wrote:

 Hi,

 My sip phone did authentication ( sending in SIP REGISTER etc ), but it is
 still not showing up.  How come?

 It was working before.

 thanks,
 juan


 On Thu, Oct 8, 2009 at 11:21 PM, Anthony Minessale 
 anthony.miness...@gmail.com wrote:

 they only set if you authenticate on sip or run the set_user app


   On Thu, Oct 8, 2009 at 5:20 AM, Juan Backson juanback...@gmail.comwrote:

  Hi,

 My application fails to set the appropriate variables using directory xml
 after using the latest trunk as of yesterday.

 My curl looks like:



 document type=freeswitch/xml
   section name=directory
 domain name=192.168.1.102
   user id=22
 params
   param name=password value=123456/param
   param name=dial-string
 value=sofia/internal/sip:200...@192.168.1.11:29440
 ;rinstance=0245b2a59ddff837/param
 /params
 variables
   variable name=account_id value=20/variable
   variable name=vm-code value=1000/variable
 /variables
   /user
 /domain
   /section
 /document



 In the info app, I am not seeing account-id and vm-code anymore.

 How to fix that?

 juan

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 ClueCon http://www.cluecon.com/
 Twitter: http://twitter.com/FreeSWITCH_wire

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 MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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 IRC: irc.freenode.net #freeswitch

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 sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
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 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] directory variables not set correct with the latest trunk

2009-10-10 Thread Brian West
It happens on invite during a call not on REGISTER... I can 100%  
confirm it works so I do not know what the heck you're doing.

/b

On Oct 10, 2009, at 6:02 PM, Juan Backson wrote:

 Hi,

 I am still stuck in trying to get curl directory variables to show  
 up in channel even after the user has registered.

 Is this something that gets changed in the latest trunk or is it  
 just my mis-configuration?

 please help.

 jb


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Re: [Freeswitch-users] directory variables not set correct with the latest trunk

2009-10-10 Thread Brian West
Also check to make sure you're returning the correct format as per the  
XML curl page.

/b

On Oct 10, 2009, at 6:02 PM, Juan Backson wrote:

 Hi,

 I am still stuck in trying to get curl directory variables to show  
 up in channel even after the user has registered.

 Is this something that gets changed in the latest trunk or is it  
 just my mis-configuration?

 please help.

 jb


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Re: [Freeswitch-users] how to match '#' in XML dialplan ?

2009-10-10 Thread daqiang wang
it's work . Thank you very much .

2009/10/11 Michael Collins m...@freeswitch.org

 Some characters need a backslash to match in a regular expression. However,
 # is not one of them. I think your regex is wrong:
 condition field=destination_number expression=^1#(d+)#(d+)$/

 It should probably be:
 condition field=destination_number expression=^1#(\d+)#(\d+)$/

 Note the backslashes in front of the d+ entries. \d means match a digit
 whereas a bare d means make a lowercase d character.

 Hope that helps.
 -MC

 P.S. - The * character does need to be escaped in regexes. See the
 default.xml dialplan file for some obvious examples.


 On Sat, Oct 10, 2009 at 6:24 AM, Milena testeado...@gmail.com wrote:

 escape character is '\'try
   condition field=destination_number expression=^1\#(d+)\#(d+)$/
 2009/10/10 daqiang wang wangdq@gmail.com

 hello every one :
I want to match the # in XML dialplan , how to do  ?
example :
   1## . how to do ?
  I do this :
  condition field=destination_number expression=^1#(d+)#(d+)$/
 but it's not work

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Re: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet Parking

2009-10-10 Thread mayamatakeshi
On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins m...@freeswitch.org wrote:

 FYI,

 The FreeSWITCH devs have added valet parking! Check it out:
 http://www.freeswitch.org/node/207

 Let us know what you think.


Very nice.

But I think a valet_unpark app is missing.
If the intention of the person sent to the valet lot is to retrieve a call
there, the person can assume the call was already retrieved by someone else
or that the caller hung up if he/she hears MOH. But it would be nicer to
have a valet_unpark app to fail and let the dialplan play a message.

regards,
takeshi
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Re: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet Parking

2009-10-10 Thread mayamatakeshi
On Sun, Oct 11, 2009 at 12:18 PM, mayamatakeshi mayamatake...@gmail.comwrote:



 On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins m...@freeswitch.orgwrote:

 FYI,

 The FreeSWITCH devs have added valet parking! Check it out:
 http://www.freeswitch.org/node/207

 Let us know what you think.


 Very nice.

 But I think a valet_unpark app is missing.
 If the intention of the person sent to the valet lot


I meant parking stall


 is to retrieve a call there, the person can assume the call was already
 retrieved by someone else or that the caller hung up if he/she hears MOH.
 But it would be nicer to have a valet_unpark app to fail and let the
 dialplan play a message.

 regards,
 takeshi

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