Re: [Freeswitch-users] Sipura Codec Problem

[Tihomir Culjaga]

just an off-topic question but it concenns mass provissioning ... does
anyone know if there is an open TR069 platform we can work on?

T.

On Wed, Nov 4, 2009 at 1:16 AM, Michael Collins  wrote:

>
>
> On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner <
> kristian.kielhof...@gmail.com> wrote:
>
>> It appears that Tony has already added an option (amazing) BUT you
>> should really be setup for central provisioning with an installed base
>> that large...  You'll eventually have issues that *NO* amount of
>> Tony/FreeSWITCH magic can fix.
>>
>> Kristian is correct. Listen to him because he's familiar with having lots
> and lots of units out in the field. The bandage Tony applied will eventually
> wear off. The long-term solution is to treat the malady and not the symptom.
> I'm certain that members of the FS community could point you toward some
> resources to assist with central provisioning.
>
> -MC
>
>
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Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones

[Jason White]

Peter J. Zandvoort  wrote:
> 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 most flexible and sophisticated tools tend to have this characteristic,
the best solution to which is a supportive community and good documentation.
FreeSWITCH has the community; the documentation is improving thanks to ongoing
efforts to extend, clarify and enhance the wiki.


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Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones

[Peter J. Zandvoort]

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 trace and put it in the pastebin. 
> (http://pastebin.freeswitch.org)
>
> You can also join the IRC channel #freeswitch on irc.freenode.net 
>  and get some real-time help. There are some 
> sharp folks in there, not the least of which are the three main 
> FreeSWITCH de

Re: [Freeswitch-users] Wiki typo

[Michael Collins]

On Tue, Nov 3, 2009 at 7:45 PM, Dmitry Gromov  wrote:

> Thanks, done - page has been corrected!
>
> On Tue, Nov 3, 2009 at 22:34, Brian West  wrote:
>
>> Yes you can login and edit the wiki yourself.
>>
>>
>>
> You know... I actually spent some time looking for login/create account
> link when I noticed this typo. No idea why I did not see it then :)
>
>
>
Thank you for not giving up! :) We appreciate it when the community helps
out. Nicely done.
-MC
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Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones

[mkitchin.pub...@gmail.com]

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 trace and put it in the pastebin. 
> (http://pastebin.freeswitch.org)
>
> You can also join the IRC channel #freeswitch on irc.freenode.net 
>  and get some real-time help. There are some 
> sharp folks in there, not the least of which are the three main 
> FreeSWITCH developers.
>
> -MC
Thank you. I think I did what you are looking for. I stopped FS and 
launched this command.
TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch
and captured all output to http://pastebin.freeswitch.org/10965
Does this tell you anything?
I'm definitely new to SIP and phone system admin in general. I have 
plenty of network and Linux experience. With that in mind, someone on 
this mailing list emailed me directly and said SipX would be a better 
fit for me. Is that blasphemy for me to even mention? I went through the 
documentation and the provisioning aspect and web interface do look 
tempting to a novice. I apologize if this is like trying to buy a chevy 
at a ford dealership. I'm looking to deploy about 150 handsets at a 
corporate office and then 10 to 12 handsets at 120 remote locations. We 
are moving from an old key system, so our current features are very 
limited. We just need a few ACD groups, call history, and the other 
general basics. I first found Asterisk and read about some of the 
shortcomings. FS looks like the most robust solution. I have no idea 
where SipX would fit in. The people here are obviously a very 
knowledgeable group and I would gladly accept any thoughts, comments, etc.





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Re: [Freeswitch-users] Wiki typo

[Dmitry Gromov]

Thanks, done - page has been corrected!

On Tue, Nov 3, 2009 at 22:34, Brian West  wrote:

> Yes you can login and edit the wiki yourself.
>
>
>
You know... I actually spent some time looking for login/create account link
when I noticed this typo. No idea why I did not see it then :)


-- 
DG
NJ
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Re: [Freeswitch-users] Wiki typo

[Brian West]

Yes you can login and edit the wiki yourself.

Thanks,
/b

On Nov 3, 2009, at 9:16 PM, Dmitry Gromov wrote:


Hi!

Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example
It lists sample sofia.conf.xml which has this parameter:


I think it should read inbound-bypass-media and not inbound-no- 
media...


I know, it says "outdated" but still, can be confusing.

Anyone here who can edit wiki and correct?

Thanks,
Dmitry

--
DG
NJ

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[Freeswitch-users] Wiki typo

[Dmitry Gromov]

Hi!

Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example
It lists sample sofia.conf.xml which has this parameter:




I think it should read inbound-*bypass*-media and not inbound-*no*-media...

I know, it says "outdated" but still, can be confusing.

Anyone here who can edit wiki and correct?

Thanks,
Dmitry

-- 
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NJ
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Re: [Freeswitch-users] Precompiled Windows Binaries

[Jeff Lenk]

Dave,

Carlos can probably be a better help here too but yes Freepbx v3 is a web
gui that is under heavy development - it probably is not yet ready for
production but looks very promising!

you can navigate to http://127.0.0.1/freepbx-v3/index.php/installer.html and
restart the installer for freepbx if you want to experiment with it.

The base FreeSWITCH installer does install and work well with windows and is
quite easy to learn and configure. Their is a lot to learn though :)

Regards,
Jeff


Dave Stevenson wrote:
> 
> Jeff,
> 
> thanks a lot for the reply. I was a little confused by the fact that the 
> "SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried 
> that I might lose something. As you say though, think that I'll cross my 
> fingers and try the updated release. I am running FreeSwitch on a test 
> machine at the moment until the target hardware arrives - hopefully 
> tomorrow, so I can afford to have a little play.
> 
> You mentioned FreePBX V3. I had been fumbling around trying to work out
> what 
> this is and from what I've read, it seems to provide a GUI Front End for 
> configuring FreeSwitch ?
> 
> I am guessing that while it has been installed with FreeSwitch, I then
> need 
> to run the FreePBX Installer to update the FreePBX/FreeSwitch
> configuration 
> on my hardware ?
> 
> 
> When I start FreeSwitch, it does not automatically load the WAMPServer.
> 
> When I start WAMPServer manually, and open up localhost (127.0.0.1) in a
> web 
> browser, I can see the WampServer logo and various tools such as phpinfo() 
> and phpmyadmin. FreePBX is there under Your Projects.
> 
> When I opened this up the first time, it appeared to want to install
> FreePBX 
> over FreeSwitch, I tried to abort this when it was going to overwrite some 
> FreeSwitch conf files and I thought I'd better not go on until I had a 
> better idea what was happening. I backed out of the FreePBX install and
> now 
> I can't get the FreePBX or phpmyadmin pages up again (missing files) so it 
> looks like I'm going to have to reinstall anyway.
> 
> So, for next time,am I right in thinking that I should proceed with
> running 
> the FreePBX install from the WAMPServer menu ?
> 
> regards
> Dave
> 
> 
> 
> - Original Message - 
> From: "Jeff Lenk" 
> To: 
> Sent: Tuesday, November 03, 2009 2:48 PM
> Subject: Re: [Freeswitch-users] Precompiled Windows Binaries
> 
> 
>>
>> Hi Dave,
>>
>> These are supported by "Carlos Talbot" . They also include Freepbx v3
>>
>> Just as you said freeswitch-1.0.4.exe is the tagged release and
>> freeswitch.exe is a newer svn snapshot.
>>
>> There should be no problems installing the new version allthough best to
>> just try and see!
>>
>> Not sure why the newest one is from October 7th.
>>
>> Jeff
>>
>>
>> Dave Stevenson wrote:
>>>
>>> Hi,
>>>
>>> I have read the Docs on the Wiki
>>> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries)
>>> but am still not sure of what the different Windows install files are.
>>> Currently, the Windows Installer directory contains :-
>>>
>>> LATEST_SVN_15106 - 6 Bytes
>>>
>>> freeswitch-1.0.4.exe - 42 Megabytes
>>>
>>> freeswitch.exe - 32 Megabytes
>>>
>>> I have installed the freeswitch-1.0.4.exe file which is dated 3rd
>>> September. The freeswitch.exe file is dated 7th October and think that
>>> it
>>> contains the minor updates since 3rd September ?
>>>
>>> Could someone who knows FreeSwitch under windows help me understand the
>>> two files please ?
>>>
>>> I chickened out of running the later exe in case it did something to the
>>> running install of FreeSwitch 1.0.4, is it safe to run the newer exe
>>> with
>>> the old one already installed ?
>>> What will it actually do ?
>>>
>>> regards
>>> Dave
>>> ___
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>>>
>>>
>>
>> -- 
>> View this message in context: 
>> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> ___
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>> 
> 
> 
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> 

-- 
View this message in context: 
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Sent from the freeswitch

Re: [Freeswitch-users] IP Phones with FreeSwitch

[Rupa Schomaker]

These phones work with FS, come by irc and you can talk to sekil about
his use of them.

In general, if you haven't invested in a bunch of phones, I'd recommend:

Polycom 330,450,550 -- pick your price point
snom - again, pick your price point

These are generally well supported over the rest.

On Tue, Nov 3, 2009 at 2:57 PM, 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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>



-- 
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Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones

[Michael Collins]

On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com <
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
trace and put it in the pastebin. (http://pastebin.freeswitch.org)

You can also join the IRC channel #freeswitch on irc.freenode.net and get
some real-time help. There are some sharp folks in there, not the least of
which are the three main FreeSWITCH developers.

-MC
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Re: [Freeswitch-users] Sipura Codec Problem

[Anthony Minessale]

Don't forget the one where there was a typo in the one for G722 so now we
are all required to emulate that typo by running a 16khz codec with 8khz
timestamps and sdp params.



On Tue, Nov 3, 2009 at 1:12 PM, Arsen Chaloyan  wrote:

> Actually, there were a few more misinterpretations in earlier software of
> Cisco Gateways, which RFC implementers had to address in RFC3551, strange
> ...
>
> RTP Payload Type 19 remains reserved because "some implementations" wrongly
> interpreted 13 decimal as 13 hexadecimal value.
> Another issue is G726 bit packing. Again "some implementations" used wrong
> bit packing and RFC3551 tried to partially resolve this conflict introducing
> new payload format named AAL2-G726 ...
>
> --
> *From:* Brian West 
> *To:* freeswitch-users@lists.freeswitch.org
> *Sent:* Tue, November 3, 2009 10:27:39 PM
> *Subject:* Re: [Freeswitch-users] Sipura Codec Problem
>
> Sounds like bad planning.  I would send out a memo to your users and
> have them fix it.  I have raised a bug multiple times with Cisco g729a
> is NOT valid.
>
> /b
>
> On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote:
>
> > Yes, that was my first option, but there many endpoints that I'm not
> > able to configure. Basically it's a broadband solution where I have
> > like 1000 endpoints that are out of my provisioning.
> >
> > Thanks,
> > M
>
>
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-- 
Anthony Minessale II

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Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch

[Tihomir Culjaga]

pity,the phone looks quite nice...

On Tue, Nov 3, 2009 at 9:18 PM, Chris Chen  wrote:

> I think you are most likely on the wrong track, 3COM phones are locked to
> either 3COM PBX or the special Asterisk edition locked-down by 3COM. You
> cannot make them work with either FreeSWITCH or any other open SIP server
> other than 3COM IP PBX systems.
> I learned this over one year ago by playing with 3COm 3102 phones myself.
>
> Chris
>
>
>
> On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson 
> wrote:
>
>>  Tihomir,
>>
>> thanks for the link, but actually, I had already found/downloaded/read and
>> almost understood that document !
>>
>> However, the options to log into the phone and configure the extension
>> number etc. do not appear on my phone.
>>
>> From reading another post on the web, I don't think that the phone has the
>> SIP software loaded until it is downloaded from the Server - I think that
>> there is a "special" version of Asterix for 3Com that does this, maybe the
>> same functionality does not exist in FreeSwitch ?
>>
>> Maybe I should have been clearer in the post below, but I think that this
>> is the root of the problem. I think that the 3Com phone is looking for
>> the Switch to download the SIP firmware to it and FreeSwitch does not seem
>> to do that.
>>
>> Given that you have pointed me in the direction of that document, are you
>> using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track,
>> but please let me know how you've made it work
>>
>> regards
>> Dave
>>
>>
>>
>>
>> - Original Message -
>>  *From:* Tihomir Culjaga 
>> *To:* freeswitch-users@lists.freeswitch.org
>> *Sent:* Tuesday, November 03, 2009 7:53 PM
>> *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with
>> FreeSwitch
>>
>> you might read this before you bigin :P
>>
>> http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf
>>
>>
>> T.
>>
>>
>> On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson > > wrote:
>>
>>>  Help please . . . .
>>>
>>> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ?
>>>
>>> I have got FreeSwitch up and running with the SoftPhone, but can't get a
>>> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP
>>> Address from DHCP and it can see the FreeSwitch server but I can't find
>>> anything in the phone to allow the extension & password to be configured.
>>> Can FreeSwitch send this data to the phone (and if so, which configuration
>>> files are involved) or must the phone be configured manually before it can
>>> talk to FreeSwitch ?
>>>
>>> Any help would be really appreciated as I'm pulling my hair out here !
>>>
>>> Regards
>>> Dave
>>>
>>> ___
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>>>
>>  --
>>
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[Freeswitch-users] Dial Plan Question

[Jerry Richards]

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] WARNING On Inbound Call Question

[Anthony Minessale]

can you try the same thing with the latest trunk or pre-release tarball.


On Tue, Nov 3, 2009 at 3:35 PM, Jerry Richards
wrote:

>
> I have my Freeswitch server with an installed Sangoma A101D card.  Most
> everything works okay, however, when I get an inbound call from the PSTN, I
> see the following warning show up in the log.  Additionally, the caller (on
> the PSTN) does not hear ringback, and if the call is not answered within
> about 12 seconds, the call ends (so it doesn't go to voice mail).  If I
> make
> a call from one internal phone to another, then it will go to voice mail
> after 30 seconds.
>
>
> Here are the two warnings:
>
> [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1]
> Rc=0 CSid=0 Seq=11
> [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to
> PROGRESS_MEDIA
>
>
> Here is the log of the warning upon an inbound call:
>
> freeswi...@teoproxy.greyhawk.tonecommander.com>
> freeswi...@teoproxy.greyhawk.tonecommander.com>
> freeswi...@teoproxy.greyhawk.tonecommander.com>
> freeswi...@teoproxy.greyhawk.tonecommander.com>
> freeswi...@teoproxy.greyhawk.tonecommander.com>
> freeswi...@teoproxy.greyhawk.tonecommander.com>
> freeswi...@teoproxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835
> [WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0
> Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176]
> 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on
> 1:1 from DOWN to RING
> 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING]
> 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig
> [START]
> 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms
> 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound
> channel OpenZAP/1:1/5384
> 2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel
> OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132]
> 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384)
> State Change CS_NEW -> CS_INIT
> 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/5384 [BREAK]
> 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/5384) Running State Change CS_INIT
> 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/5384) State INIT
> 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384)
> State Change CS_INIT -> CS_ROUTING
> 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
> OpenZAP/1:1/5384 [BREAK]
> 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
> (OpenZAP/1:1/5384) State INIT going to sleep
> 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
> (OpenZAP/1:1/5384) Running State Change CS_ROUTING
> 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484
> (OpenZAP/1:1/5384) State ROUTING
> 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384
> CHANNEL ROUTING
> 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78
> OpenZAP/1:1/5384 Standard ROUTING
> 2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing
> 4253813176->5384 in context default
> Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false
> Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~
> /^true$/ break=on-false
> Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~
> /^true$/ break=on-false
> Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true
> Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example]
> Dialplan: OpenZAP/1:1/5384 Action set(open=true)
> Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false
> Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI]
> destination_number(5384) =~ /^9(\d+)$/ break=on-false
> Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept]
> continue=false
> Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept]
> destination_number(5384) =~ /^(5380)$/ break=on-false
> Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept]
> continue=false
> Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept]
> destination_number(5384) =~ /^\*8$/ break=on-false
> Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false
> Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext]
> destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false
> Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384)
> =~
> /^870$/ break=on-false
> Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true
> Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~
> /^true$/ break=never
> Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~
> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: OpenZAP/1:1/5384 Absol

Re: [Freeswitch-users] IP Phones with FreeSwitch

[Dave Stevenson]

Thanks a lot - to both William and Shelby,

that makes me more confident about trying out at least one Cisco and Rupa 
has just given me a few more options, so, hopefully, I won't make the 3Com 
mistake again !


regards
Dave
- Original Message - 
From: "William Suffill" 
To: 
Sent: Tuesday, November 03, 2009 9:10 PM
Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch


> Cisco 7960 and the like that they push on the enterprise level for
> call manager also can be flashed with sip based firmware. I've only
> used the 7960 with the sip firmware.
>
>
> SPA942 and the like that used to be under Linksys/Sipura before that
> are targeted more toward smaller businesses and run SIP out of the box
> without any license complications.
>
> -- W
>
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Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch

[Tihomir Culjaga]

well, if it is a sip phone than you should be able to input your
username&password somewhere.
Usually, SIP phones downloads their configuration using dhcp/tftp|http
method... the FW is downloaded just once if you need to upgrade the phone...

I don't have any of these phones on my desk, just found the manual on the
web.

anyhow, freeswitch is expecting a SIP phone to register and thats it :P ...
there is no specific phone provisioning from FS side.


T.



On Tue, Nov 3, 2009 at 9:03 PM, Dave Stevenson wrote:

>  Tihomir,
>
> thanks for the link, but actually, I had already found/downloaded/read and
> almost understood that document !
>
> However, the options to log into the phone and configure the extension
> number etc. do not appear on my phone.
>
> From reading another post on the web, I don't think that the phone has the
> SIP software loaded until it is downloaded from the Server - I think that
> there is a "special" version of Asterix for 3Com that does this, maybe the
> same functionality does not exist in FreeSwitch ?
>
> Maybe I should have been clearer in the post below, but I think that this
> is the root of the problem. I think that the 3Com phone is looking for the
> Switch to download the SIP firmware to it and FreeSwitch does not seem to do
> that.
>
> Given that you have pointed me in the direction of that document, are you
> using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track,
> but please let me know how you've made it work
>
> regards
> Dave
>
>
>
>
> - Original Message -
> *From:* Tihomir Culjaga 
> *To:* freeswitch-users@lists.freeswitch.org
> *Sent:* Tuesday, November 03, 2009 7:53 PM
> *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with
> FreeSwitch
>
> you might read this before you bigin :P
>
> http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf
>
>
> T.
>
>
> On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson 
> wrote:
>
>>  Help please . . . .
>>
>> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ?
>>
>> I have got FreeSwitch up and running with the SoftPhone, but can't get a
>> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP
>> Address from DHCP and it can see the FreeSwitch server but I can't find
>> anything in the phone to allow the extension & password to be configured.
>> Can FreeSwitch send this data to the phone (and if so, which configuration
>> files are involved) or must the phone be configured manually before it can
>> talk to FreeSwitch ?
>>
>> Any help would be really appreciated as I'm pulling my hair out here !
>>
>> Regards
>> Dave
>>
>> ___
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>>
>  --
>
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Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch

[Chris Chen]

I think you are most likely on the wrong track, 3COM phones are locked to
either 3COM PBX or the special Asterisk edition locked-down by 3COM. You
cannot make them work with either FreeSWITCH or any other open SIP server
other than 3COM IP PBX systems.
I learned this over one year ago by playing with 3COm 3102 phones myself.

Chris


On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson wrote:

>  Tihomir,
>
> thanks for the link, but actually, I had already found/downloaded/read and
> almost understood that document !
>
> However, the options to log into the phone and configure the extension
> number etc. do not appear on my phone.
>
> From reading another post on the web, I don't think that the phone has the
> SIP software loaded until it is downloaded from the Server - I think that
> there is a "special" version of Asterix for 3Com that does this, maybe the
> same functionality does not exist in FreeSwitch ?
>
> Maybe I should have been clearer in the post below, but I think that this
> is the root of the problem. I think that the 3Com phone is looking for the
> Switch to download the SIP firmware to it and FreeSwitch does not seem to do
> that.
>
> Given that you have pointed me in the direction of that document, are you
> using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track,
> but please let me know how you've made it work
>
> regards
> Dave
>
>
>
>
> - Original Message -
> *From:* Tihomir Culjaga 
> *To:* freeswitch-users@lists.freeswitch.org
> *Sent:* Tuesday, November 03, 2009 7:53 PM
> *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with
> FreeSwitch
>
> you might read this before you bigin :P
>
> http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf
>
>
> T.
>
>
> On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson 
> wrote:
>
>>  Help please . . . .
>>
>> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ?
>>
>> I have got FreeSwitch up and running with the SoftPhone, but can't get a
>> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP
>> Address from DHCP and it can see the FreeSwitch server but I can't find
>> anything in the phone to allow the extension & password to be configured.
>> Can FreeSwitch send this data to the phone (and if so, which configuration
>> files are involved) or must the phone be configured manually before it can
>> talk to FreeSwitch ?
>>
>> Any help would be really appreciated as I'm pulling my hair out here !
>>
>> Regards
>> Dave
>>
>> ___
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>  --
>
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[Freeswitch-users] Error checking for PMP [general error]

[Jerry Richards]

When I start Freeswitch, I see an "Error checking for PMP [general error]"
as shown below.  Does anyone know what could cause this?


[r...@teoproxy bin]# ./freeswitch
Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch
-waste.
auto-adjusting stack size for optimal performance
2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing
Engine.
2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch
thread 0
2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT
2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5
2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5
2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5
2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5
2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5
2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP
[general error]
2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP
2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT
detected!
2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB
2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task
thread
2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1
heartbeat (core) to run at 1257185563
2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up
environment.
2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules.
2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530
definitions
2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [CORE_SOFTTIMER_MODULE]
2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding
Timer 'soft'
2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [CORE_PCM_MODULE]

Best Regards,
Jerry


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

[Anthony Minessale]

There are 2 ways to use the auto in

one is to attended transfer the call into the extension with auto in
the other is to bind_meta_app a call to valet_park + auto in

blind transfer to auto in only has one leg so the guy you transferred is the
only one who can hear it because when you press the blind xfer key you
hangup the call on your side.


On Tue, Nov 3, 2009 at 3:28 AM, Brian Stafford <
brian.staff...@lattice-voice.com> wrote:

> Brian Stafford wrote:
> > Brian West wrote:
> >
> >> You have to be doing it wrong then.
> >>
> >> Can you show us your dialplan you should have two extensions one for
> >> the lot range and one to attended transfer someone into the lot.
> >>
> >> /b
> >>
> >>
> > The relevant excerpt from the dialplan is
> >
> > 
> > 
> > 
> > 
> > 
> > 
> >
> > 
> > 
> > 
> > 
> > 
> > 
> >
> > x410-419 are the slots and 420 parks a call. Parking by picking one of
> > 410-419 works fine and subsequently dialling them from another works
> > fine, I added x420 for the auto feature.
> >
> > Regards
> > Brian
> >
> > _
>
> Any clues what I'm doing wrong?  Is more information needed?
>
> Brian
>
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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] no REINVITE on Blind Transfer with bypass_media

[Humberto Quintana]

Hi,

I tried r15332 and set in the sofia profile:

a) bypass_media_after_bridge=true only
b) bypass_media_after_bridge=true, param name="media-option" 
value="resume-media-on-hold"/>


In both cases FS is hanging up the initial call (A to FS) after accepting the 
REFER to C:

A <- reINVITE with FS' SDP <- FS
A -> 200 -> FS
A <- ACK <- FS
A <- BYE <- FS

The call to C is not even tried.

I found this line is the logs that could give some idea:

2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup 
sofia/external/514xxx...@a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]  
after sending the ACK for the reINVITE


Regards,


Humberto

>please try r15326
>I think i have it working.
>
>I recommend for optimal results you set bypass_media_after_bridge=true
>either as a global or in your DP in place of bypass_media=true
>
>
>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana hotmail.com>wrote:
>
>>  Hi Mike,
>>
>> I re-tried with trunk rev 15319 but I got almost the same behavior: There
>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted.  But
>> still there is no reINVITE for A (with C's SDP) after the call from FS to C
>> is established.
>>
>> Anyway, we decided for now to do a different implementation but if you want
>> to explore more in this issue count me in ;-)
>>
>>
>> Thank you very much!
>>
>> Humberto

  
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Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch

[Dave Stevenson]

Tihomir,

thanks for the link, but actually, I had already found/downloaded/read and 
almost understood that document !

However, the options to log into the phone and configure the extension number 
etc. do not appear on my phone.

>From reading another post on the web, I don't think that the phone has the SIP 
>software loaded until it is downloaded from the Server - I think that there is 
>a "special" version of Asterix for 3Com that does this, maybe the same 
>functionality does not exist in FreeSwitch ?

Maybe I should have been clearer in the post below, but I think that this is 
the root of the problem. I think that the 3Com phone is looking for the Switch 
to download the SIP firmware to it and FreeSwitch does not seem to do that. 

Given that you have pointed me in the direction of that document, are you using 
3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but 
please let me know how you've made it work

regards
Dave



  - Original Message - 
  From: Tihomir Culjaga 
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, November 03, 2009 7:53 PM
  Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch


  you might read this before you bigin :P

  http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf


  T.



  On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson  
wrote:

Help please . . . . 

Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ?

I have got FreeSwitch up and running with the SoftPhone, but can't get a 
3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP 
Address from DHCP and it can see the FreeSwitch server but I can't find 
anything in the phone to allow the extension & password to be configured. Can 
FreeSwitch send this data to the phone (and if so, which configuration files 
are involved) or must the phone be configured manually before it can talk to 
FreeSwitch ?

Any help would be really appreciated as I'm pulling my hair out here !

Regards
Dave

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[Freeswitch-users] Newbie trying to setup Cisco 7940 phones

[mkitchin.pub...@gmail.com]

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

dsi> sh conf
-- Current *FLASH* Configuration --

Platform : Cisco Systems, Inc. IP Phone CP-7940G
Elapsed Time: 01:01:06

dhcp_server : Disabled
my_ip_addr : 10.86.11.50
subnet_mask : 255.255.0.0
defaultgw : 10.86.0.1
dyn_dns_addr_1 : 0.0.0.0
dyn_dns_addr_2 : 0.0.0.0
dns_addr : 10.85.0.11
dns_backup_1: 10.85.0.10
primary_tftp_addr : 10.86.10.58
dyn_tftp_addr : 0.0.0.0
my_mac_addr : 0012:7f98:eaa9
domain_name : dsi-corp.net
my_name : SIP00127F98EAA9
Status Flags : 1231

image_version : "P003-8-12-00"
FirmLoadID : "PC030301"
DSPLoadID : "PS03AT38"
network_media_type : Auto
network_port2_type : Hub/Switch
dscpForAudio : 184
phone_label : "Matthew Kitchin"
tftp_cfg_dir : ""
phone_password : **
phone_prompt : "dsi"
language : english
sntp_mode : Unicast
sntp_server : 10.85.0.10
time_zone : CST
dst_offset : 01/00
dst_start_month : March
dst_start_day : 0
dst_start_day_of_week : Sunday
dst_start_week_of_month : 8
dst_start_time : 02/00
dst_stop_month : October
dst_stop_day : 0
dst_stop_day_of_week : Sunday
dst_stop_week_of_month : 8
dst_stop_time : 02/00
dst_auto_adjust : 1
time_format_24hr : 1
date_format : M/D/Y
nat_enable : 0
nat_address : UNPROVISIONED
voip_control_port : 5060
start_media_port : 16384
end_media_port : 32766
sync : "1"
xml_card_dir : ""
xml_card_file : "CARD.XML"
telnet_level : 1
services_url : ""
directory_url : ""
logo_url : ""
http_proxy_addr : UNPROVISIONED
http_proxy_port : 80
garp_enable : 0
enable_vad : 0
dial_template : "dialplan"
callerid_blocking : 0
anonymous_call_block : 0
autocomplete : 0
messages_uri : ""
dnd_control : 2
preferred_codec : g711ulaw
dtmf_outofband : avt_always
dtmf_avt_payload : 101
dtmf_db_level : 3
dtmf_inband : 0
call_manager1_addr : "UNPROVISIONED"
call_manager2_addr : "UNPROVISIONED"
call_manager3_addr : "UNPROVISIONED"
call_manager1_sip_port : 5060
call_manager2_sip_port : 5060
call_manager3_sip_port : 5060
call_manager5_addr : "UNPROVISIONED"
call_manager5_sip_port : 5060
call_manager4_addr : "UNPROVISIONED"
call_manager4_sip_port : 0
line1_name : "1008"
line2_name : "1001"
line1_authname : "1008"
line2_authname : "1001"
line1_password : **
line2_password : **
line1_shortname : "1008"
line2_shortname : "1001"
line1_displayname : "1008"
line2_displayname : "UNPROVISIONED"
line1_contact : "UNPROVISIONED"
line2_contact : "UNPROVISIONED"
proxy1_address : "nshplpbx1.unix"
proxy2_address : "nshplpbx1.unix"
proxy1_port : 5060
proxy2_port : 5060
sip_retx : 10
sip_invite_retx : 6
timer_t1 : 500
timer_t2 : 4000
timer_invite_expires : 180
timer_register_expires : 3600
proxy_register : 1
proxy_backup : ""
proxy_emergency : ""
proxy_backup_port : 5060
proxy_emergency_port : 5060
outbound_proxy : UNPROVISIONED
outbound_proxy_port : 5060
nat_received_processing : 0
mwi_status : 0
call_waiting : 1
user_info : phone
cnf_join_enable : 0
remote_party_id : 1
semi_attended_transfer : 1
transfer_onhook_enabled : 0
call_hold_ringback : 3
stutter_msg_waiting : 0
cfwd_url : ""
call_stats : 0
auto_answer : 0
local_cfwd_enable : 1
timer_register_delta : 5
sip_max_forwards : 70
rfc_2543_hold : 0
version_stamp : ""
timer_keepalive_expires : 120
connection_monitor_duration : 120
encrypt_key : **
2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5067 0 acls to check for proxy
2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5085 network ip is a proxy [0]
2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl 
"domains". Falling back to Digest auth.
2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5067 0 acls to check for proxy
2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5085 network ip is a proxy [0]
2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl 
"domains". Falling back to Digest auth.
2009-11-03 15:39:50.797901 [NOTICE] switch_channel.c:613 New Channel 
sofia/internal/1...@

Re: [Freeswitch-users] IP Phones with FreeSwitch

[William Suffill]

Cisco 7960 and the like that they push on the enterprise level for
call manager also can be flashed with sip based firmware. I've only
used the 7960 with the sip firmware.


SPA942 and the like that used to be under Linksys/Sipura before that
are targeted more toward smaller businesses and run SIP out of the box
without any license complications.

-- W

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Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch

[Dave Stevenson]

Chris,

thanks a lot for the response. It's not the answer that I wanted, but it is 
what I was coming round to thinking.

As much as I'm disappointed (particularly as I've just got the phone), but at 
least it's a definitive answer and I can avoid wasting any more time with it, 
so thanks again.

Oh well, off to try and find some open SIP phones that will actually work for 
me,


regards
Dave
  - Original Message - 
  From: Chris Chen 
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, November 03, 2009 8:18 PM
  Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch


  I think you are most likely on the wrong track, 3COM phones are locked to 
either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot 
make them work with either FreeSWITCH or any other open SIP server other than 
3COM IP PBX systems.
  I learned this over one year ago by playing with 3COm 3102 phones myself.

  Chris



  On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson  
wrote:

Tihomir,

thanks for the link, but actually, I had already found/downloaded/read and 
almost understood that document !

However, the options to log into the phone and configure the extension 
number etc. do not appear on my phone.

From reading another post on the web, I don't think that the phone has the 
SIP software loaded until it is downloaded from the Server - I think that there 
is a "special" version of Asterix for 3Com that does this, maybe the same 
functionality does not exist in FreeSwitch ?

Maybe I should have been clearer in the post below, but I think that this 
is the root of the problem. I think that the 3Com phone is looking for the 
Switch to download the SIP firmware to it and FreeSwitch does not seem to do 
that. 

Given that you have pointed me in the direction of that document, are you 
using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, 
but please let me know how you've made it work

regards
Dave



  - Original Message - 
  From: Tihomir Culjaga 
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, November 03, 2009 7:53 PM
  Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch


  you might read this before you bigin :P

  http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf


  T.



  On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson 
 wrote:

Help please . . . . 

Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ?

I have got FreeSwitch up and running with the SoftPhone, but can't get 
a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP 
Address from DHCP and it can see the FreeSwitch server but I can't find 
anything in the phone to allow the extension & password to be configured. Can 
FreeSwitch send this data to the phone (and if so, which configuration files 
are involved) or must the phone be configured manually before it can talk to 
FreeSwitch ?

Any help would be really appreciated as I'm pulling my hair out here !

Regards
Dave

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Re: [Freeswitch-users] Sipura Codec Problem

[Brian West]

At some point the paint will be rubbed off the magic lamp.

/b

On Nov 3, 2009, at 1:11 PM, Kristian Kielhofner wrote:

> It appears that Tony has already added an option (amazing) BUT you
> should really be setup for central provisioning with an installed base
> that large...  You'll eventually have issues that *NO* amount of
> Tony/FreeSWITCH magic can fix.


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Re: [Freeswitch-users] Sipura Codec Problem

[Brian West]

THE BLOODY MADNESS!!! I can only stop if people start saying 'NO'.  :)

/b

On Nov 3, 2009, at 1:21 PM, Anthony Minessale wrote:

> Don't forget the one where there was a typo in the one for G722 so  
> now we are all required to emulate that typo by running a 16khz  
> codec with 8khz timestamps and sdp params.


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

[Anthony Minessale]

I don't know what you are talking about anymore.

The scenario I had tested is when a call is bridged in bypass_media=true
bridge
and you blind transfer that call back to the dialplan

as soon as it hits the routing state it will resume media.


it has been confirmed to not work and confirmed to have been fixed several
time and if you are still having a problem you must have something blocking
some of your packets or something .

You have to understand that sip is a protocol and your description is
completely non-standard.
Perhaps you should get a console trace and attach it to a jira.  The trace
probably makes more sense to me.

sofia profile internal siptrace on
console loglevel debug

reproduce and attach the whole capture.



On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote:

>
> Hi,
>
> I tried r15332 and set in the sofia profile:
>
> a) bypass_media_after_bridge=true only
> b) bypass_media_after_bridge=true, param name="media-option"
> value="resume-media-on-hold"/>
> 
>
> In both cases FS is hanging up the initial call (A to FS) after accepting
> the REFER to C:
>
> A <- reINVITE with FS' SDP <- FS
> A -> 200 -> FS
> A <- ACK <- FS
> A <- BYE <- FS
>
> The call to C is not even tried.
>
> I found this line is the logs that could give some idea:
>
> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup
> sofia/external/514xxx...@a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]
> after sending the ACK for the reINVITE
>
>
> Regards,
>
>
> Humberto
>
> >please try r15326
> >I think i have it working.
> >
> >I recommend for optimal results you set bypass_media_after_bridge=true
> >either as a global or in your DP in place of bypass_media=true
> >
> >
> >On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana  hotmail.com>wrote:
> >
> >>  Hi Mike,
> >>
> >> I re-tried with trunk rev 15319 but I got almost the same behavior:
> There
> >> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted.
>  But
> >> still there is no reINVITE for A (with C's SDP) after the call from FS
> to C
> >> is established.
> >>
> >> Anyway, we decided for now to do a different implementation but if you
> want
> >> to explore more in this issue count me in ;-)
> >>
> >>
> >> Thank you very much!
> >>
> >> Humberto
>
>
> _
> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they
> e-mail you.
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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AIM: anthm
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IRC: irc.freenode.net #freeswitch

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sip:8...@conference.freeswitch.org 
iax:gu...@conference.freeswitch.org/888
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Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media

[Brian West]

Do you have ANY nat involved?

/b

On Nov 3, 2009, at 6:05 PM, Humberto Quintana wrote:

> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/ 
> external/514xxx...@a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]
> after sending the ACK for the reINVITE


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

[Ujjval Karihaloo]

Was that sarcasm or you really mean it?



-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, November 02, 2009 9:08 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator

you know I have heard this before... It seems to ONLY be AT&T

/b

On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote:

> Yes, I think I did. However here is what furthur testing revelas. If  
> I dial in from AT&T cell phone, I do not see any DTMF using Don's  
> IVR.xml.conf to call my conf app. But when I dial the same number  
> using a Verizon Cell, it works.
>
> When I dial a number that is provisioned to call the Conf App  
> directly from the public.xml dialplan...it works even with the same  
> AT&T cell phone...
>
> Strange behaviour


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Re: [Freeswitch-users] IP Phones with FreeSwitch

[Peder]

FYI, you can't do "presence" with the Cisco phones, so you can't see if
someone is on the phone.

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Dave
Stevenson
Sent: Tuesday, November 03, 2009 3:29 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch

Thanks a lot - to both William and Shelby,

that makes me more confident about trying out at least one Cisco and Rupa 
has just given me a few more options, so, hopefully, I won't make the 3Com 
mistake again !


regards
Dave
- Original Message - 
From: "William Suffill" 
To: 
Sent: Tuesday, November 03, 2009 9:10 PM
Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch


> Cisco 7960 and the like that they push on the enterprise level for
> call manager also can be flashed with sip based firmware. I've only
> used the 7960 with the sip firmware.
>
>
> SPA942 and the like that used to be under Linksys/Sipura before that
> are targeted more toward smaller businesses and run SIP out of the box
> without any license complications.
>
> -- W
>
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Re: [Freeswitch-users] IP Phones with FreeSwitch

[Dave Stevenson]

Rupa,

thanks a lot for the pointers - I'm just about to try to pick up some 
phones, so the tips are timely.

Actually, I have been trying the IRC thing today, but keep getting 
"connection refused", it's been a few years since I used IRC, but I think I 
have a Firewall problem that I'm working on.

Hopefully, I'll be there soon,

regards
Dave





- Original Message - 
From: "Rupa Schomaker" 
To: 
Sent: Tuesday, November 03, 2009 9:19 PM
Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch


> These phones work with FS, come by irc and you can talk to sekil about
> his use of them.
>
> In general, if you haven't invested in a bunch of phones, I'd recommend:
>
> Polycom 330,450,550 -- pick your price point
> snom - again, pick your price point
>
> These are generally well supported over the rest.
>
> On Tue, Nov 3, 2009 at 2:57 PM, 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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>>
>
>
>
> -- 
> -Rupa
>
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Re: [Freeswitch-users] Error checking for PMP [general error]

[Rupa Schomaker]

If you don't have a router with NAT-PMP enabled then it is expected.
Same if you don't have upnp.

If you are behind a NAT, it is in your best interest to enable one or
the other in your router. It will save you a bunch of headaches...

On Tue, Nov 3, 2009 at 3:25 PM, Jerry Richards
 wrote:
>
> When I start Freeswitch, I see an "Error checking for PMP [general error]"
> as shown below.  Does anyone know what could cause this?
>
>
> [r...@teoproxy bin]# ./freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch
> -waste.
> auto-adjusting stack size for optimal performance
> 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing
> Engine.
> 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch
> thread 0
> 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT
> 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5
> 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5
> 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5
> 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5
> 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5
> 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP
> [general error]
> 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP
> 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT
> detected!
> 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB
> 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task
> thread
> 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1
> heartbeat (core) to run at 1257185563
> 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up
> environment.
> 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules.
> 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530
> definitions
> 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889
> Successfully Loaded [CORE_SOFTTIMER_MODULE]
> 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding
> Timer 'soft'
> 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889
> Successfully Loaded [CORE_PCM_MODULE]
>
> Best Regards,
> Jerry
>
>
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-- 
-Rupa

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[Freeswitch-users] WARNING On Inbound Call Question

[Jerry Richards]

I have my Freeswitch server with an installed Sangoma A101D card.  Most
everything works okay, however, when I get an inbound call from the PSTN, I
see the following warning show up in the log.  Additionally, the caller (on
the PSTN) does not hear ringback, and if the call is not answered within
about 12 seconds, the call ends (so it doesn't go to voice mail).  If I make
a call from one internal phone to another, then it will go to voice mail
after 30 seconds.


Here are the two warnings:

[WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1]
Rc=0 CSid=0 Seq=11 
[WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to
PROGRESS_MEDIA


Here is the log of the warning upon an inbound call:

freeswi...@teoproxy.greyhawk.tonecommander.com> 
freeswi...@teoproxy.greyhawk.tonecommander.com> 
freeswi...@teoproxy.greyhawk.tonecommander.com> 
freeswi...@teoproxy.greyhawk.tonecommander.com> 
freeswi...@teoproxy.greyhawk.tonecommander.com> 
freeswi...@teoproxy.greyhawk.tonecommander.com> 
freeswi...@teoproxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835
[WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0
Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176]
2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on
1:1 from DOWN to RING
2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig
[START]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound
channel OpenZAP/1:1/5384
2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel
OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384)
State Change CS_NEW -> CS_INIT
2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
OpenZAP/1:1/5384 [BREAK]
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
(OpenZAP/1:1/5384) Running State Change CS_INIT
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
(OpenZAP/1:1/5384) State INIT
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384)
State Change CS_INIT -> CS_ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
OpenZAP/1:1/5384 [BREAK]
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
(OpenZAP/1:1/5384) State INIT going to sleep
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
(OpenZAP/1:1/5384) Running State Change CS_ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484
(OpenZAP/1:1/5384) State ROUTING
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384
CHANNEL ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78
OpenZAP/1:1/5384 Standard ROUTING
2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing
4253813176->5384 in context default
Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~
/^true$/ break=on-false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~
/^true$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true
Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example]
Dialplan: OpenZAP/1:1/5384 Action set(open=true)
Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI]
destination_number(5384) =~ /^9(\d+)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept]
continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept]
destination_number(5384) =~ /^(5380)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept]
destination_number(5384) =~ /^\*8$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext]
destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) =~
/^870$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~
/^true$/ break=never
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: OpenZAP/1:1/5384 Absolute Condition [global]
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe
r})
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-last_dial/gl

[Freeswitch-users] IP Phones with FreeSwitch

[Dave Stevenson]

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
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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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>   


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[Freeswitch-users] FS and Skinny (SCCP)

[mm_202]

FS doesnt support SCCP (from what I gathered, just because no one has
bothered coding it).

Are there other users out there has use SCCP and FS?  (with some
middleware in between)

If enough people would find a use for it, I'd be willing to actually
code it (esp if someone offered a bounty).
So, would anyone besides me want/use a SCCP endpoint in FS?

-- mm_202.

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Re: [Freeswitch-users] Sipura Codec Problem

[Brian West]

Yah this one is LLLAME :P

We have some dyslexic engineers.

/b

On Nov 3, 2009, at 1:12 PM, Arsen Chaloyan wrote:

> Another issue is G726 bit packing. Again "some implementations" used  
> wrong bit packing and RFC3551 tried to partially resolve this  
> conflict introducing new payload format named AAL2-G726 ...


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Re: [Freeswitch-users] Sipura Codec Problem

[Kristian Kielhofner]

It appears that Tony has already added an option (amazing) BUT you
should really be setup for central provisioning with an installed base
that large...  You'll eventually have issues that *NO* amount of
Tony/FreeSWITCH magic can fix.

On Tue, Nov 3, 2009 at 1:11 PM, Mariano de Llano
 wrote:
> Yes, that was my first option, but there many endpoints that I'm not
> able to configure. Basically it's a broadband solution where I have
> like 1000 endpoints that are out of my provisioning.
>
> Thanks,
> M
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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Re: [Freeswitch-users] Sipura Codec Problem

[Michael Collins]

On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> It appears that Tony has already added an option (amazing) BUT you
> should really be setup for central provisioning with an installed base
> that large...  You'll eventually have issues that *NO* amount of
> Tony/FreeSWITCH magic can fix.
>
> Kristian is correct. Listen to him because he's familiar with having lots
and lots of units out in the field. The bandage Tony applied will eventually
wear off. The long-term solution is to treat the malady and not the symptom.
I'm certain that members of the FS community could point you toward some
resources to assist with central provisioning.

-MC
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Re: [Freeswitch-users] Sipura Codec Problem

[Anthony Minessale]

I am willing to support this with the note that its incorrect and will not
support it by default but update to trunk and try:


this should fix it for you, SIGH


On Tue, Nov 3, 2009 at 12:27 PM, Brian West  wrote:

> Sounds like bad planning.  I would send out a memo to your users and
> have them fix it.  I have raised a bug multiple times with Cisco g729a
> is NOT valid.
>
> /b
>
> On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote:
>
> > Yes, that was my first option, but there many endpoints that I'm not
> > able to configure. Basically it's a broadband solution where I have
> > like 1000 endpoints that are out of my provisioning.
> >
> > Thanks,
> > M
>
>
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Re: [Freeswitch-users] portaudio error

[Andrew Thompson]

On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote:
> Hello
> 
> Debian lenny with svn15321
> 
> freeswi...@internal> load mod_portaudio
> -ERR [module load file routine returned an error]
> 
> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input 
> devicefreeswi...@internal> 2009-11-03 11:56:47.047969 [ERR] 
> mod_portaudio.c:974 Cannot find an input device
> 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading 
> module /opt/freeswitch/mod/mod_portaudio.so
> **Module load routine returned an error**
>
Try installing the alsa development headers, it's got some stupid name
on debian like libasound2-devel or something. Then re-build the
portaudio module and library (a couple well placed make cleans should do
it).

Andrew

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Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch

[Tihomir Culjaga]

you might read this before you bigin :P

http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf


T.


On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson wrote:

>  Help please . . . .
>
> Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ?
>
> I have got FreeSwitch up and running with the SoftPhone, but can't get a
> 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP
> Address from DHCP and it can see the FreeSwitch server but I can't find
> anything in the phone to allow the extension & password to be configured.
> Can FreeSwitch send this data to the phone (and if so, which configuration
> files are involved) or must the phone be configured manually before it can
> talk to FreeSwitch ?
>
> Any help would be really appreciated as I'm pulling my hair out here !
>
> Regards
> Dave
>
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Re: [Freeswitch-users] Sipura Codec Problem

[Anthony Minessale]

so imagine how much money all those sipuras cost.
They get all the money *and* have a bug and we are free and are supposed to
break the rules for them.


On Tue, Nov 3, 2009 at 12:11 PM, Mariano de Llano  wrote:

> Yes, that was my first option, but there many endpoints that I'm not
> able to configure. Basically it's a broadband solution where I have
> like 1000 endpoints that are out of my provisioning.
>
> Thanks,
> M
>
> On 03/11/2009, at 14:58, Brian West wrote:
>
> > FIx your sipura to NOT include the a in the codec its in the admin
> > section of the UI on the ATA.
> >
> > /b
> >
> > On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote:
> >
> >> Hi,
> >>
> >> I'm having a problem with a Sipura, it is sending for the G729  the
> >> tag "G729a" witch is not correct due the RFC.
> >>
> >> Media Attribute (a): rtpmap:18 G729a/8000
> >>
> >> FS is returning (200OK)
> >>
> >> Media Attribute (a): rtpmap:96 G729/8000
> >>
> >> I think that the problem is that FS is not matching the codec, so it
> >> returns the first dynamic payload which is 96.
> >>
> >> I think that I've seen post with a similar issue, and the solution
> >> was
> >> to change the tag before it hit the switch, so, what I've done is to
> >> change the "switch_r_sdp" (I have the rest of the parameters correct
> >> due I also use it to dynamically change the codecs order) and it's
> >> changing the SDP, but when FS sends the 200OK it is returning to the
> >> endpoint:
> >>
> >> Media Attribute (a): rtpmap:96 G729/8000
> >>
> >> Which is exactly the same problem that I have without the
> >> transformation of the SDP.
> >>
> >> Is it correct? Do I have another solution?
> >>
> >> Thanks
> >>
> >
> >
> > ___
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>
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Re: [Freeswitch-users] Sipura Codec Problem

[Brian West]

Sounds like bad planning.  I would send out a memo to your users and  
have them fix it.  I have raised a bug multiple times with Cisco g729a  
is NOT valid.

/b

On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote:

> Yes, that was my first option, but there many endpoints that I'm not
> able to configure. Basically it's a broadband solution where I have
> like 1000 endpoints that are out of my provisioning.
>
> Thanks,
> M


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Re: [Freeswitch-users] portaudio error

[Frank Carmickle]

On Tue, Nov 03, Andrew Thompson wrote:
> On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote:
> > Hello
> > 
> > Debian lenny with svn15321
> > 
> > freeswi...@internal> load mod_portaudio
> > -ERR [module load file routine returned an error]
> > 
> > 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input 
> > devicefreeswi...@internal> 2009-11-03 11:56:47.047969 [ERR] 
> > mod_portaudio.c:974 Cannot find an input device
> > 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error 
> > Loading module /opt/freeswitch/mod/mod_portaudio.so
> > **Module load routine returned an error**
> >
> Try installing the alsa development headers, it's got some stupid name
> on debian like libasound2-devel or something. Then re-build the
> portaudio module and library (a couple well placed make cleans should do
> it).

Hi

Libasound2-dev is still installed.  I have had PA working in the passed.  I 
think it was as of svn 14000 or so.  Thanks for the help.  

--FC


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[Freeswitch-users] Sipura Codec Problem

[Mariano de Llano]

Hi,

I'm having a problem with a Sipura, it is sending for the G729  the  
tag "G729a" witch is not correct due the RFC.

Media Attribute (a): rtpmap:18 G729a/8000

FS is returning (200OK)

Media Attribute (a): rtpmap:96 G729/8000

I think that the problem is that FS is not matching the codec, so it  
returns the first dynamic payload which is 96.

I think that I've seen post with a similar issue, and the solution was  
to change the tag before it hit the switch, so, what I've done is to  
change the "switch_r_sdp" (I have the rest of the parameters correct  
due I also use it to dynamically change the codecs order) and it's  
changing the SDP, but when FS sends the 200OK it is returning to the  
endpoint:

Media Attribute (a): rtpmap:96 G729/8000

Which is exactly the same problem that I have without the  
transformation of the SDP.

Is it correct? Do I have another solution?

Thanks


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Re: [Freeswitch-users] Sipura Codec Problem

[Arsen Chaloyan]

Actually, there were a few more misinterpretations in earlier software of Cisco 
Gateways, which RFC implementers had to address in RFC3551, strange ...

RTP Payload Type 19 remains reserved because "some implementations" wrongly 
interpreted 13 decimal as 13 hexadecimal value.
Another issue is G726 bit packing. Again "some implementations" used wrong bit 
packing and RFC3551 tried to partially resolve this conflict introducing new 
payload format named AAL2-G726 ...





From: Brian West 
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, November 3, 2009 10:27:39 PM
Subject: Re: [Freeswitch-users] Sipura Codec Problem

Sounds like bad planning.  I would send out a memo to your users and  
have them fix it.  I have raised a bug multiple times with Cisco g729a  
is NOT valid.

/b

On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote:

> Yes, that was my first option, but there many endpoints that I'm not
> able to configure. Basically it's a broadband solution where I have
> like 1000 endpoints that are out of my provisioning.
>
> Thanks,
> M


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Re: [Freeswitch-users] Sipura Codec Problem

[Mariano de Llano]

Yes, that was my first option, but there many endpoints that I'm not  
able to configure. Basically it's a broadband solution where I have  
like 1000 endpoints that are out of my provisioning.

Thanks,
M

On 03/11/2009, at 14:58, Brian West wrote:

> FIx your sipura to NOT include the a in the codec its in the admin
> section of the UI on the ATA.
>
> /b
>
> On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote:
>
>> Hi,
>>
>> I'm having a problem with a Sipura, it is sending for the G729  the
>> tag "G729a" witch is not correct due the RFC.
>>
>> Media Attribute (a): rtpmap:18 G729a/8000
>>
>> FS is returning (200OK)
>>
>> Media Attribute (a): rtpmap:96 G729/8000
>>
>> I think that the problem is that FS is not matching the codec, so it
>> returns the first dynamic payload which is 96.
>>
>> I think that I've seen post with a similar issue, and the solution  
>> was
>> to change the tag before it hit the switch, so, what I've done is to
>> change the "switch_r_sdp" (I have the rest of the parameters correct
>> due I also use it to dynamically change the codecs order) and it's
>> changing the SDP, but when FS sends the 200OK it is returning to the
>> endpoint:
>>
>> Media Attribute (a): rtpmap:96 G729/8000
>>
>> Which is exactly the same problem that I have without the
>> transformation of the SDP.
>>
>> Is it correct? Do I have another solution?
>>
>> Thanks
>>
>
>
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Re: [Freeswitch-users] Sipura Codec Problem

[Brian West]

FIx your sipura to NOT include the a in the codec its in the admin  
section of the UI on the ATA.

/b

On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote:

> Hi,
>
> I'm having a problem with a Sipura, it is sending for the G729  the
> tag "G729a" witch is not correct due the RFC.
>
> Media Attribute (a): rtpmap:18 G729a/8000
>
> FS is returning (200OK)
>
> Media Attribute (a): rtpmap:96 G729/8000
>
> I think that the problem is that FS is not matching the codec, so it
> returns the first dynamic payload which is 96.
>
> I think that I've seen post with a similar issue, and the solution was
> to change the tag before it hit the switch, so, what I've done is to
> change the "switch_r_sdp" (I have the rest of the parameters correct
> due I also use it to dynamically change the codecs order) and it's
> changing the SDP, but when FS sends the 200OK it is returning to the
> endpoint:
>
> Media Attribute (a): rtpmap:96 G729/8000
>
> Which is exactly the same problem that I have without the
> transformation of the SDP.
>
> Is it correct? Do I have another solution?
>
> Thanks
>


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[Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch

[Dave Stevenson]

Help please . . . . 

Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ?

I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com 
hardware phone to talk to FreeSwitch. I have the phone getting its IP Address 
from DHCP and it can see the FreeSwitch server but I can't find anything in the 
phone to allow the extension & password to be configured. Can FreeSwitch send 
this data to the phone (and if so, which configuration files are involved) or 
must the phone be configured manually before it can talk to FreeSwitch ?

Any help would be really appreciated as I'm pulling my hair out here !

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Re: [Freeswitch-users] Sipura Codec Problem

[Anthony Minessale]

I think you can edit the prefs in your sipura and change it to the correct
string.


On Tue, Nov 3, 2009 at 11:47 AM, Mariano de Llano  wrote:

> Hi,
>
> I'm having a problem with a Sipura, it is sending for the G729  the
> tag "G729a" witch is not correct due the RFC.
>
> Media Attribute (a): rtpmap:18 G729a/8000
>
> FS is returning (200OK)
>
> Media Attribute (a): rtpmap:96 G729/8000
>
> I think that the problem is that FS is not matching the codec, so it
> returns the first dynamic payload which is 96.
>
> I think that I've seen post with a similar issue, and the solution was
> to change the tag before it hit the switch, so, what I've done is to
> change the "switch_r_sdp" (I have the rest of the parameters correct
> due I also use it to dynamically change the codecs order) and it's
> changing the SDP, but when FS sends the 200OK it is returning to the
> endpoint:
>
> Media Attribute (a): rtpmap:96 G729/8000
>
> Which is exactly the same problem that I have without the
> transformation of the SDP.
>
> Is it correct? Do I have another solution?
>
> Thanks
>
>
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Re: [Freeswitch-users] SIP Overlap support?

[Anthony Minessale]

The patch was it's ability to accept subsequent invites.
Your problem is that in sip each new attempt to send an invite is another
call.

484 is a final response so the call with too few digits is terminated.


On Tue, Nov 3, 2009 at 9:57 AM, Dennis  wrote:

> hi anthony,
>
> i believe, that there is no problem with the communication between fs
> and the cirpack (everything works to smooth as if this could be
> possible). if fs sends the 484, the cirpack sends more digits to fs
> (if there are some), so this works as it should. the problem is, that
> fs ends the session/socket after a 484, so that the cirpack sends the
> following digits into another socket.
>
> you wrote about a "1 line patch", which might not have been
> implemented - at least it seems so.
>
> is there a way to get someone of the sofia devs to fix this small
> problem, so that fs sends the 484 without ending the session/socket
> and waiting for an answer of the cirpack? we would take care of the
> rest.
>
> kind regards,
> dennis
>
>
> 2009/10/15 Anthony Minessale :
> > right you can reply 484 in your dp at any time
> > 
> >
> > then it should try again.
> >
> > The bit i can't remember is if we committed a certain 1 line patch that
> > makes sofia parse the next invite to the same call properly, the patch
> was
> > to the sofia lib itself so test it and see.  I may need to dig up the
> answer
> > again from the sofia dev.
>
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[Freeswitch-users] portaudio error

[Frank Carmickle]

Hello

Debian lenny with svn15321

freeswi...@internal> load mod_portaudio
-ERR [module load file routine returned an error]

2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input 
devicefreeswi...@internal> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 
Cannot find an input device
2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading 
module /opt/freeswitch/mod/mod_portaudio.so
**Module load routine returned an error**


  




























  

 
Any help would be appreciated.

--FC

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[Freeswitch-users] FreeSWITCH 1.0.5 Scheduled For November 10; 1.0.5pre5 Now Available

[Michael Collins]

Greetings!

The latest FreeSWITCH pre-release is now available:

http://www.freeswitch.org/node/215

Please update and test as soon as possible. With the community's help we
should be able to hit our target of releasing version 1.0.5 next Tuesday
November 10. The FreeSWITCH developers appreciate all the hard work that the
community does on behalf of the project. Like most open source projects,
FreeSWITCH needs the community to "give back" a little and you all certainly
do that. Please keep up the great work.

-Michael
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Re: [Freeswitch-users] SIP Overlap support?

[Dennis]

hi anthony,

i believe, that there is no problem with the communication between fs
and the cirpack (everything works to smooth as if this could be
possible). if fs sends the 484, the cirpack sends more digits to fs
(if there are some), so this works as it should. the problem is, that
fs ends the session/socket after a 484, so that the cirpack sends the
following digits into another socket.

you wrote about a "1 line patch", which might not have been
implemented - at least it seems so.

is there a way to get someone of the sofia devs to fix this small
problem, so that fs sends the 484 without ending the session/socket
and waiting for an answer of the cirpack? we would take care of the
rest.

kind regards,
dennis


2009/10/15 Anthony Minessale :
> right you can reply 484 in your dp at any time
> 
>
> then it should try again.
>
> The bit i can't remember is if we committed a certain 1 line patch that
> makes sofia parse the next invite to the same call properly, the patch was
> to the sofia lib itself so test it and see.  I may need to dig up the answer
> again from the sofia dev.

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Re: [Freeswitch-users] Precompiled Windows Binaries

[Dave Stevenson]

Jeff,

thanks a lot for the reply. I was a little confused by the fact that the 
"SVN Snapshot" was some 10MB smaller than the Full 1.0.4 file so worried 
that I might lose something. As you say though, think that I'll cross my 
fingers and try the updated release. I am running FreeSwitch on a test 
machine at the moment until the target hardware arrives - hopefully 
tomorrow, so I can afford to have a little play.

You mentioned FreePBX V3. I had been fumbling around trying to work out what 
this is and from what I've read, it seems to provide a GUI Front End for 
configuring FreeSwitch ?

I am guessing that while it has been installed with FreeSwitch, I then need 
to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration 
on my hardware ?


When I start FreeSwitch, it does not automatically load the WAMPServer.

When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web 
browser, I can see the WampServer logo and various tools such as phpinfo() 
and phpmyadmin. FreePBX is there under Your Projects.

When I opened this up the first time, it appeared to want to install FreePBX 
over FreeSwitch, I tried to abort this when it was going to overwrite some 
FreeSwitch conf files and I thought I'd better not go on until I had a 
better idea what was happening. I backed out of the FreePBX install and now 
I can't get the FreePBX or phpmyadmin pages up again (missing files) so it 
looks like I'm going to have to reinstall anyway.

So, for next time,am I right in thinking that I should proceed with running 
the FreePBX install from the WAMPServer menu ?

regards
Dave



- Original Message - 
From: "Jeff Lenk" 
To: 
Sent: Tuesday, November 03, 2009 2:48 PM
Subject: Re: [Freeswitch-users] Precompiled Windows Binaries


>
> Hi Dave,
>
> These are supported by "Carlos Talbot" . They also include Freepbx v3
>
> Just as you said freeswitch-1.0.4.exe is the tagged release and
> freeswitch.exe is a newer svn snapshot.
>
> There should be no problems installing the new version allthough best to
> just try and see!
>
> Not sure why the newest one is from October 7th.
>
> Jeff
>
>
> Dave Stevenson wrote:
>>
>> Hi,
>>
>> I have read the Docs on the Wiki
>> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries)
>> but am still not sure of what the different Windows install files are.
>> Currently, the Windows Installer directory contains :-
>>
>> LATEST_SVN_15106 - 6 Bytes
>>
>> freeswitch-1.0.4.exe - 42 Megabytes
>>
>> freeswitch.exe - 32 Megabytes
>>
>> I have installed the freeswitch-1.0.4.exe file which is dated 3rd
>> September. The freeswitch.exe file is dated 7th October and think that it
>> contains the minor updates since 3rd September ?
>>
>> Could someone who knows FreeSwitch under windows help me understand the
>> two files please ?
>>
>> I chickened out of running the later exe in case it did something to the
>> running install of FreeSwitch 1.0.4, is it safe to run the newer exe with
>> the old one already installed ?
>> What will it actually do ?
>>
>> regards
>> Dave
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>>
>>
>
> -- 
> View this message in context: 
> http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> ___
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Re: [Freeswitch-users] Precompiled Windows Binaries

[Jeff Lenk]

Hi Dave,

These are supported by "Carlos Talbot" . They also include Freepbx v3

Just as you said freeswitch-1.0.4.exe is the tagged release and
freeswitch.exe is a newer svn snapshot.

There should be no problems installing the new version allthough best to
just try and see!

Not sure why the newest one is from October 7th.

Jeff


Dave Stevenson wrote:
> 
> Hi,
> 
> I have read the Docs on the Wiki
> (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries)
> but am still not sure of what the different Windows install files are.
> Currently, the Windows Installer directory contains :-
> 
> LATEST_SVN_15106 - 6 Bytes
> 
> freeswitch-1.0.4.exe - 42 Megabytes
> 
> freeswitch.exe - 32 Megabytes
> 
> I have installed the freeswitch-1.0.4.exe file which is dated 3rd
> September. The freeswitch.exe file is dated 7th October and think that it
> contains the minor updates since 3rd September ?
> 
> Could someone who knows FreeSwitch under windows help me understand the
> two files please ?
> 
> I chickened out of running the later exe in case it did something to the
> running install of FreeSwitch 1.0.4, is it safe to run the newer exe with
> the old one already installed ?
> What will it actually do ?
> 
> regards
> Dave
> ___
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> 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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[Freeswitch-users] Precompiled Windows Binaries

[Dave Stevenson]

Hi,

I have read the Docs on the Wiki 
(http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) but 
am still not sure of what the different Windows install files are. Currently, 
the Windows Installer directory contains :-

LATEST_SVN_15106 - 6 Bytes

freeswitch-1.0.4.exe - 42 Megabytes

freeswitch.exe - 32 Megabytes

I have installed the freeswitch-1.0.4.exe file which is dated 3rd September. 
The freeswitch.exe file is dated 7th October and think that it contains the 
minor updates since 3rd September ?

Could someone who knows FreeSwitch under windows help me understand the two 
files please ?

I chickened out of running the later exe in case it did something to the 
running install of FreeSwitch 1.0.4, is it safe to run the newer exe with the 
old one already installed ?
What will it actually do ?

regards
Dave___
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[Freeswitch-users] Users hanged up for unknown reason

[Maciej Aniserowicz]

Hi,
I have a strange problem. I control FS with commands sent by tcp in response
to events published via tcp. I do something like:
1) call 1st user
2) call 2nd user
3) 1st and 2nd talk
4) call another user
5) 1st and another talk
etc...

Sometimes (quite regularly) users are hanged up (with cause NORMAL_CLEARING)
even if they do not hangup manually.

I pasted one such scenario in pastebin
(http://pastebin.freeswitch.org/10955), it includes logs from commands sent
by me and events received from FS. Could someone take a look and see what am
I doing wrong?
The scenario includes 3 users
1st user (Unique-ID: f076261a-4537-40f2-b46d-933141320314) is supposed to be
connected all the time but gets diconnected
2nd user (Unique-ID: ebdfb398-ec82-4760-9f79-81364e0f37b6) is supposed to
talk for a few seconds and get killed
3rd user (Unique-ID: d5cd839e-793c-4b3c-adda-327841672a5f) is supposed to
work like 2nd user

All of them are simulated by dialplan extensions (using answer and playback
tools), but the same thing happends for xlite or cisco phone.

Maciej Aniserowicz


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

[Brian Stafford]

Brian Stafford wrote:
> Brian West wrote:
>   
>> You have to be doing it wrong then.
>>
>> Can you show us your dialplan you should have two extensions one for  
>> the lot range and one to attended transfer someone into the lot.
>>
>> /b
>>   
>> 
> The relevant excerpt from the dialplan is
>
> 
> 
> 
> 
> 
> 
>
> 
> 
> 
> 
> 
> 
>
> x410-419 are the slots and 420 parks a call. Parking by picking one of 
> 410-419 works fine and subsequently dialling them from another works 
> fine, I added x420 for the auto feature.
>
> Regards
> Brian
>
> _

Any clues what I'm doing wrong?  Is more information needed?

Brian

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[Freeswitch-users] How to get digitals and stop play when speak tts? Just like session:playAndGetDigits do

[Lei Tang]

Hi all, I'm writing lua ivr scirpt, Does some known how to get digitals and
stop play when speak tts?   Just like  session:playAndGetDigits do. Thanks
lots!

Best Regards!
-- 
Lei.Tang
lei.tl...@gmail.com
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