Re: [Freeswitch-users] APT Utility

[Shelby Ramsey]

David,

apt is the pack management system for the debian line of distros 
(including Ubuntu). Google apt-get or aptitude for more information on 
the utility.

This probably isn't the best list for topics not related to FS (apt is a 
linux utility) ... but here is a brief rundown:

apt-get install make flex patch gcc g++ autoconf automake libtool 
libncurses5-dev ncurses-dev python-MySQLdb subversion -y
cd /usr/src
## DO ONE OF THE FOLLOWING TO GET THE SRC (TRUNK IS BEST FOR NOW -- NO 
STABLE RELEASE)##
svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk
or
svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
cd freeswitch/
./bootstrap.sh
./configure
make
make install

Type that as you see it on the command line (do sudo -i and type in the 
root password first).

I will say that FS requires some basic knowledge of Linux to get running 
... and certainly to manage / maintain. Try digging around for some 
Ubuntu how to's for some basic info.

Hope this helps.

SDR



David V. Fansler wrote:
>
> Greetings – I am trying to startup a freeSwitch on a P4 running Ubuntu 
> 9.04 “Jaunty”. I know very little about Linux. I decided to try this 
> after reading the article in Linux Pro Magazine. I have been following 
> the detailed instructions in the wiki for using Ubuntu Jaunty, however 
> I have run into an unknown – “Use your favorite *APT* utility to get 
> the needed packages”.
>
> I am good at following direct instructions – but this statement is too 
> vague for my minimal minimal – did I mention minimal - knowledge of Linux.
>
> Could someone please give me detailed instructions on how to use APT 
> utility to get the needed packages – and what are the needed packages?
>
> Thanks kindly,
>
> David
>
> David V. Fansler
>
> s/v Annabelle
>
> dfans...@dv-fansler.com
>
> www.dv-fansler.com
>
> 
>
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Re: [Freeswitch-users] Hardware echo cancellation.

[Shelby Ramsey]

Given that it's an IVR system I think you'll find the DTMF results 
better with hardware based echo cans ... and to be frank the hardware 
based echo cancellations are really not that much more expensive on 
cards from folks like Sangoma.

SDR

Tim Uckun wrote:
> I am about to build a new machine as a VOIP server. I am going to get
> either a quad core intel or a six core AMD processor with at least
> eight gigabytes of RAM in it.   Given that much horsepower I am
> wondering if there is any need to purchase hardware with echo
> cancellation (I am thinking about redfone devices)..  I can save some
> money by not getting the echo cancellation.
>
> So is it worth saving that money? Is it always better to have hardware
> echo cancellation? Is a quad core capable of dealing with echo
> cancellation needs of an IVR which is going to take lots of
> simultaneous calls?
>
> Thanks.
>
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Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones

[Shelby Ramsey]

Peter,

Did you look at http://www.cudatel.com?  Probably just what you are 
looking for.  GUI goodness based on FS.

SDR

Peter J. Zandvoort wrote:
> Matthew, 
>
> I'm about in the same boat as you are, just on a smaller scale. We have a
> ton of Nortel telephony gear, but it's time to move out of the 90's and
> enter this millennium. My Cisco quote was in the same ballpark as yours. 
>
> The Cisco stuff is mature, rock solid, meshes very well with their network
> gear and is actually relatively easy to set up and maintain if you know your
> way around IOS. I just refuse to pay that kind of money for yet another
> semi-proprietary solution.
>
> After looking at various asterisk distributions, SipX, 3CX and
> what-have-you, I've come to the conclusion that FreeSWITCH is by far the
> most advanced platform out there. Its architecture and performance is
> literally light years ahead of the rest and I have yet to come up with
> something that it can't do. But all that comes at a price: The learning
> curve is like scaling a brick wall. The developers and the community are
> great and available, but just starting out with SIP and voip in general,
> this may not be the best platform. So let the blasphemy begin :)
>
> SipX was a breeze to install (insert CD, boot, next next next...) and looks
> pretty solid. I believe they actually use FreeSWITCH for their voicemail and
> conferencing, internally. I just couldn't get my head around their GUI, ACD
> was too basic and had all kinds of issues getting stuff to "just work".
>
> 3CX (Windows Only) was completely painless. It just worked. But I'm still
> not convinced that I want to run all my voice on a single windows box. Plus
> it's not free/open/etc and I don't want to lock myself in again.
>
> Although it's an asterisk based solution, I found trixbox to be very easy.
> Setup is automatic and everything "just worked". The GUI is simple and
> logical enough that I can let somebody else handle the day-to-day phone
> setup and basic admin. I have my doubts about it scaling to 250 users,
> though.
>
> This may be a completely flawed strategy and I may very well be shooting
> myself in the foot by doing this, but I plan on piloting a trixbox install
> with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH
> box next to it for the more advanced stuff. Once I get more comfortable with
> the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH,
> I have a feeling that that trixbox is going to get phased out...
>
> Peter
>
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> mkitchin.pub...@gmail.com
> Sent: Tuesday, November 03, 2009 11:10 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
>
> Michael Collins wrote:
>   
>> On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com 
>>  > > wrote:
>>
>> I'm working on an alternative to a $120,000 Cisco phone system that my
>>
>> company is looking at. I got Freeswitch installed on CentOS last week
>> using the Quick and Dirty instructions. That part was painless. We
>> had a
>> few 7940s laying around. After some wrestling with it, I got the
>> latest
>> SIP firmware installed and what I hoped was a functional config
>> (attached). X-Lite phones can call each other no problem. 7940s
>> can call
>> X-Lite no problem. Anytime I try and call a 7940, it goes straight to
>> voicemail. I attached a log file that shows the activity when
>> trying to
>> call a7940 from X-Lite.
>> X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is
>> nshplpbx1.unix/10.85.0.53 . Everything is on
>> the same LAN. Different
>> subnets, but no firewalls.
>> I didn't see anything that said posting attachments was frowned
>> upon. I
>> apologize if it isn't appropriate. I'm guessing this is something
>> simple
>> and I'm just clueless on how to diagnose the issue.
>> I'm not tied to using this model for good, but it is what we had
>> laying
>> around. Any help would be greatly appreciated. Next step is
>> configuring
>> it to talk to Verizon VOIP over a DS3.
>>
>> Thanks,
>> Matthew Kitchin
>>
>>
>> Matthew,
>> Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We 
>> think you'll find FS is as powerful as any software out there right now.
>>
>> Here's a handy wiki page that will help you get the diagnosing skills 
>> you need:
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>> I'd say first thing to do is capture the SIP traffic to see if there 
>> are any clues. A "normal temporary failure" doesn't give you a lot of 
>> detail. :) If you're new to SIP debugging then the best thing to do is 
>> to capture the SIP 

Re: [Freeswitch-users] IP Phones with FreeSwitch

[Shelby Ramsey]

Any of the Cisco phones with a SIP image should work fine ... no license 
required.

SDR

Dave Stevenson wrote:
> Hi again,
>  
> sorry to be here again !
>  
> OK, now that I know that 3Com phones and FreeSwitch don't mix, my next 
> question is about Cisco !
>  
> I see that the FreeSwitch Interoperability list includes Cisco phones 
> such as the 7940 and 7960.
>  
> I believe that these phones need user licenses to work with Cisco Call 
> Manager.
>  
> What I'd like to confirm is that I would not need any Cisco licenses 
> or anything else to get a Cisco IP phone working with FreeSwitch.
>  
> Again, I'd really appreciate feedback from anyone using either of 
> these (or other) Cisco phones with FreeSwitch on whether any 
> additional licenses or software are required to work with an "out of 
> the box" FreeSwitch installation ?
>  
> regards
> Dave
> 
>
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Re: [Freeswitch-users] Dial Plan Question

[Shelby Ramsey]

I think the real question is what are you trying to do ... for some 
things it's very easy to just whip up a static XML file and be done with 
it.  For others you probably want some sort of interaction with a DB. 

The options here are pretty endless:
--   XML curl
 -- handing off the call to a script call from a static dial plan 
(use lua if there is going to be any load)
--   event_socket
--   mod_lcr

But ultimately I think it's what you're trying to accomplish that 
matters.  For a PBX install I'd say static files is probably about as 
easy as it is going to get.  For delivering a service you'd probably 
want interaction with a DB.  I've use XML curl a lot and have even 
starting using direct DB queries from static dialplans using 
mod_memcache and memcachedb (not memcache ... persistent storage).

SDR





Jerry Richards wrote:
> My understanding of DialPlan/CallRouting is that it can be accomplished via
> static XML tags, or alternatively, via a DialPlan Application that
> interfaces with the dptools module.
>
> Question:  If my above assumption is true, how does one select one approach
> over the other?  What is the criteria/considerations that would govern the
> decision?
>
> Best Regards,
> Jerry
>
>
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Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question

[Shelby Ramsey]

Dave,

As I mentioned before ... the bind_server_ip directive is only for 
mod_dingaling.  I don't think sofia will use it.

As for IRC ... I have no knowledge of Windows machines or apps ... but I 
know there is an IRC client for FireFox called chatzilla.

SDR



Dave Stevenson wrote:
> Thanks Shelby,
>
> the vars.xml file tells me that if external_sip_ip is unspecified (which I 
> think it is), then the bind_server_ip is used, i.e., "auto"
>
> Getting off topic, I stumbled on freenode today, I wasn't sure what I needed 
> to do to get IRC running on Windows - yet, another "to do" !
>
> (The link on http://wiki.freeswitch.org/wiki/Main_Page to IRC Chat 
> (irc://irc.freenode.net/freeswitch) appears to be an invalid link)
>
>
> regards
> Dave
>
>
> - Original Message - 
> From: "Shelby Ramsey" 
> To: 
> Sent: Saturday, October 31, 2009 7:07 PM
> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>
>
>   
>> Dave,
>>
>> I'm not sure how the file structure works on Windows.  But sofia is the
>> SIP endpoint (and what you're wanting to confgure).
>>
>> The bind_server setting only relates to mod_dingaling (I think).  Yes
>> you can change the IP address ... but you'll need to restart FS or
>> reload the sofia profiles.  The conf settings for external_sip_ip, etc.
>> are the ones that matter.
>>
>> Good luck!  You can also hop on freenode @ #freeswitch and everyone
>> there is quite helpful.
>>
>> SDR
>>
>>
>>
>>
>> Dave Stevenson wrote:
>> 
>>> Hi Shelby,
>>>
>>> thanks for the reply on the IP Address Question.
>>>
>>> I do not have a $FS_DIR/conf/sofia_profiles/ directory ?
>>>
>>> $FS_DIR/conf/vars.xml has a line :-
>>> 
>>>
>>> I'd guess that this means that the Server uses the machines IP address
>>> automatically, so, I can change the IP address to fixed and don't need to
>>> make any changes at the Server end - just need to update the Clients.
>>>
>>> regards
>>> Dave
>>>
>>>
>>> - Original Message - 
>>> From: "Shelby Ramsey" 
>>> To: 
>>> Sent: Saturday, October 31, 2009 6:13 PM
>>> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>>>
>>>
>>>
>>>   
>>>> Dave,
>>>>
>>>> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
>>>> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
>>>> or use the API commands for reloading the sofia profiles.
>>>>
>>>> You'll definitely want to set a static IP address on the FS server (or
>>>> make sure the DHCP server always assigns the same IP address to the FS
>>>> server).  Otherwise you'll constantly be changing the clients that have
>>>> to connect to FS.
>>>>
>>>> SDR
>>>>
>>>>
>>>> Dave Stevenson wrote:
>>>>
>>>> 
>>>>> Hi,
>>>>>
>>>>> I am new to FreeSwitch  - and VOIP too (as the following questions
>>>>> will indicate).
>>>>>
>>>>> I am trying to setup a VOIP system at home and am just getting
>>>>> started, I have been trawling through the Docs/Wiki and realise that I
>>>>> have a lot to learn !
>>>>>
>>>>> Anyway, here's the first (of what I'm sure will be many !) questions,
>>>>> hopefully someone will take the type to help me out..
>>>>>
>>>>> OK, I have already worked out that FreeSwitch seems to be much better
>>>>> supported under Linux than Windows (is that a fair comment ?), but
>>>>> while I have a long term intention to give Linux a go, it's something
>>>>> that is on the back burner (and has been for some time). At the
>>>>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>>>>> have loaded the latest Windows Install. Everything seems to be working
>>>>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>>>>> interface won't be here until next week, so at the moment, I'm just
>>>>> trying out the system with X-Lite.
>>>>>
>>>>> The first question that I have is about WAPSERVER. This was installed
>>>>> as part of th

Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question

[Shelby Ramsey]

Dave,

I'm not sure how the file structure works on Windows.  But sofia is the 
SIP endpoint (and what you're wanting to confgure).

The bind_server setting only relates to mod_dingaling (I think).  Yes 
you can change the IP address ... but you'll need to restart FS or 
reload the sofia profiles.  The conf settings for external_sip_ip, etc. 
are the ones that matter. 

Good luck!  You can also hop on freenode @ #freeswitch and everyone 
there is quite helpful.

SDR




Dave Stevenson wrote:
> Hi Shelby,
>
> thanks for the reply on the IP Address Question.
>
> I do not have a $FS_DIR/conf/sofia_profiles/ directory ?
>
> $FS_DIR/conf/vars.xml has a line :-
> 
>
> I'd guess that this means that the Server uses the machines IP address 
> automatically, so, I can change the IP address to fixed and don't need to 
> make any changes at the Server end - just need to update the Clients.
>
> regards
> Dave
>
>
> - Original Message - 
> From: "Shelby Ramsey" 
> To: 
> Sent: Saturday, October 31, 2009 6:13 PM
> Subject: Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question
>
>
>   
>> Dave,
>>
>> You should check out the settings in $FS_DIR/conf/sofia_profiles/ and
>> $FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS
>> or use the API commands for reloading the sofia profiles.
>>
>> You'll definitely want to set a static IP address on the FS server (or
>> make sure the DHCP server always assigns the same IP address to the FS
>> server).  Otherwise you'll constantly be changing the clients that have
>> to connect to FS.
>>
>> SDR
>>
>>
>> Dave Stevenson wrote:
>> 
>>> Hi,
>>>
>>> I am new to FreeSwitch  - and VOIP too (as the following questions
>>> will indicate).
>>>
>>> I am trying to setup a VOIP system at home and am just getting
>>> started, I have been trawling through the Docs/Wiki and realise that I
>>> have a lot to learn !
>>>
>>> Anyway, here's the first (of what I'm sure will be many !) questions,
>>> hopefully someone will take the type to help me out..
>>>
>>> OK, I have already worked out that FreeSwitch seems to be much better
>>> supported under Linux than Windows (is that a fair comment ?), but
>>> while I have a long term intention to give Linux a go, it's something
>>> that is on the back burner (and has been for some time). At the
>>> moment, the current plan is to use FreeSwitch under WindowsXP and I
>>> have loaded the latest Windows Install. Everything seems to be working
>>> fine so far - well, in test mode anyway, my VOIP phones and teleco
>>> interface won't be here until next week, so at the moment, I'm just
>>> trying out the system with X-Lite.
>>>
>>> The first question that I have is about WAPSERVER. This was installed
>>> as part of the FreeSwitch install. At the moment, I am manually
>>> starting FreeSwitch and WAPSERVER does not seem to start
>>> automatically. When I manually start it, it says that it is OFFLINE
>>> until I manually put it ONLINE. So :-
>>>
>>> 1. What does WAPSERVER actually do ?
>>> 2. How should it be started ?
>>> 3. My next step is to configure the Server PC to auto-logon to my
>>> Windows 2003 Server domain and put FreeSwitch in the Startup Group -
>>> do I need to add WAPSERVER there too ?
>>>
>>> The next question is about the ServerIP address. Not thinking far
>>> enough ahead, the machine that I installed FreeSwitch on uses DHCP to
>>> get its IP Address. I guess that the FreeSwitch server should really
>>> have a static IP Address. If I change the PC to have a static IP
>>> Address, then
>>>
>>> 1. Do I need to re-install FreeSwitch
>>> 2. If not, is there a or is there command to get FreeSwitch to updated
>>> itself with a new IP Address ?
>>> 3. Maybe the Server does not need to know the change - maybe if only
>>> affect the clients (Phones/SoftPhones ?)
>>>
>>> Thanks & Best Regards
>>> Dave
>>> 
>>>
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Re: [Freeswitch-users] New FreeSwitch User - WAPSERVER Question

[Shelby Ramsey]

Dave,

You should check out the settings in $FS_DIR/conf/sofia_profiles/ and 
$FS_DIR/conf/vars.xml.  To make the changes take effect just restart FS 
or use the API commands for reloading the sofia profiles.

You'll definitely want to set a static IP address on the FS server (or 
make sure the DHCP server always assigns the same IP address to the FS 
server).  Otherwise you'll constantly be changing the clients that have 
to connect to FS.

SDR


Dave Stevenson wrote:
> Hi,
>  
> I am new to FreeSwitch  - and VOIP too (as the following questions 
> will indicate).
>  
> I am trying to setup a VOIP system at home and am just getting 
> started, I have been trawling through the Docs/Wiki and realise that I 
> have a lot to learn !
>  
> Anyway, here's the first (of what I'm sure will be many !) questions, 
> hopefully someone will take the type to help me out..
>  
> OK, I have already worked out that FreeSwitch seems to be much better 
> supported under Linux than Windows (is that a fair comment ?), but 
> while I have a long term intention to give Linux a go, it's something 
> that is on the back burner (and has been for some time). At the 
> moment, the current plan is to use FreeSwitch under WindowsXP and I 
> have loaded the latest Windows Install. Everything seems to be working 
> fine so far - well, in test mode anyway, my VOIP phones and teleco 
> interface won't be here until next week, so at the moment, I'm just 
> trying out the system with X-Lite.
>  
> The first question that I have is about WAPSERVER. This was installed 
> as part of the FreeSwitch install. At the moment, I am manually 
> starting FreeSwitch and WAPSERVER does not seem to start 
> automatically. When I manually start it, it says that it is OFFLINE 
> until I manually put it ONLINE. So :-
>  
> 1. What does WAPSERVER actually do ?
> 2. How should it be started ?
> 3. My next step is to configure the Server PC to auto-logon to my 
> Windows 2003 Server domain and put FreeSwitch in the Startup Group - 
> do I need to add WAPSERVER there too ?
>  
> The next question is about the ServerIP address. Not thinking far 
> enough ahead, the machine that I installed FreeSwitch on uses DHCP to 
> get its IP Address. I guess that the FreeSwitch server should really 
> have a static IP Address. If I change the PC to have a static IP 
> Address, then
>  
> 1. Do I need to re-install FreeSwitch
> 2. If not, is there a or is there command to get FreeSwitch to updated 
> itself with a new IP Address ?
> 3. Maybe the Server does not need to know the change - maybe if only 
> affect the clients (Phones/SoftPhones ?)
>  
> Thanks & Best Regards
> Dave
> 
>
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Re: [Freeswitch-users] IVR Intro Clipped

[Shelby Ramsey]

Jerry,

Put the sleep after the answer.  That should fix it.

Shelby

Jerry Richards wrote:
> I modified my dialplan as shown, but the clipping persists.  Should the
> sleep be placed somewhere else?
>
> 
>   
>  
>  
>  
>  
>   
> 
>
> Best Regards,
> Jerry
>  
>
> -Original Message-
> From: Brian West [mailto:br...@freeswitch.org] 
> Sent: Wednesday, October 28, 2009 1:51 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] IVR Intro Clipped
>
> Sleep 1000 ms... we usually bring up media too fast before the other end is
> ready.
>
> /b
>
> On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:
>
>   
>> I notice that when I call IVR from the PSTN, the "Welcome to 
>> Freeswitch..."
>> introduction is clipped at the beginning, so it sounds like "come to 
>> Freeswitch".  If I call 5000 internally, then I always hear the full 
>> introduction.  What can I do to resolve this?
>>
>> My XML config looks like:
>>
>> 
>>   
>>  
>>  
>>  
>>   
>> 
>>
>> Best Regards,
>> Jerry
>>
>>
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>> 
>
>
>
>
>
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Re: [Freeswitch-users] Estimating Call Capacity

[Shelby Ramsey]

Cliff,

Try using xml_rpc ... status or show channels will give you what you need.

SDR

Cliff Wells wrote:
> A little off-topic, but since call-capacity is the subject, what are
> people using to analyze their CDR's to discover this?   I'm handling
> about 30k calls per day but have only a bandwidth-based guesstimate of
> the peak number of concurrent calls I'm handling.
>
> If there's an open source solution, I'd appreciate a pointer.
>
> Regards,
> Cliff
>
> On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
>   
>> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
>> server from FS with FS on a Dell R710 with their current best
>> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
>> GB memory. The web app server is less than half the power of the R710.
>> I maxed the web app server at 300 calls per second (both setting up
>> and tearing down) and the R710 running FS was 65% idle. No audio was
>> being proxied through FS, though. If I were running the web app server
>> on an equivalent R710, they probably would have been on-par with each
>> other in performance. Extrapolating, I expect that in such a case I
>> should be able to get at least 650 CPS out of FS, though for
>> production I would probably limit it to 400 CPS or less so I leave
>> room for miscellaneous tasks. I maxed out the R710 at over 16,000
>> simultaneous calls (again, no audio proxying) but the only reason I
>> couldn't do more was because I hit some sort of thread creation limit
>> in Linux. There was about 17 GB of memory used for this many calls.
>> This should give you some ballpark idea of what you can accomplish
>> with FS.
>>
>> At some point, I will track down and resolve the thread creation
>> issue, at which time I believe call limits will be limited either by a
>> complex combination of available memory, the speed of the processor,
>> the cost of thread context switching, calls per second setup rate, and
>> call duration.
>>
>> --
>> Eliot Gable
>>
>> 
>>> -Original Message-
>>>
>>> From: freeswitch-users-boun...@lists.freeswitch.org 
>>> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of 
>>> Giovanni Maruzzelli
>>>
>>> Sent: Monday, October 26, 2009 4:56 PM
>>>
>>> To: freeswitch-users@lists.freeswitch.org
>>>
>>> Subject: Re: [Freeswitch-users] Estimating Call Capacity
>>>
>>>
>>>
>>> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
>>>
>>>  wrote:
>>>
>>>   
 Here are a few benchmarks that I had stumbled upon.
 
 http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
 
>>>
>>> Please remember NO benchmarks are endorsed by the FS community or
>>>
>>> developers, because there are just too many variables, and a simple
>>>
>>> figure is just useful for marketing hype, not for real dimensioning.
>>>
>>>
>>>
>>> You MUST do your own benchmarking, so you get an idea about how to
>>>
>>> dimension for your own use case and hardware.
>>>
>>>
>>>
>>>
>>>
>>>   
 Thanks,
 
 Vinuth.
 
 On Tue, Oct 27, 2009 at 1:43 AM, Brian West  wrote:
 
> I highly doubt it... You can wait for someone to post their results
>   
> but in the end you'll have to do your own load testing because not
>   
> everyone's numbers will jive with your use case.  Which is the reason
>   
> the project never posts or endorses a set call count.
>   
> /b
>   
> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>   
>> Are there any benchmarking test results available publicly?
>> 
>> 
>> 
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>>>
>>>
>>>
>>>
>>>
>>> --
>>>
>>> Sincerely,
>>>
>>>
>>>
>>> Giovanni Maruzzelli
>>>
>>> Cell : +39-347-2665618
>>>
>>>
>>>
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Re: [Freeswitch-users] dynamic gateway

[Shelby Ramsey]

Hello:

Your issue is that this portion of the above string "sipinterface_1" has to
be a valid sip profile.  To do this:
  -- determine the appropriate sip profile you want to use (usually
external)
  -- send the traffic to whatever ip you want via this portion of your
string $...@192.168.111.101:5060

You can't dynamically change the /sipinterface_1/ parameter.

SDR
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Re: [Freeswitch-users] Hi Guys

[Shelby Ramsey]

Well ... the question is pretty generic.

But based on these assumptions:
  -- no media (bypass media)
  -- routing done via XML dialplan

Something along the lines of a quad core machine with 4 gigs of ram would be
overkill for 692 calls.

Things to remember:
  -- the more cores the better (FS is heavily threaded)
  -- the more memory the better
  -- 64 bit is way better than 32 bit

SDR





>
>
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Re: [Freeswitch-users] Hi Guys

[Shelby Ramsey]

Question:

Will the call flow look like this (above was not very clear):

web --> FS --> Cantata --> PSTN (via TDM circuits)

Or are you trying to replace the Cantata?

SDR
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Re: [Freeswitch-users] Can freeswitch forward an un-implemented SIP method?(like a SIP proxy server)

[Shelby Ramsey]

This is because FS is a B2BUA ... not a proxy.  You should consider
OpenSER/SIPS/Kaemillio for this type of application.

SDR
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Re: [Freeswitch-users] Dialplan Issue

[Shelby Ramsey]

Just to confirm ... works like a champ.

Thanks again!!!
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Re: [Freeswitch-users] Dialplan Issue

[Shelby Ramsey]

Brian,

Thanks for the info.  I guess I'll go read section 19.1 of RFC3261 again.  I
do think the above has a valid host portion (I don't think the port is
required).

I'm not so sure that putting params in the user portion of the uri is valid
(from the RFC it states sip:user:passw...@host:port;uri-parameters?headers).

The issue is that in the real world this is done all the time  SIP is
fantastic :)

Shelby


On Thu, Oct 1, 2009 at 9:42 AM, Brian West  wrote:

>
> On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:
>
> 
>
>
> This will produce an INVALID sip uri... You can not feed this to sofia
> it'll get PISSED.
>
> Its missing the host portion.
>
>
> So the issue is the placement of the user params  if they are before
> the @ FS will send a 500 internal server error ... if they are after the @
> FS will send a 302.  Unfortunately placing the user params after the @
> doesn't quite conform to the way other devices expect to receive the 302 for
> this application.
>
> Any help would be greatly appreciated.
>
> Shelby
>
> PS ... hats off to the author of mod_memcache ... that is extremely useful!
>
>
>
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Re: [Freeswitch-users] Dialplan Issue

[Shelby Ramsey]

Tony,

Once again ... you are the man!

I'll try this right now.

SDR
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[Freeswitch-users] Dialplan Issue

[Shelby Ramsey]

Hello:

I asked this on IRC yesterday and I think I confused everyone involved.  So
I apologize in advance here for reposting the question and if I wasted
anyone's time.

So here is the issue I'm having.  I'm trying to use FS as a redirect server
(specifically to serve up LNP queries via 302 redirects).  But I'm having an
issue where based on the string in the dialplan FS will respond with a 500
internal error message instead of a 300 redirect.

The call flow should be this:
   -- remote party sends an Invite to my FS instance
   -- FS should respond with a 302

The following works as expected (FS will send a 302 when it receives an
Invite):



However if I do this (which is the way the response should look) FS will
respond with a 500 internal server error:



So the issue is the placement of the user params  if they are before the
@ FS will send a 500 internal server error ... if they are after the @ FS
will send a 302.  Unfortunately placing the user params after the @ doesn't
quite conform to the way other devices expect to receive the 302 for this
application.

Any help would be greatly appreciated.

Shelby

PS ... hats off to the author of mod_memcache ... that is extremely useful!
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Re: [Freeswitch-users] [Freeswitch-dev] Please Help: TDM Hardware For Testing

[Shelby Ramsey]

MC,

Just sent you the message offline ...

SDR

Michael Collins wrote:
> Hello all!
>
> We could use your help with something. We've had several volunteers 
> willing to assist with testing and debugging various scenarios with 
> TDM circuits. However, there is a need to get TDM hardware into the 
> hands of those volunteers. If you have old TDM hardware, be it analog 
> or digital, that is sitting around collecting dust and you'd like to 
> donate it (or loan it) to the cause then please email me off list.
>
> Thanks!
> -MC
> 
>
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Re: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid

[Shelby Ramsey]

PAI is defined @ http://www.ietf.org/rfc/rfc3325.txt

For what they are trying to do Sonus suggested P-Charge-Info (which I think
it is still in the draft stage) -->
http://www.ietf.org/internet-drafts/draft-york-sipping-p-charge-info-06.txt
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Re: [Freeswitch-users] Gafachi no passing caller number

[Shelby Ramsey]

Klaus,

Use ngrep and see if the From / RPID headers are correct in the SIP
message.  This will let you know if FS is doing the correct thing.

SDR
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Re: [Freeswitch-users] Clustering Freeswitch

[Shelby Ramsey]

Andres,

OpenSIP's works very well as a load balancer.  You could also use DNS SRV
(if the clients support it), round robin DNS 

Really just depends on what you are trying to accomplish.

SDR
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Re: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5

[Shelby Ramsey]

use the -nc flag ... that will do the trick
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Re: [Freeswitch-users] new module: mod_memcache

[Shelby Ramsey]

Rupa,

This is a big contribution!  Thanks!  Can't wait to play with this.

SDR
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Re: [Freeswitch-users] Buzzing when people speak in conference

[Shelby Ramsey]

I have in a previous life seen this quite a bit with the PolyCom phones ...
people tend to put their phone on the speaker on conference calls and I have
seen this type of interference caused by a computer speaker and even a
motorola cell phone.

So I would first force everyone to use the handset 1st ... if that solves it
then track down the guilty speaker or cell phone.

SDR
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Re: [Freeswitch-users] live iso image with freeswitch

[Shelby Ramsey]

Here is a really good start --> http://wiki.freeswitch.org/wiki/SBC_Setup

Overall I think you're going to just have to install Linux on a box and get
after it.  It may be more painful in the short term ... but in the long term
your life will be much better.

The devs / community here are pretty incredible with their desire and
efforts to help everyone (and all platforms) ... but in all reality it's a
huge task.  If you stick with Linux / FS everything will just work and there
is a tremendous amount of resources on the web.

SDR
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Re: [Freeswitch-users] Is mod vmd working?

[Shelby Ramsey]

Mark,

Because it didn't detect a "beep".  It will be be there as vmd_detect=true
if it does.  I'm not sure exactly how reliable it's "beep" detection is.

SDR
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Re: [Freeswitch-users] Is mod vmd working?

[Shelby Ramsey]

Mark,

It does work ... but I can't really attest to how well ... especially
compared to other things out there.  I started capturing this in CDR's to
see and it didn't seem like it worked very well.

If this is really critical to you, you might want to ping Ken Rice.  I know
he might have a better option.

SDR
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Re: [Freeswitch-users] Outbound Codec

[Shelby Ramsey]

Yes.  You can set this in the sip profile settings ... and then when it's
called in the bridge statement it will just work.

For example you take a call on profile external (which allows multiple
codecs) and then bridge it via profile internal (which only allows one
codec).

SDR
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Re: [Freeswitch-users] Rewriting Remote Party ID

[Shelby Ramsey]

Anthony,

That is awesome.  This is something that will be a BIG help.

SDR



>
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Re: [Freeswitch-users] To do telephony functions from web page

[Shelby Ramsey]

Rex,

I've never actually used PHP for this type of thing ... but you might want
to start by looking here:

http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/php/single_command.php?r=12216

or

http://wiki.freeswitch.org/wiki/PHP_Event_Socket

Good luck.  I'm sure some other folks here who use PHP for this type of app
will be able to assist more.

SDR
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Re: [Freeswitch-users] To do telephony functions from web page

[Shelby Ramsey]

Rex:

The basis for xml_rpc or mod_event is something along the lines of:

api $command

As an example to originate a call (using xml_rpc or mod_event) you would do:

api originate sofia/external/$some...@$ip:$PORT $EXTENSION xml $context

What language are you trying to do this in?

SDR
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Re: [Freeswitch-users] Compile Errors ...

[Shelby Ramsey]

Thanks for the help.  That did the trick.

SDR
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Re: [Freeswitch-users] Compile Errors ...

[Shelby Ramsey]

Did this (to make sure I started from scratch):

rm -rf /usr/src/freeswitch.trunk
rm -rf /usr/local/freeswitch
svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk
cd freeswitch.trunk/
./bootstrap.sh
./configure
make

... and then it pukes all over the place with errors compiling odbc support
... but I thought that odbc was disabled by default.

Errors look like this:

Compiling src/switch_odbc.c ...
In file included from src/switch_odbc.c:33:
./src/include/switch_odbc.h:36:17: error: sql.h: No such file or directory
./src/include/switch_odbc.h:43:20: error: sqlext.h: No such file or
directory
./src/include/switch_odbc.h:45:22: error: sqltypes.h: No such file or
directory
In file included from src/switch_odbc.c:33:
./src/include/switch_odbc.h:66: error: expected declaration specifiers or
'...' before 'SQLHSTMT'
./src/include/switch_odbc.h:96: error: expected declaration specifiers or
'...' before 'SQLHSTMT'
src/switch_odbc.c:43: error: expected specifier-qualifier-list before
'SQLHENV'
src/switch_odbc.c: In function 'switch_odbc_handle_new':
src/switch_odbc.c:76: error: 'switch_odbc_handle_t' has no member named
'env'
src/switch_odbc.c:76: error: 'SQL_NULL_HANDLE' undeclared (first use in this
function)
src/switch_odbc.c:76: error: (Each undeclared identifier is reported only
once
src/switch_odbc.c:76: error: for each function it appears in.)
src/switch_odbc.c:77: error: 'switch_odbc_handle_t' has no member named
'state'
src/switch_odbc.c: In function 'switch_odbc_handle_disconnect':
src/switch_odbc.c:96: error: 'switch_odbc_handle_t' has no member named
'state'
cc1: warnings being treated as errors
src/switch_odbc.c:97: warning: implicit declaration of function
'SQLDisconnect'
src/switch_odbc.c:97: error: 'switch_odbc_handle_t' has no member named
'con'
src/switch_odbc.c:105: error: 'switch_odbc_handle_t' has no member named
'state'
src/switch_odbc.c: In function 'switch_odbc_handle_connect':
src/switch_odbc.c:113: error: 'SQLINTEGER' undeclared (first use in this
function)
src/switch_odbc.c:113: error: expected ';' before 'err'
src/switch_odbc.c:116: error: 'SQLSMALLINT' undeclared (first use in this
function)
src/switch_odbc.c:116: error: expected ';' before 'valueLength'


SDR
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Re: [Freeswitch-users] Compile Errors ...

[Shelby Ramsey]

Yeah ... that's what I did when I first got the error.  I'll try it again.

Thanks for the help!

SDR
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Re: [Freeswitch-users] Compile Errors ...

[Shelby Ramsey]

Mike,

the entire output can be seen @
http://www.sipinterchange.com/downloads/fs_compile_err.txt

I don't see anything else.  It's really odd ...

SDR
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Re: [Freeswitch-users] Compile Errors ...

[Shelby Ramsey]

Mike,

This is what I get when I run make core (after I did a checkout on latest
svn trunk, ./bootstrap.sh, ./configure):

src/switch_console.c:35:28: error: switch_version.h: No such file or
directory
src/switch_console.c: In function 'switch_console_process':
src/switch_console.c:233: error: 'SWITCH_VERSION_FULL' undeclared (first use
in this function)
src/switch_console.c:233: error: (Each undeclared identifier is reported
only once
src/switch_console.c:233: error: for each function it appears in.)
make[1]: *** [libfreeswitch_la-switch_console.lo] Error 1
make: *** [core] Error 2

Thanks!

SDR
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[Freeswitch-users] Compile Errors ...

[Shelby Ramsey]

Hello,

I'm getting this all over the place today:

make[5]: *** No rule to make target `/usr/src/freeswitch/libfreeswitch.la',
needed by `mod_commands.so'.  Stop.
make[4]: *** [all] Error 1
make[3]: *** [mod_commands-all] Error 1
make[2]: *** [all-recursive] Error 1

I normally do this:

svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk
./bootstrap.sh
./configure
make
make install

Thx!

SDR
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Re: [Freeswitch-users] SIP dump to DB

[Shelby Ramsey]

Sorry .. I didn't give enough detail.  My point was to dump it via NGREP ...
parse it using something else to get it into a database where it would be
usable.  Then you can match calls from the CDR (using the UUID) to the
database.  The benefit is that you don't have to put the burden on your FS
boxes to do it ... Just monitor from another device and then dump it into
the database.  Of course you better have a beast of database if you want to
do 10,000 writes per second :) or be running something like NDB that scales
well.

There are other tools like scapy as well that can be quite useful in fact.

Shelby
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Re: [Freeswitch-users] SIP dump to DB

[Shelby Ramsey]

Why not just use NGREP and then dump the packets at a more reasonable pace?
You aren't going to be able to analysis in real time anyway.
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Re: [Freeswitch-users] Sending channel variables

[Shelby Ramsey]

I'm assuming that you are saying these are 2 boxes  if the protocol is a
sip you can append a sip header ... _sip_h_X-  This should be available
as a channel variable on FS A.
SDR

On Fri, Feb 13, 2009 at 6:10 AM, Evgeniy Zolotov  wrote:

>Hello!
>
> I'm trying to make such scheme:
>
> ---> FS_A --> FS_B --> record
>
> Incoming calls to FS_A are redirected to FS_B with the help of this
> context:
>
> 
>  
>
>
>data="sofia/outbound/$...@1.2.3.4:5080" />
>
>  
> 
>
> FS_B records them to the file:
>
> 
>  
>  
>  
>   data="$${base_dir}/recordings/test/testrec.wav" />
>  
> 
>
> This works good. But I have a question - in what manner I can send back
> (from FS_B to FS_A) some channel variables?
>
>   Thanks, Evgeniy.
>
>
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Re: [Freeswitch-users] FS equiv for waitforextension

[Shelby Ramsey]

Nik,

I'm not sure if this is the right way ... but I use application="read"
data="0 1 /path/silence.wav var 1000 #

I'm sure there is a better way ... but this seems to work for me.

SDR

On Thu, Feb 12, 2009 at 11:51 AM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>  HI,
>
>
>
> Is there an equivalent function in FS to waitforexten ?  Closest I've seen
> is collectInput?
>
>
>
> Right now I'm using stream file, which is ok if they hit a digit before
> stream ends, but I want them to have a certain period after the file is
> played to hit a button.
>
>
>
> Regards,
>
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Re: [Freeswitch-users] Dynamic Dialplan

[Shelby Ramsey]

Doug,
Ken is right on this one.  I know there are some guys on the list (like Ken)
that could help you write a module.  It's probably the best way to go (if
you're going to have all agents running off of one or two boxes).

If you're going to spread the agents / calls around on multiple boxes or use
a combo of OpenSIPS / OpenSer / pick your flavor ... and FS then using
xml_curl will work fine.  We've got that working today and it's been
acceptable.

SDR
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Re: [Freeswitch-users] AMD Functionality

[Shelby Ramsey]

Nik,

Right now there is mod_vmd.  It sets the channel variable vmd_detect if it
detects a beep.  If a beep is detected it will set vmd_detect=TRUE.  If no
beep is detected then it won't do anything.

Example of usage as follows (with the outcome being hangup if answering
machine is detected):


  




  
  


  


And Ken Rice also has a mod that can be licensed that probably is a better
solution and works much better.

Hope this helps.

SDR
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Re: [Freeswitch-users] Call accounting - CDR's

[Shelby Ramsey]

Even better ... Thanks Ken!

SDR
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Re: [Freeswitch-users] Call accounting - CDR's

[Shelby Ramsey]

Just out of curiosity ... You actually set the values in xml_cdr_conf.xml to
an invalid value ... and then FS tries it and then dumps it into the
err_dir?

Nik,

Just configure the xml_conf_cdr and it will post all of the channel
variables to your web server ... you can look at the variables and see what
you want.

Or I actually like Ken's suggestion ... that makes a lot of sense ... same
benefit of having all of the channel variables ... no overhead.

SDR
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Re: [Freeswitch-users] Call accounting - CDR's

[Shelby Ramsey]

Nik,

There are a bunch of ways to do this ... mod_xml_cdr posts to a url then you
can parse and dump ... or you can use mod_cdr_csv which allows you to
dictate exactly what you want to collect and then parse the file and dump
into mysql.

There are also a couple of examples here -->
http://wiki.freeswitch.org/wiki/Mod_cdr for hacking up the cdr_csv.conf and
making it do what you want.

SDR
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Re: [Freeswitch-users] Transcoding G723

[Shelby Ramsey]

Steve,

You definitely have a better grasp on this topic than me.  But I think it's
a tough sell on the host based processing ... when you look at products like
what audio codes can do on a card (3 DS-3's worth of transcoding) ... but I
have had a couple of soft switch vendors claim though that they could do
2,000 calls per host (but I seriously doubt it).

I agree that producing a card that does 120 channels is pretty worthless ...
but having something that could do say a 1000 would be very helpful.

G723 is a pretty big deal internationally ... I'm even seeing crazy requests
like AMR from folks trying to originate VoIP off of mobile devices in
Europe.

But I agree just being able to g729 --> ulaw or ulaw --> g729 would be a
great first step (host based or otherwise).

SDR
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Re: [Freeswitch-users] Transcoding G723

[Shelby Ramsey]

Thanks Moises.  It looks like good work.  When is Sangoma coming out with a
similar product ... Doug told me it was in the works, then not in the works,
then back in the works ...
The problem is this particular card is PCI only and it will only do 120
channels 

Thanks!

SDR
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Re: [Freeswitch-users] Transcoding G723

[Shelby Ramsey]

>
> Brian,
>

Thanks for the link.  Is anyone using this in the real world?  I did think
it was interesting that the author was from Sangoma ...

SDR
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Re: [Freeswitch-users] Transcoding G723

[Shelby Ramsey]

This is a tough deal ... and something that I do think keeps FS out of the
same boat as proprietary solutions like Nextone, etc.  In the real world
(from a service provider view) you have customers who want to send one thing
(i.e. g729 and g723 a lot from international carriers) and your vendors who
will only accept a limited set (specifically g729 (maybe) and certainly g711
ulaw).  So you have to really restrict what people can send you and in some
cases it can be a deal killer.  I'm seeing more and more wholesale vendors
(especially smaller niche guys) getting away from accepting anything other
than g711.

I would be interested in seeing if there would be a way to have the RTP
transverse a media processing blade like the ones offered from Audiocodes
etc.
Most have some method to tell the device to set up ports and bridge without
being involved in the signaling itself.

There are a couple of major advantages:
  -- removing the transcoding from the host to risc based processors
  -- not worrying about the licensing because it comes with the card and
would support all codecs

SDR
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Re: [Freeswitch-users] XML CDR ERROR ...

[Shelby Ramsey]

Just to make sure I'm clear:  -- start FS (without  in
modules.conf.xml)
  -- but have xml_cdr.conf.xml in autoload configs
  -- then reload xml
  -- then execute load mod_xml_cdr via the api

That seems like a challenging way to start FS ...

SDR

On Thu, Feb 5, 2009 at 1:42 AM, Brian West  wrote:

> Make sure your config file is installed and issue a reloadxml then
> load mod_xml_cdr
>
> /b
>
> On Feb 5, 2009, at 1:33 AM, Shelby Ramsey wrote:
>
> > Hello,
> >
> > I'm having it on both Fedora and Ubuntu boxes:
> >
> > 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open
> > of xml_cdr.conf failed
> > 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839
> > switch_loadable_module_load_file() Error Loading module /usr/local/
> > freeswitch/mod/mod_xml_cdr.so
> > **Module load routine returned an error**
> >
> > Details:
> >   -- Ubuntu 6.04 LTS
> >   -- Fedora 8
> >
> > Tried a couple of things:
> >   -- messing with libcurl
> >   -- ./configure --without-libcurl
> >
> > Thanks!
> >
> > SDR
>
>
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[Freeswitch-users] XML CDR ERROR ...

[Shelby Ramsey]

Hello,
I'm having it on both Fedora and Ubuntu boxes:

2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open of
xml_cdr.conf failed
2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_xml_cdr.so
**Module load routine returned an error**

Details:
  -- Ubuntu 6.04 LTS
  -- Fedora 8

Tried a couple of things:
  -- messing with libcurl
  -- ./configure --without-libcurl

Thanks!

SDR
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Re: [Freeswitch-users] Generating calls from external source

[Shelby Ramsey]

Nik,
There are a lot of ways to make FS dial out and deliver messaging etc.  We
are going through the process of replacing * for this purpose.  For us
(getting started with the help of our friends here on the list) it has been
pretty easy.

With * we were using AMI to originate calls ... to migrate to FS we just
changed that to use event_socket with bgapi to originate the call and
connect the call to a context and extension.  There are several ways to get
the dialplan to FS after that ... a script, xml_curl, or statically
configured in the conf directory.

So as an example the application we have just logs into the FS socket
(similar to * but much better) and then rips off calls like this:

bgapi originate{$set_some_vars}sofia/external/$...@$ip:$PORT $EXTENSION xml
$CONTEXT

The beauty of it all is that:
  -- a lot of flexibility in what you can do (like drive the call through
events)
  -- the CDR reporting is about 3 million times better than *
  -- obviously higher capacity

I'd start playing with event_socket and some static dialplans to get the
feel for it ... but if you have an application written already to work with
* (i.e. the logic and backend) it will be very easy to migrate and you'll be
glad you did it!

Shelby



On Tue, Feb 3, 2009 at 10:53 AM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

> Are you suggesting that I should process the call externally instead of
> using the dialplan?  That would be neat as the audio file select could
> be driven from the db select for the number.  I presume that I could
> also bridge the call to another number as well dependant on DTMF
> selection?
>
> Regards
>
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Raul
> Fragoso
> Sent: 03 February 2009 13:12
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Generating calls from external source
>
> In addition do David's suggestion, you probably want to have your
> application to watch for some specific events after the call is
> originated and take action based on them. For example, you could watch
> for the CHANNEL_ANSWER event and play some audio file waiting for some
> digit, which is generated by the DTMF event.
> To watch only for those specific events, you should do the following
> just after authentication (still using Perl as an example, but the
> mod_event_socket is language agnostic), then you will receive those
> events from FreeSWITCH through the socket stream:
>
> ...
> print $sock "auth XXX\n\n";
> print $sock "event plain CHANNEL_ANSWER DTMF\n\n";
> ...
>
> To see a list of available events, please look at the following wiki
> pages:
> http://wiki.freeswitch.org/wiki/Mod_event_socket#event
> http://wiki.freeswitch.org/wiki/Event_list
>
> Regards,
>
> Raul
>
> On Tue, 2009-02-03 at 09:46 +, David Knell wrote:
> > Hi Nik,
> >
> >
> > Here's a snipped in Perl that launches an outbound call:
> >
> >
> > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr =>
> > '127.0.0.1', PeerPort => 8021)) {
> > print $sock "auth XXX\n\n";
> > print $sock "api originate {softivr_id=$siid,src_softivr_id=
> > $siid,softivr_outdial=true}sofia/frombt/$...@1.2.3.4 $service\n\n";
> > $sock->close();
> > }
> >
> >
> > - it does no error checking or anything, but (line by line) it:
> >  - opens a socket to the event socket interface
> >  - authenticates
> >  - issues an originate which dials out to the number in $ntd.  The
> > bits in {} set a bunch of variables on the channel, which are used by
> > the software which processes the call later on.  The call is linked to
> > the extension in $service - FS looks this up in the dialplan - which
> > handles our end.
> >  - closes the socket
> >
> >
> > Cheers --
> >
> >
> > Dave
> >
> >
> >
> > > Thanks for that, coming from a C++ background it's a refreshing
> > > change to be looking at something that seems logical and efficient.
> > >
> > > I'd briefly looked at the event socket and wondered if that was the
> > > way to go.  I presume that there's some sort of event generation
> > > that can trigger and external process as well somewhere, though all
> > > I need to do is update mysql (hopefully using some sort of pooled
> > > connection)
> > >
> > > I'm not using a TDM card, I have a direct interconnect with the PSTN
> > > breakout provider with 1,500 channels available to me.  I'm finding
> > > Asterisk proving to be less than stable at high call volumes and
> > > load values spike at more than 100 calls with billing/accounting in
> > > place, hence my interest in FS.  The only thing that's concerning me
> > > is XML at the moment.  Lots of code and very wordy.  I'm sure I'll
> > > appreciate why XML given time
> > >
> > > Regards,
> > >
> > >
> > > 
> > > From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.

Re: [Freeswitch-users] auto dialing question ...

[Shelby Ramsey]

MC / Anthony,

Muchas gracias ... looks like I'm going to be on my way to being * free :)
 Ran a small production test today (about 20,000 dials) and going to get
after it tomorrow with a real campaign.

You guys kick ass.

Thanks again for the assistance fine sirs.

Shelby


On Fri, Jan 23, 2009 at 6:54 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> its the same idea where i gave you that example to call 9998 call 1234
> where your extension is delivered via curl when 1234 is requested
> use the "read" and "transfer" apps in the xml you return, you can get it
> working statically first in your regular dialplan.
>
>
>
> On Fri, Jan 23, 2009 at 6:20 PM, Shelby Ramsey wrote:
>
>> Thanks Michael.
>> Here are some really simple examples (we limit what people do through the
>> web ... nothing really fancy ... just some good old fashion robo calls).
>> ultimately today it looks like this extensions_table (context, exten,
>> priority, app, appdata):
>>
>> campaign,100,1,ANSWER() --> answers call
>> campaign,100,2,WAIT(1) --> this is a pause
>> campaign,100,3,PLAYBACK($INTROFILE) --> plays intro file
>> campaign,100,4,BACKGROUND($MESSAGE) --> plays message
>> campaign,100,5,WAITEXTEN(5) --> WAIT for DTMF
>> campaign,100,6,HANGUP()
>> Here are DTMF options
>> campaign,1,SYSTEM($SCRIPT campaign $EXTEN)
>> campaign,2,SYSTEM($SCRIPT campaign $EXTEN)
>> campaign,3,DIAL,zap/g1/$ANI
>>
>> anothercampaign,112,1,ANSWER()
>> anothercampaign,112,2,BACKGROUND($SOMEFILE)
>> anothercampaign,112,3,WAITEXTEN(5)
>> anothercampaign,112,4,HANGUP()
>> Options:
>> anothercampaign,1,DIAL(sip/$...@$ip)
>> anothercampaign,2,PLAYBACK($GOODBYE)
>>
>> And they pretty much all look like that ... it's easy to return the stuff
>> for extension 100 via XML ... but the challenge is the DTMF options
>> (relating to the same context) ... or maybe I'm just missing something
>> (which is a definite possibility).  We don't ever do anything complicated in
>> the IVRs (TTS or ASR) but there is just a lot of them that all get
>> controlled, manipulated via a web interface.
>>
>> I can originate a call ... do all of that ... just trying to figure out a
>> "simple" way to return the above via xml.
>>
>> Thanks for your help Anthony and Michael as always!
>>
>> Shelby
>>
>>
>>
>>
>>
>> On Fri, Jan 23, 2009 at 5:55 PM, Michael Collins wrote:
>>
>>> I see your dilemma. To keep things dynamic you definitely want to use
>>> your XML_CURL stuff. Like you said, nothing static.
>>>
>>> Can you post a sample call-flow in plain English? I'm curious about
>>> something. I don't want to say anything else. Just post a simple call
>>> flow:
>>> Initiate call
>>> Wait for answer/busy/timeout/invalid
>>> On busy/timeout/invalid: update db and move on
>>> On answer play greeting, accept digit, route based on digit
>>>
>>> Something like that would help me conceptualize what you are trying to
>>> do.
>>>
>>> Thanks
>>> MC
>>>
>>> P.S. - I've done a little bit of outbound IVR calling so hopefully I
>>> can assist you.
>>>
>>> On Fri, Jan 23, 2009 at 3:39 PM, Shelby Ramsey 
>>> wrote:
>>> > Anthony / Michael,
>>> > Thanks for the quick responses.  What I don't want to do is "drive the
>>> call"
>>> > (by that listen on a socket ... do this on this event ... or anything
>>> else
>>> > that my very limited FS foo would break) ... Just want to start it and
>>> then
>>> > give it instructions on where to go.
>>> > So I guess a better question would be ... how do I give directions to
>>> FS for
>>> > this (and I get the 1st part ... that's obvious ... really lost on the
>>> DTMF
>>> > digit part) ... and please keep in mind we're talking hundreds of
>>> extensions
>>> > / IVR's and distributed machines so I can't have any dependancy on
>>> static
>>> > conf files other than maybe something like what Michael mentioned where
>>> I
>>> > point every call to something:
>>> > [campaign]
>>> > exten => 100,1,ANSWER()
>>> > exten => 100,n,PLAYBACK(somefile)
>>> > exten => 100,n,BACKGROUND(somefile)
>>> > exten => 100,n,WAITEXTEN(4)
>>> > exten => 100,n,HANGUP()
>>> > b

Re: [Freeswitch-users] auto dialing question ...

[Shelby Ramsey]

Thanks Michael.
Here are some really simple examples (we limit what people do through the
web ... nothing really fancy ... just some good old fashion robo calls).
ultimately today it looks like this extensions_table (context, exten,
priority, app, appdata):

campaign,100,1,ANSWER() --> answers call
campaign,100,2,WAIT(1) --> this is a pause
campaign,100,3,PLAYBACK($INTROFILE) --> plays intro file
campaign,100,4,BACKGROUND($MESSAGE) --> plays message
campaign,100,5,WAITEXTEN(5) --> WAIT for DTMF
campaign,100,6,HANGUP()
Here are DTMF options
campaign,1,SYSTEM($SCRIPT campaign $EXTEN)
campaign,2,SYSTEM($SCRIPT campaign $EXTEN)
campaign,3,DIAL,zap/g1/$ANI

anothercampaign,112,1,ANSWER()
anothercampaign,112,2,BACKGROUND($SOMEFILE)
anothercampaign,112,3,WAITEXTEN(5)
anothercampaign,112,4,HANGUP()
Options:
anothercampaign,1,DIAL(sip/$...@$ip)
anothercampaign,2,PLAYBACK($GOODBYE)

And they pretty much all look like that ... it's easy to return the stuff
for extension 100 via XML ... but the challenge is the DTMF options
(relating to the same context) ... or maybe I'm just missing something
(which is a definite possibility).  We don't ever do anything complicated in
the IVRs (TTS or ASR) but there is just a lot of them that all get
controlled, manipulated via a web interface.

I can originate a call ... do all of that ... just trying to figure out a
"simple" way to return the above via xml.

Thanks for your help Anthony and Michael as always!

Shelby





On Fri, Jan 23, 2009 at 5:55 PM, Michael Collins  wrote:

> I see your dilemma. To keep things dynamic you definitely want to use
> your XML_CURL stuff. Like you said, nothing static.
>
> Can you post a sample call-flow in plain English? I'm curious about
> something. I don't want to say anything else. Just post a simple call
> flow:
> Initiate call
> Wait for answer/busy/timeout/invalid
> On busy/timeout/invalid: update db and move on
> On answer play greeting, accept digit, route based on digit
>
> Something like that would help me conceptualize what you are trying to do.
>
> Thanks
> MC
>
> P.S. - I've done a little bit of outbound IVR calling so hopefully I
> can assist you.
>
> On Fri, Jan 23, 2009 at 3:39 PM, Shelby Ramsey 
> wrote:
> > Anthony / Michael,
> > Thanks for the quick responses.  What I don't want to do is "drive the
> call"
> > (by that listen on a socket ... do this on this event ... or anything
> else
> > that my very limited FS foo would break) ... Just want to start it and
> then
> > give it instructions on where to go.
> > So I guess a better question would be ... how do I give directions to FS
> for
> > this (and I get the 1st part ... that's obvious ... really lost on the
> DTMF
> > digit part) ... and please keep in mind we're talking hundreds of
> extensions
> > / IVR's and distributed machines so I can't have any dependancy on static
> > conf files other than maybe something like what Michael mentioned where I
> > point every call to something:
> > [campaign]
> > exten => 100,1,ANSWER()
> > exten => 100,n,PLAYBACK(somefile)
> > exten => 100,n,BACKGROUND(somefile)
> > exten => 100,n,WAITEXTEN(4)
> > exten => 100,n,HANGUP()
> > but in that same context is someone triggers DTMF:
> > exten => 1,1,DOSOMETHING
> > exten => 2,1,DOSOMETHING
> > I was imaging issuing originate via XML_RPC ... something like originate
> > sofia/$...@$ip $SOMEEXTEN then on answer when FS tries to connect to
> > $SOMEEXTEN it will ask me what to do via xml_curl ... where I would
> normally
> > respond with something like this:
> >   > type="freeswitch/xml"> 
> >> field="destination_number" expression="">  > data="hangup_after_bridge=true"/>  > data="continue_on_fail=true"/>  > data="call_timeout=180"/>  > data="proxy_media=true"/>  > data="pass_rfc2833=true"/>  > data="accountcode=$CUSTOMER" />  > data="origination_caller_id_name=NULL" />  > data="origination_caller_id_number=$CIDNUM" />  > data="effective_caller_id_name=NULL" />  > data="effective_caller_id_number=$CIDNUM" />  > data="userfield=$BUNCHOFCRAPFORMYCDR" />  > data="sofia/external/$...@$providerip" />  
> > 
> >  
> > The challenge I've got is I have no idea how to do stuff like the IVR
> > mentioned above (the playback part is easy) ... but I can't grasp
> > conceptually how to get the "context" with "multiple extensions" part
> back
> > to F

Re: [Freeswitch-users] auto dialing question ...

[Shelby Ramsey]

Anthony / Michael,
Thanks for the quick responses.  What I don't want to do is "drive the call"
(by that listen on a socket ... do this on this event ... or anything else
that my very limited FS foo would break) ... Just want to start it and then
give it instructions on where to go.

So I guess a better question would be ... how do I give directions to FS for
this (and I get the 1st part ... that's obvious ... really lost on the DTMF
digit part) ... and please keep in mind we're talking hundreds of extensions
/ IVR's and distributed machines so I can't have any dependancy on static
conf files other than maybe something like what Michael mentioned where I
point every call to something:

[campaign]
exten => 100,1,ANSWER()
exten => 100,n,PLAYBACK(somefile)
exten => 100,n,BACKGROUND(somefile)
exten => 100,n,WAITEXTEN(4)
exten => 100,n,HANGUP()

but in that same context is someone triggers DTMF:
exten => 1,1,DOSOMETHING
exten => 2,1,DOSOMETHING

I was imaging issuing originate via XML_RPC ... something like originate
sofia/$...@$ip $SOMEEXTEN then on answer when FS tries to connect to
$SOMEEXTEN it will ask me what to do via xml_curl ... where I would normally
respond with something like this:

  


 

The challenge I've got is I have no idea how to do stuff like the IVR
mentioned above (the playback part is easy) ... but I can't grasp
conceptually how to get the "context" with "multiple extensions" part back
to FS via this method (is it possible?)...

Sorry for what is probably a very simple answer and any AST references (but
I've been using it in heavy production environments for about 5 years). Just
trying to "port" what I do today without making my brain melt out of my ears
(and it doesn't take much for that to happen).

Shelby

PS ... Really enjoy the list. I usually fall out of my chair laughing once a
day from your remarks Anthony. Keep it coming!


On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> Does AST mean Asterisk Open Source PBX ?
>
> If so, then yes I am familiar with it's archetechure as I am a former
> developer from that project.
>
> You have 3 choices with FreeSWITCH
>
> 1) You can open a dedicated connection to mod_event_socket or XMLRPC per
> call and issue the originate command from there:
> This will block until you know for sure the outcome of the attempt.  If
> it's success it will give you the uuid if not it gives you the cause code.
>
> 2) You can use a single mod_event_socket or XMLRPC connection to send all
> calls but use the bgapi mechanism which will do the same as above
> only asynchronously, The command will return immediately and the result
> will be fired as an event that you can pick up on the same or different
> event_socket connection or
> other event consumer such as a custom C,perl,lua etc module.
>
> 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call files
> that will tell you when where and why the calls failed or did not fail.
>
>
>
> On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins wrote:
>
>> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey 
>> wrote:
>> > Sorry for the double post ... actually hit send too early ...
>> > OK ... Here goes another I'm doing this with AST  ... but I want to move
>> it
>> > to FS.  Searched via google site:lists.freeswitch.org auto dialer and
>> others
>> > ... nothing useful.
>> > Today I have a platform for auto dialing with AST (centrally managed ...
>> > about 10 machines) and we do this:
>> >   -- Remote machines query central DB for numbers to call based on
>> certain
>> > configs
>> >   -- Use AMI to generate the call
>> >   -- If call gets answered, extension info queried via rta (central db
>> > again)
>> > The nice thing about all of this is it's relatively easy to manage
>> (through
>> > one central web interface we built) and it works ... the bad part is
>> > reporting ...
>> > So ... conceptually I'm trying to accomplish the same thing ...
>> > Today we use FS a lot for termination of VoIP traffic ... all done via
>> > XML_CURL ... which is awesome  (not to xml cdr ... and the "proxying" of
>> > media) ...
>> > Would like to do something like:
>> >   -- originate request (looks simple enough)
>> >   -- on answer XML_CURL posts info
>>
>> Several choices, depending upon how much you want it handled inside
>> the dialplan vs. handled in the scripting language. For the sake of
>> testing you could do something like this:
>> 
>>  
>>
>>
>&

[Freeswitch-users] auto dialing question ...

[Shelby Ramsey]

Sorry for the double post ... actually hit send too early ...

OK ... Here goes another I'm doing this with AST  ... but I want to move it
to FS.  Searched via google site:lists.freeswitch.org auto dialer and others
... nothing useful.
Today I have a platform for auto dialing with AST (centrally managed ...
about 10 machines) and we do this:
  -- Remote machines query central DB for numbers to call based on certain
configs
  -- Use AMI to generate the call
  -- If call gets answered, extension info queried via rta (central db
again)

The nice thing about all of this is it's relatively easy to manage (through
one central web interface we built) and it works ... the bad part is
reporting ...

So ... conceptually I'm trying to accomplish the same thing ...

Today we use FS a lot for termination of VoIP traffic ... all done via
XML_CURL ... which is awesome  (not to xml cdr ... and the "proxying" of
media) ...

Would like to do something like:
  -- originate request (looks simple enough)
  -- on answer XML_CURL posts info

But for the life of me I can't figure out how to translate this into the xml
response ...

[campaign]
exten => 100,1,ANSWER()
exten => 100,n,WAIT(2)
exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile)
exten => 100,n,WAITEXTEN(10)
exten => 100,n,HANGUP()

exten => 1,1,PLAYBACK(goodbye)
 and so on ...

I've looked at the ivr.conf stuff but it's all static and all of this has to
be manageable via a web interface  meaning dumping into a DB and
returning an XML response seems reasonable ... but trying to stick or modify
static text files from the web interface is too much text parsing and bad
things will happen ...

Any thoughts or pointing me in the right direction would be appreciated.

Shelby
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[Freeswitch-users] Auto dialing ...

[Shelby Ramsey]

OK ... Here goes another I'm doing this with AST  ... but I want to move it
to FS.  Searched via google site:lists.freeswitch.org auto dialer and others
... nothing useful.
Today I have a platform for auto dialing with AST (centrally managed ...
about 10 machines) and we do this:
  -- Remote machines query central DB for numbers to call based on certain
configs
  -- Use AMI to generate the call
  -- If call gets answered, extension info queried via rta (central db
again)

The nice thing about all of this is it's relatively easy to manage (through
one central web interface we built) and it works ... the bad part is
reporting ... as anyone knows on this list that has used AST for auto
dialing in this way (via .call or AMI) every call looks like it fails
instead of showing a real cause code.

So ... conceptually I'm trying to accomplish the same thing ...

Today we use FS a lot for termination of VoIP traffic ... all done via
XML_CURL ... which is awesome!

Would like to do something like:
  -- originate request
  -- on answer XML_CURL posts info
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Re: [Freeswitch-users] SBC configuration

[Shelby Ramsey]

Hey Ashley,
When you say "SBC" what are you trying to accomplish specifically?

FS should be able to accomplish what you want ... but can you provide some
more details as far as what you want to accomplish and I'm sure someone
 here will be glad to help.

SR

On Mon, Jan 12, 2009 at 2:02 AM, Ashley van Gerven wrote:

> Hi,
>
> I am interested in testing Freeswitch acting as an SBC. Is it simply a
> matter of configuring the dialplan correctly, using RE's so that inbound
> calls are just forwarded to our internal PBX and outbound calls from the PBX
> are forwarded to the VOIP provider?
>
> Or do I need to create an application that specifcally creates a new call
> and then joins the inbound and outbound calls?
>
> I haven't been able to find info on the wiki or google re. SBC setup.
>
>
> thanks
> Ash
>
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Re: [Freeswitch-users] Redirecting a call from one FS to another FS?

[Shelby Ramsey]

I agree with Ray ... using a 3XX series message is a bad idea ... or you
could put OpenSer in front using the LCR module ... 503 to OpenSer and it
would route to the next gateway in the gateway group.
I have yet to work with any carrier that handles 3XX series correctly except
for some of the TCAP guys.

On Thu, Dec 18, 2008 at 11:58 AM, Raymond Chandler <
intralan...@freeswitch.org> wrote:

> Gabriel Kuri wrote:
> > I've tried to do the same and in my own experience, most carriers don't
> > accept 302 redirects. What I've seen is they take the 302 as a failure
> > and move on to the next switch, so worse case with 3 switches, it will
> > take 2 retries before hitting the switch you want them to redirect to.
> >
> >
>
> could also just respond with a 503 in which case all carriers should
> fail over to the next one...
>
> -Ray
>
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Re: [Freeswitch-users] how to handle returned sip 302 dialplan

[Shelby Ramsey]

Chav,
We recently / are still going through the same process (in order to route on
LRN) vs NPANXX or LATA based routing.  Here was the best way that we came up
with to do it:
  -- we use xml_curl exclusively for routing decisions
  -- so in the cgi script that xml_curl hits one of the things it can (and
does based on certain parameters) is fire off a url to another LNP server
that we built
  -- the LNP server actually does the dip (either from a cache) and returns
the info

We felt this was much better for a few reasons:
  -- caching the LNP data for a 24 hour period would save us in excess of
$100k a year
  -- having a specialized mechanism to do this was much easier to implement
for the cgi process than supporting 302 redirects directly on the FS boxes
was much easier (which just wasn't possible with the cgi mechanism)
  -- every LNP provider returns 302's slightly different ... so we didn't
want to have to reinvent the wheel on the FS machines if we ever wanted to
add redundancy or switch providers

Guess it all depends on your config ... but this was the easiest and most
cost-effective means for us to implement.


On Sat, Dec 13, 2008 at 6:38 PM, Chav Paskov  wrote:

> Brian West wrote:
> > Chav,
> > Once the 302 is received by FreeSWITCH it will follow it to the
> > contact listed in the 302.  What else are you needing to do?
> >
> > /b
> >
> > On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
> >
> >> *User-Agent: eXosip/3.1.0^M
> >> Content-Length:
> >>
> >>
> >> my question is:
> >>
> >> Is it possible to send the call to z.z.z.z , receive the SIP 302 ,
> >> process the data in Contact field  and redirect to the new destination
> >> contained in *Contact: 
> >> 
> >;npdi^M
> >> *without closing the session.
> >> i red something about>> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
> >>
> >> Any ideas on this matter  will be highly appreciated.
> >> Best Regards
> >> Chav
> >>
> >
> > 
> >
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> >
> Thanks Brian,
>
> probably i should have explained it in more details.
> this whole thing started as an attempt to  implement lata ocn /local
> number portability/ instead of  pure per destination routing.
> At the moment i have a access to  a service provider  who does
> "dipping"  and returns  the  LATA OCN  data associated with  any dialed
> destination number. it is returned as Contact:  and Content-length:
> fields in 302 message.
>
> in other words:
>
> 1. i'm sending to this provider let say - 202555  as a destination
> number.
> 2. they do the dipping  and will return to me  either the new dest #  if
> 202555 has been ported or if it was not
> in content-length field they'll send  lata, ocn  and state and 10 digits
> number.
> 3. once received i have to  compare the received lata , ocn and state
> date with  a compiled  rate deck / blended from 5 different vendors/
> and pick the lowest rate - effectively building  LCR based on LATA OCN
> STATE info.
>
> Hope this will help to clear the picture.
> Regards
> Chav
>
>
>
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Re: [Freeswitch-users] CDR logs - adding a custom field

[Shelby Ramsey]

Are there any good examples floating around of XML parsers for this to dump
to MySQL?

On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins  wrote:

> On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu 
> wrote:
> > Thanks Michael,
> >
> > I'm going to use XML, since I don't really know what variables I want.
> > Another problem with CSV is that many people parse them with regular
> > expressions and scripts break when you add a new column.
> >
>
> This is true. If you build a proper parser for your XML it will easily
> be able to handle new channel variables.
> -MC
>
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Re: [Freeswitch-users] Proto specific hangup cause issue

[Shelby Ramsey]

Hello,
This is just my 2 cents ... but my experience has been that trying to catch
all of the various variables (i.e. from XML_CDR) or otherwise can be a
little trying (a row in your CDR database could be over 100 fields long!).

The best option here is to catch the UUID's for the 2 call legs, capture all
SIP messaging, parse and dump the messaging, and then correlate the calls
from the CDR from there.

Much easier than trying to do it from FS ... and most folks want to see SIP
captures anyway (very broad set of tools to debug).

Measuring things like ASR, PDD, etc in my opinion is much easier from the
raw messaging than trying to do something with FS CDR records.



On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <[EMAIL 
PROTECTED]>wrote:

>
> We are currently in the migration process from our
> current system to a FS based setup. We are in the process of
> adapting our billing and routing to FS. All the  CDRs (and variables)
> related issues that we have been discussing on this mailing list
> come from the need to extract the same level of information from FS as
> we do with our current closed source proprietary system. So, we
> chose FS because of the versatility it provides in every aspect (event
> handling, config implementation etc.) and we strongly believe that all
> these additions/fixes would be beneficial to many potential FS users.
>
> We are at your disposal for more details in case you need
> more information about what exactly we are trying to do. Basically,
> our approach is from the VoIP carrier's point of view rather than the
> PBX user's/implementor's. So, the details that we asked to be introduced
> to FS come from real life issues that we have faced during the last few
> years
> with various platforms and troubleshooting experiences with other VoIP
> carriers.
>
>
>
>
> Michael Collins wrote:
>
> Thanks for your feedback. It definitely helps to know not only what
> you need FS to do but why you need it to do so.
>
> Do you have FS in production right now? Just curious.
>
> Thanks,
> MC
>
> On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos<[EMAIL PROTECTED]> 
> <[EMAIL PROTECTED]> wrote:
>
>
>  "I already added 2 patches for you right.  Just be clear about what you
> want."
>
> And I am grateful of that.
>
> "it is protocol neutral, that's why it starts with sip_"
>
> I didn't know that. I thought that the sip_ variables are protocol specific.
> So one would expect there to be an iax_hangup_disposition,
> woomera_hangup_disposition etc?
>
> "Maybe you should beat around the bush less with your "requirements" for
> your application you are expecting me to support for you."
>
> I am just trying to gather statistics for my providers as I would with any
> VoIP softswitch. (hangup causes per terminator per destination)
> I don't think that this is a specific "application" rather than a general
> necessity for VoIP carriers. It is also very useful for troubleshooting
> purposes : when I look at my CDRs to find a call that I got a complain for,
> I want to be able to tell if it was me or the provider who
> hanged up and gave a specific hangup cause, so that I can troubleshoot the
> issue better.
>
> "Just be clear about what you want."
>
> I want FS to reach that level of detailing and maturity in all aspects so
> that it could be the softswitch of choice by any VoIP entrepreneur
> (or hobbyist) and it is my strong belief that this can only be done by the
> community giving feedback to the programmers about what
> they find useful or not (i.e. experience from real-life situations). The
> patches that you made the last few days were not intended for
> me exclusively but for anyone that will face the same situations using FS.
> If you want the community to stop sending feedback about
> features/improvements you may as well close down this mailing list or just
> use it as an announcement board.
>
> I wish I was a c programmer and get involved with the project actively. But
> I am not. And as far as I can tell most of the registered users
> in this list aren't either. So they only way we can help is by testing and
> suggesting.
>
> Anthony Minessale wrote:
>
> it is protocol neutral, that's why it starts with sip_
>
> the variable can be any of:
>
> send_bye
> recv_bye
> send_cancel
> send_refuse
>
>
> using that value you can determine the information you asked.  I answered
> your specific question which was:
> determining "which side hanged up".  Maybe you should beat around the bush
> less with your "requirements" for your application you are expecting me to
> support for you.
>
> I already added 2 patches for you right.  Just be clear about what you want.
>
>
>
> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]> 
> <[EMAIL PROTECTED]>
> wrote:
>
>
>  Not necessarily. For instance I got a "send_cancel" when the
> calling party hanged up before the other party could pick up.
> Also, shouldn't something like that be protocol/technology
> neutral?
>

Re: [Freeswitch-users] Question on CDR CSV

[Shelby Ramsey]

Thanks Brian!

On Sat, Nov 15, 2008 at 1:33 PM, Brian West <[EMAIL PROTECTED]> wrote:

> Yes its correct.  we log to the account code specific file and the
> master csv file.
>
> /b
>
> On Nov 15, 2008, at 1:26 PM, Shelby Ramsey wrote:
>
> > Is this correct?  Just want to make sure I don't miss or duplicate
> > calls in the CDR records ...
>
>
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[Freeswitch-users] Question on CDR CSV

[Shelby Ramsey]

Hello,
Looking at the cdr-csv files ... it would appear that the call gets logged
to 2 files:
  -- an accountcode.csv file
  -- and the master.csv file

Is this correct?  Just want to make sure I don't miss or duplicate calls in
the CDR records ...

Thanks!

SR
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Re: [Freeswitch-users] 487's ...

[Shelby Ramsey]

Thanks Anthony!  That seems to have done the trick.
SR

On Tue, Nov 11, 2008 at 8:10 AM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:

> you can work around this issue by disabling 100rel
> 
> in the sip profile.
>
> We have an issue open with sofia to correct this?
>
>
> On Tue, Nov 11, 2008 at 2:29 AM, Shelby Ramsey <[EMAIL PROTECTED]>wrote:
>
>> Mike,
>> Looks like upgrading to trunk ... let me know if you have any thoughts.
>>
>> Here is the trace:
>>
>> U 4.71.122.167:5060 -> 4.71.122.130:5060
>> SIP/2.0 183 Session Progress.
>> Via: SIP/2.0/UDP 4.71.122.130;branch=z9hG4bK340a.42319c57.0.
>> Via: SIP/2.0/UDP 4.71.122.250:5060;branch=z9hG4bK7552b82e;rport=5060.
>> Record-Route: .
>> From: "8885551000" 
>> >;tag=as25098621.
>> To: 
>> >;tag=gjK5v6DmZcjjK.
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 INVITE.
>> Contact: .
>> User-Agent: sicdevdal0002.sipinterchange.com.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: 100rel, timer, precondition, path, replaces.
>> Allow-Events: talk.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 243.
>> .
>> v=0.
>> o=Quintum 8080133184107754731 3681448569772521226 IN IP4 4.71.122.167.
>> s=VoipCall.
>> c=IN IP4 4.71.122.167.
>> t=0 0.
>> m=audio 15446 RTP/AVP 0 101.
>> c=IN IP4 4.71.122.167.
>> a=rtpmap:0 pcmu/8000/1.
>> a=rtpmap:101 telephone-event/8000/1.
>> a=ptime:20.
>>
>>
>> U 4.71.122.131:5060 -> 4.71.122.167:5060
>> SIP/2.0 200 OK.
>> Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
>> Content-Length: 0.
>> CSeq: 107077572 PRACK.
>> From: "NULL"
>> >;tag=HUcyy1yQvN84e.
>> Record-Route: .
>> To: ;tag=26651084-1b896f.
>> Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKH0y57Q25QyytH.
>> .
>>
>>
>> U 4.71.122.130:5060 -> 4.71.122.167:5060
>> CANCEL sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
>> Record-Route: .
>> Via: SIP/2.0/UDP 4.71.122.130;branch=z9hG4bK340a.42319c57.0.
>> Via: SIP/2.0/UDP 4.71.122.250:5060;branch=z9hG4bK7552b82e;rport=5060.
>> From: "8885551000" 
>> >;tag=as25098621.
>> To: >.
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 CANCEL.
>> User-Agent: SIPINTERCHANGE.COM.
>> Max-Forwards: 69.
>> Content-Length: 0.
>> X-Proc: LCR applied.
>> .
>>
>>
>> U 4.71.122.167:5060 -> 4.71.122.130:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 4.71.122.130;branch=z9hG4bK340a.42319c57.0.
>> Via: SIP/2.0/UDP 4.71.122.250:5060;branch=z9hG4bK7552b82e;rport=5060.
>> Record-Route: .
>> From: "8885551000" 
>> >;tag=as25098621.
>> To: 
>> >;tag=gjK5v6DmZcjjK.
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 CANCEL.
>> Content-Length: 0.
>> .
>>
>>
>> U 4.71.122.167:5060 -> 4.71.122.130:5060
>> SIP/2.0 487 Request Terminated.
>> Via: SIP/2.0/UDP 4.71.122.130;branch=z9hG4bK340a.42319c57.0.
>> Via: SIP/2.0/UDP 4.71.122.250:5060;branch=z9hG4bK7552b82e;rport=5060.
>> From: "8885551000" 
>> >;tag=as25098621.
>> To: 
>> >;tag=gjK5v6DmZcjjK.
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 INVITE.
>> User-Agent: sicdevdal0002.sipinterchange.com.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: 100rel, timer, precondition, path, replaces.
>> Allow-Events: talk.
>> Content-Length: 0.
>> .
>>
>>
>> U 4.71.122.130:5060 -> 4.71.122.167:5060
>> ACK sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP 4.71.122.130;branch=z9hG4bK340a.42319c57.0.
>> From: "8885551000" 
>> >;tag=as25098621.
>> Call-ID: [EMAIL PROTECTED]
>> To: 
>> >;tag=gjK5v6DmZcjjK.
>> CSeq: 102 ACK.
>> SICPROXY0002.
>> Content-Length: 0.
>> .
>>
>>
>> U 4.71.122.167:5060 -> 4.71.122.131:5060
>> CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 4.71.122.167;rport;branch=z9hG4bKgQ5c6vH2tN87N.
>> Max-Forwards: 68.
>> From: "NULL" 
>> >;tag=HUcyy1yQvN84e.
>> To: .
>> Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
>> CSeq: 107077571 CANCEL.
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>> Content-Length: 0.
>> .
>>
>>
>> U 

Re: [Freeswitch-users] 487's ...

[Shelby Ramsey]

60;branch=z9hG4bKgQ5c6vH2tN87N.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077571 INVITE.
From: "NULL"
>;tag=HUcyy1yQvN84e.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKgQ5c6vH2tN87N.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: db4848c8-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077567 INVITE.
From: "NULL"
>;tag=F9ScUBXg23UZQ.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKe5jU26FU03U2e.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077571 INVITE.
From: "NULL"
>;tag=HUcyy1yQvN84e.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKgQ5c6vH2tN87N.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: db4848c8-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077567 INVITE.
From: "NULL"
>;tag=F9ScUBXg23UZQ.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKe5jU26FU03U2e.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077571 INVITE.
From: "NULL"
>;tag=HUcyy1yQvN84e.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKgQ5c6vH2tN87N.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: db4848c8-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077567 INVITE.
From: "NULL"
>;tag=F9ScUBXg23UZQ.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKe5jU26FU03U2e.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077571 INVITE.
From: "NULL"
>;tag=HUcyy1yQvN84e.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKgQ5c6vH2tN87N.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: db4848c8-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077567 INVITE.
From: "NULL"
>;tag=F9ScUBXg23UZQ.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKe5jU26FU03U2e.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077571 INVITE.
From: "NULL"
>;tag=HUcyy1yQvN84e.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKgQ5c6vH2tN87N.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: db4848c8-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077567 INVITE.
From: "NULL"
>;tag=F9ScUBXg23UZQ.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKe5jU26FU03U2e.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077571 INVITE.
From: "NULL"
>;tag=HUcyy1yQvN84e.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKgQ5c6vH2tN87N.
.


U 4.71.122.131:5060 -> 4.71.122.167:5060
SIP/2.0 487 Request Terminated.
Call-ID: e01ea049-2a6c-122c-58a6-001e680489ed.
Content-Length: 0.
CSeq: 107077571 INVITE.
From: "NULL"
>;tag=HUcyy1yQvN84e.
Record-Route: .
To: .
Via: SIP/2.0/UDP 4.71.122.167;rport=5060;branch=z9hG4bKgQ5c6vH2tN87N.

Thanks for the help!

SR


On Mon, Nov 10, 2008 at 10:59 PM, Shelby Ramsey <[EMAIL PROTECTED]> wrote:

> Mike,
> It is 1.01 ... will upgrade to trunk and let you know.
>
> Thanks for the quick response.
>
> SR
>
> On Mon, Nov 10, 2008 at 7:50 PM, Michael Jerris <[EMAIL PROTECTED]> wrote:
>
>> Is this with current svn trunk?  I think we have fixed this issue now
>> in trunk, could you please test and let us know.
>>
>> Mike
>>
>> On Nov  10, 2008, at 7:54 PM, "Shelby Ramsey" <[EMAIL PROTECTED]>
>> wrote:
>>
>> > Hello FS'ers,
>> >
>> > I'm having an issue where I get a call and send it via a profile ...
>> > but if the "b" leg cancels the request (a 487) then fs does not
>> > respond resulting in lots of 487's coming back at me ...
>> >
>> > Of course I'm not sure why I'm getting 487's from the b leg side ...
>> >
>> > SR
>> > ___
>> > Freeswitch-users mailing list
>> > Freeswitch-users@lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/

Re: [Freeswitch-users] 487's ...

[Shelby Ramsey]

Mike,
It is 1.01 ... will upgrade to trunk and let you know.

Thanks for the quick response.

SR

On Mon, Nov 10, 2008 at 7:50 PM, Michael Jerris <[EMAIL PROTECTED]> wrote:

> Is this with current svn trunk?  I think we have fixed this issue now
> in trunk, could you please test and let us know.
>
> Mike
>
> On Nov  10, 2008, at 7:54 PM, "Shelby Ramsey" <[EMAIL PROTECTED]>
> wrote:
>
> > Hello FS'ers,
> >
> > I'm having an issue where I get a call and send it via a profile ...
> > but if the "b" leg cancels the request (a 487) then fs does not
> > respond resulting in lots of 487's coming back at me ...
> >
> > Of course I'm not sure why I'm getting 487's from the b leg side ...
> >
> > SR
> > ___
> > Freeswitch-users mailing list
> > Freeswitch-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> ___
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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[Freeswitch-users] RPID ...

[Shelby Ramsey]

Hello,
Any way to disable sending / using RPID in the dialplan and not at the SIP
profile level?

Thanks!

SR
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[Freeswitch-users] 487's ...

[Shelby Ramsey]

Hello FS'ers,
I'm having an issue where I get a call and send it via a profile ... but if
the "b" leg cancels the request (a 487) then fs does not respond resulting
in lots of 487's coming back at me ...

Of course I'm not sure why I'm getting 487's from the b leg side ...

SR
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Re: [Freeswitch-users] xml_curl ...

[Shelby Ramsey]

OK ... I added continue_on_fail=true and this fixed the issue.  Works with
hangup_after_bridge true or false.
SR

Thanks for the assi

On Fri, Nov 7, 2008 at 3:46 PM, Shelby Ramsey <[EMAIL PROTECTED]> wrote:

> Mike,
> Thanks for the info on the .cgi ...
>
> I altered the hangup_after_bridge ... see XML below:
>
> 
> 
>   
> 
>   
> 
>   
>   
>   
>   
>   
>/>
>/>
>   
>   
>   
>/>
>data="origination_caller_id_number=+17133000522" />
>   
>data="effective_caller_id_number=+17133000522" />
>   
>   
> 
>   
> 
>   
> 
>
> I had it set to true based on an example on the wiki (dialplan, example 7):
>
> 
>   
> 
>  />
>  />
>   
> 
>
>
> I had previously tried the hangup_after_bridge=false ...
>
> Any thoughts?
>
> Thanks for all the help!
>
> SR
>
>
> On Fri, Nov 7, 2008 at 1:22 PM, Michael Jerris <[EMAIL PROTECTED]> wrote:
>
>> 
>> change to
>>
>> 
>>
>> also, if your serving up from xml_curl, you can do the conditions on your
>> cgi and just have a blank condition tag, no reason to have the switch do the
>> regex as well.
>>
>> Mike
>>
>> On Nov 7, 2008, at 1:35 PM, Shelby Ramsey wrote:
>>
>> Hello,
>> I have a question re: xml_curl ... if I reply with this (from the /tmp/
>> file created by fs after xml_curl debug_on):
>>
>> 
>> 
>>   
>> 
>>   
>> 
>>   
>>   
>>   
>>   
>>   
>>   > />
>>   > data="origination_caller_id_number=7135454263" />
>>   > />
>>   > data="effective_caller_id_number=7133000522" />
>>   
>>   
>>   
>> 
>>   
>> 
>>   
>> 
>>
>> FS sends call to sofia/internal/[EMAIL PROTECTED] (which returns a
>> 404) but does not send to the next "action".
>>
>> Somewhat of a FS newbie but I thought this should work ... looking at the
>> log files I can see this:
>>
>> 2008-11-07 11:45:00 [DEBUG] switch_core_state_machine.c:140
>> switch_core_standard_on_execute() sofia/external/[EMAIL PROTECTED] 
>> bridge(sofia/internal/
>> [EMAIL PROTECTED])
>>
>> but it does not do the same thing to the next "action" -- > application="bridge" data="sofia/internal/[EMAIL PROTECTED]" />
>>
>>
>> ___
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>> http://www.freeswitch.org
>>
>>
>
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Re: [Freeswitch-users] xml_curl ...

[Shelby Ramsey]

Mike,
Thanks for the info on the .cgi ...

I altered the hangup_after_bridge ... see XML below:



  

  

  
  
  
  
  
  
  
  
  
  
  
  
  
  
  
  

  

  


I had it set to true based on an example on the wiki (dialplan, example 7):


  



  



I had previously tried the hangup_after_bridge=false ...

Any thoughts?

Thanks for all the help!

SR


On Fri, Nov 7, 2008 at 1:22 PM, Michael Jerris <[EMAIL PROTECTED]> wrote:

> 
> change to
>
> 
>
> also, if your serving up from xml_curl, you can do the conditions on your
> cgi and just have a blank condition tag, no reason to have the switch do the
> regex as well.
>
> Mike
>
> On Nov 7, 2008, at 1:35 PM, Shelby Ramsey wrote:
>
> Hello,
> I have a question re: xml_curl ... if I reply with this (from the /tmp/
> file created by fs after xml_curl debug_on):
>
> 
> 
>   
> 
>   
> 
>   
>   
>   
>   
>   
>/>
>data="origination_caller_id_number=7135454263" />
>   
>data="effective_caller_id_number=7133000522" />
>   
>   
>   
> 
>   
> 
>   
> 
>
> FS sends call to sofia/internal/[EMAIL PROTECTED] (which returns a
> 404) but does not send to the next "action".
>
> Somewhat of a FS newbie but I thought this should work ... looking at the
> log files I can see this:
>
> 2008-11-07 11:45:00 [DEBUG] switch_core_state_machine.c:140
> switch_core_standard_on_execute() sofia/external/[EMAIL PROTECTED] 
> bridge(sofia/internal/
> [EMAIL PROTECTED])
>
> but it does not do the same thing to the next "action" --  application="bridge" data="sofia/internal/[EMAIL PROTECTED]" />
>
>
> ___
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> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
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[Freeswitch-users] xml_curl ...

[Shelby Ramsey]

Hello,
I have a question re: xml_curl ... if I reply with this (from the /tmp/ file
created by fs after xml_curl debug_on):



  

  

  
  
  
  
  
  
  
  
  
  
  
  

  

  


FS sends call to sofia/internal/[EMAIL PROTECTED] (which returns a
404) but does not send to the next "action".

Somewhat of a FS newbie but I thought this should work ... looking at the
log files I can see this:

2008-11-07 11:45:00 [DEBUG] switch_core_state_machine.c:140
switch_core_standard_on_execute()
sofia/external/[EMAIL PROTECTED] bridge(sofia/internal/
[EMAIL PROTECTED])

but it does not do the same thing to the next "action" -- 

Any thoughts?

Thanks for any help!

SR
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[Freeswitch-users] RPID and PAI questions ...

[Shelby Ramsey]

Hello,

Couple of quick questions:
-- How do I keep FS from sending RPID?  And strip the header? And  
likewise from respecting it?
-- How do I add a PAI header -- example here of what I thought would  
work --  ... but it doesn't seem  
to work (proceeding action is to dial a resource outside of the box
-- How do I strip PAI headers from incoming calls?

Thanks for any help!

Shelby

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