[Freeswitch-users] Mutual Registration of servers

[Samuel Abekah-Mensah]

Pardon me if this has been addressed already.
How does one go about having in the simplest instance 2 servers 
registering with each other on startup whereby the users registering 
would be able to call each other.
The 2 servers are in different domains.

Thanks.

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Re: [Freeswitch-users] Audiocodes PRI Gateway

[Imthiyaz Ahmed]

We are using Audiocodes and Sangoma netborder express GW with
Freeswitch . it works well.

Thanks
Imthiyaz

On Mon, Dec 7, 2009 at 9:34 AM, jay binks  wrote:
> Guys,
>   im after info from people with experience with AudioCodes Mediant 2k PRI
> Gateways.
> specifically how well they inter-op with Freeswitch, and how compliant their
> SIP stack is.
> I guess the bottom line is, would you recommend these gateways or would you
> suggest something else ?
> --
> Sincerely
>
> Jay
>
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-- 
Best Regards
G.Imthiyaz Ahmed
PeopleTech systems (P) ltd
http://peopletech.co.in

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Re: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario

[Michael Jerris]

Please report bugs to jira.freeswitch.org.

Mike

On Dec 6, 2009, at 11:45 PM, Seven Du wrote:

> Hi,
> 
> I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
> to trunk code, no sound after att_xfer.
> 
> Then I rebuild FS 15807 with a fresh checkout, but still using the old
> conf/ settings, sound is ok, but there are other problems:
> 
> A call B, and B att_xfer C
> 
> 1) origination_cancel_key not working. no even no DTMF log in FS when
> I press # or any other key, I tried with Zoiper and Snom(on the B leg)
> 2) when C answers, B immediately hangup, so B has no chance talk to C
> 
> Could this be a problem? I pasted logs:
> 
> http://pastebin.freeswitch.org/11417


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[Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario

[Seven Du]

Hi,

I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
to trunk code, no sound after att_xfer.

Then I rebuild FS 15807 with a fresh checkout, but still using the old
conf/ settings, sound is ok, but there are other problems:

A call B, and B att_xfer C

1) origination_cancel_key not working. no even no DTMF log in FS when
I press # or any other key, I tried with Zoiper and Snom(on the B leg)
2) when C answers, B immediately hangup, so B has no chance talk to C

Could this be a problem? I pasted logs:

http://pastebin.freeswitch.org/11417

Thanks.

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[Freeswitch-users] Audiocodes PRI Gateway

[jay binks]

Guys,
  im after info from people with experience with AudioCodes Mediant 2k PRI
Gateways.

specifically how well they inter-op with Freeswitch, and how compliant their
SIP stack is.

I guess the bottom line is, would you recommend these gateways or would you
suggest something else ?

-- 
Sincerely

Jay
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Re: [Freeswitch-users] Bad sound quality while eavesdropping

[Michael Jerris]

This bug has been now closed out in jira due to no response for requested 
information.  If you wish to resolve this issue please follow up on your bugs 
when information is requested.

Mike

On Oct 12, 2009, at 12:04 PM, Maciej Aniserowicz wrote:

> 
> Nope, I wanted to make sure that this is indeed a bug. I opened an issue in
> JIRA before regarding some other matter and it turned out to be my mistake,
> so I decided to try mailing list first this time.
> MA
> 
> 
> 
> Brian West wrote:
>> 
>> Did you open a jira and attach all the info?
>> 
>> /b
>> 
>> On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
>> 
>>> Yes, I confirmed that with Wireshark (filter "rtp and ip.src ==  
>>> ). RTP packets are sent every 20ms.
>>> 
>>> MAniserowicz


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[Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled.

[DJB]

I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 
(exported) with only one thing difference which is the first one is running 
with -hp enabled; however, I have noticed that the one with -hp option consumed 
double in memory usage than the other one.  

I wonder whether anyone can explain why.  Thank you.

Please see below:

--
top - 01:01:42 up 53 days,  2:45,  1 user,  load average: 0.22, 0.28, 0.29
Tasks: 143 total,   1 running, 142 sleeping,   0 stopped,   0 zombie
Cpu(s):  0.9%us,  0.2%sy,  0.0%ni, 96.4%id,  2.5%wa,  0.0%hi,  0.0%si,  0.0%st
Mem:   8174164k total,  7550092k used,   624072k free,   187568k buffers
Swap: 10223608k total,0k used, 10223608k free,  5417524k cached

  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND

30750 root  -2 -10 1823m 1.5g  20m S  8.6 19.8   1153:40 freeswitch

4418 session(s) 14/100

root 30750  2.1 19.9 1879252 1634300 ? S___
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Re: [Freeswitch-users] Database suggestions/pointers/?

[Tim Uckun]

On Mon, Dec 7, 2009 at 9:07 AM, Steve Klein  wrote:
> Greetings. We need to add database access to an IVR application we are
> prototyping. Based on FS “best practice” suggestions, we are using Lua for
> the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite
> would be a recommended approach. However, we can’t find any examples of this
> combo anywhere. So, what is the “best practice” scripting + database
> recommendation for a high-volume database-driven FS app?
>

I would suggest you take a look at freeswitcher
(http://github.com/bougyman/freeswitcher).

The good thing is that it's ruby and therefore you can use any
database compatible with ruby (that's all of them pretty much). You
can also use an ORM of your choice or if you don't want to use an ORM
you can use the amazingly fantastic sequel library.

Being ruby it will run outside of the freeswitch memory space and you
will have to use the inbound/outbound socket API. That may be a good
thing if you want to separate your database and IVR logic from the
machine running your freeswitch.

Ruby is pretty easy to pick up if you don't know it and there are a
wealth of libraries if you want to do other things like connect to web
sites, manipulate XML, etc.

There is also a liverpie http://github.com/jsgoecke/liverpie which is
more of a proxy thing you can interface with any language.

I am sure lua is nice but it seems like people are having some
problems with ODBC, memory leaks etc when it comes to databases. If
you go a ruby library that all goes away.

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[Freeswitch-users] lua+sqlite example?

[Steve Klein]

Greetings. We are attempting to add sqlite access to an IVR application we
are prototyping. We are using lua for the scripts. Is there an example
anywhere of a lua + sqlite script? Do we need to install luasql? Any
help/pointers greatly appreciated.

 

--Steve Klein

 

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[Freeswitch-users] Database suggestions/pointers/?

[Steve Klein]

Greetings. We need to add database access to an IVR application we are
prototyping. Based on FS "best practice" suggestions, we are using Lua for
the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite
would be a recommended approach. However, we can't find any examples of this
combo anywhere. So, what is the "best practice" scripting + database
recommendation for a high-volume database-driven FS app?

 

Any help/pointers greatly appreciated!

 

--Steve Klein

 

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Re: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall???

[mailinglist]

Hi Joseph
 
Ahh, yes, that got rid of that error :-)
Now on to the next one. 
So now it's connecting, both at my provider, and my softphone.
Now I have to figure out why it tells me 'Call failed: not found' when I try to 
call out of the system...
But I think that's a task for tomorrow when I'm more awake :-D
 
Thanks!
Fribert

>>> 06-12-2009 kl. 22:01 skrev "Joseph L. Casale"  i 
>>> meddelelsen 
>>> :

>Registrations:
>=
>=
>As far as I can see, everything looks ok, except for the
>2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 
>87.61.18.196
>I'm wondering WHY it wants a domain on the external IP???
> 
> 
>I then started the SIP softphone, and got:
> 
> 2009-12-06 19:36:23.588241 [WARNING] sofia_reg.c:1755 Can't find user 
> [1...@87.61.18.196]
>You must define a domain called '87.61.18.196' in your directory and add a 
>user with the id="1001" at 
>   tribute
>and you must configure your device to use the proper domain in it's 
>authentication credentials.

Yea, it looks like your server is taking the domain of the wan nic. I don't 
begin to claim I know
all there is to know about this (still lurking while I learn as well...) but I 
got a lab'ed up pfSense
box to work only after I edited vars.xml and set:

  

Where 10.0.0.1 was the ip my internal.xml bound to. I assumed it had something 
to do with nat
and clients in the lan accessing the wan ip.

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Re: [Freeswitch-users] Lua and database access to core_db

[Anthony Minessale]

Yes, exactly my point.

Like I said you have several choices be paitent till we have time to
code it for free, post a bounty to increase the chance somone will do it
from the community, hire someone to set it up for you or keep trying
yourself.

Did I miss something?

On Dec 6, 2009 3:38 PM, "Jon Bruel"  wrote:

 The MySQL version is 5.1.37. Well I’m not an expert on every field, and I
have no skills in the C, include libraries, and the art of compiling. For
this I have to follow the guidelines. But it wouldn’t harm the FS project if
it generally became more accessible to the race of non-specialists, which I
hereby represent.



*Jon Brüel*

  --

*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony
Minessale
*Sent:* 6. december 2009 20:42
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Lua and database access to core_db

  Most of this is unfortunatly because you do not have the proper skill to
set it up because, wit...

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Re: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service?

[Peter P GMX]

Oh that's a lot of money,

anybody else needs this feature, so we may share a bounty?

Best
regards
Peter

Anthony Minessale schrieb:
>
> Someone else was asking about this too.
> I could probably write a dictaction mod in c like the one I made for
> asterisk starting at about $3k depending on the featureset required.
>
>> On Dec 6, 2009 10:30 AM, "Peter P GMX" > > wrote:
>>
>> Hello,
>>
>> I would like to offer a dictation service to a secretary.
>> Means:
>>
>>* the boss is dictating some text on a certain phone number
>>* the secretary picks up the recording on the phone and types the
>>  text into the computer
>>
>> As the secretary is not able to type in as fastly as heir boss is able
>> to speak, she needs some kind of pause and rewind button.
>> 1st question: Is there any functionality available for example in
>> uuid_broadcast?
>> 2nd question: How much would be the effort to implement this
>> (uuid_broadcast_pause, uuid_broadcast_UNpause,
>> uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this.
>>
>> Best regards
>> Peter
>>
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> 
>
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Re: [Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

The MySQL version is 5.1.37. Well I'm not an expert on every field, and I have 
no skills in the C, include libraries, and the art of compiling. For this I 
have to follow the guidelines. But it wouldn't harm the FS project if it 
generally became more accessible to the race of non-specialists, which I hereby 
represent.

Jon Brüel


From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony 
Minessale
Sent: 6. december 2009 20:42
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua and database access to core_db


Most of this is unfortunatly because you do not have the proper skill to set it 
up because, with the proper skills, all of the ways you tried would have ended 
sucessfully.  I say that beacause I have had many users use each of the 
different methods in your list of failures only they were sucessful.

What you are asking for is possible but would require many hours of coding just 
to help solve your problem.
You would have to wait a really long time until someone had the time to do it 
for free or post a bounty for it.  Probably about 1k in consulting time.  It 
may be cheaper for you to pay a consultant to set up one of the ways known to 
work.  These are your options as I see it.
On Dec 6, 2009 12:20 PM, "Lon Baker" 
mailto:l...@kickasspixels.com>> wrote:

Jon,

What version of MySQL are you using?


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Re: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall???

[Joseph L. Casale]

>Registrations:
>=
>=
>As far as I can see, everything looks ok, except for the
>2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 
>87.61.18.196
>I'm wondering WHY it wants a domain on the external IP???
> 
> 
>I then started the SIP softphone, and got:
> 
> 2009-12-06 19:36:23.588241 [WARNING] sofia_reg.c:1755 Can't find user 
>[1...@87.61.18.196]
>You must define a domain called '87.61.18.196' in your directory and add a 
>user with the id="1001" at 
>   tribute
>and you must configure your device to use the proper domain in it's 
>authentication credentials.

Yea, it looks like your server is taking the domain of the wan nic. I don't 
begin to claim I know
all there is to know about this (still lurking while I learn as well...) but I 
got a lab'ed up pfSense
box to work only after I edited vars.xml and set:

  

Where 10.0.0.1 was the ip my internal.xml bound to. I assumed it had something 
to do with nat
and clients in the lan accessing the wan ip.

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Re: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360...

[Klaus Hochlehnert]

Ok, set_profile_var did the trick and also works with intercepted calls.

Thanks, Klaus

From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony 
Minessale
Sent: Sunday, December 06, 2009 8:32 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] caller_id_name + caller_id_number and Snom 
360...


Or set it to true depending on the case
Also consider using set_profile_var to set the caller id explicitly instead of 
using effective.  There is also effective_callee_id name and number you could 
set on the a leg.  You'll have to expirement but the one mathieu said is your 
best bet.

On Dec 6, 2009 12:47 PM, "Mathieu Rene" 
mailto:mrene_li...@avgs.ca>> wrote:
Hi Klaus,

Try setting ignore_display_updates=false

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca





On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote:

> Hi, >   > I just checked the SIP traces and it looks like FS sends a sipfrag 
> message to the phone ...
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Re: [Freeswitch-users] Lua and database access to core_db

[Anthony Minessale]

Most of this is unfortunatly because you do not have the proper skill to set
it up because, with the proper skills, all of the ways you tried would have
ended sucessfully.  I say that beacause I have had many users use each of
the different methods in your list of failures only they were sucessful.

What you are asking for is possible but would require many hours of coding
just to help solve your problem.
You would have to wait a really long time until someone had the time to do
it for free or post a bounty for it.  Probably about 1k in consulting time.
It may be cheaper for you to pay a consultant to set up one of the ways
known to work.  These are your options as I see it.

 On Dec 6, 2009 12:20 PM, "Lon Baker"  wrote:

Jon,

What version of MySQL are you using?


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[Freeswitch-users] A few questions about Polycom setup

[Yehavi Bourvine]

Hello,

  I have a few questions about Ploycom's usage and provisioning for which I
found no answers neither at the docs nor on the WEB:


   - I would like to enable SIP/TLS. for this I have to import the root
   certificate. How can I do it via the XML config files? the only method I
   found is via the phone's interface, but what do you do when you have tens
   and more of them?
   - Since the phone is limited to 3way conference I would like it to use a
   conference room on the server. I've defined:

   

   - The result is that when A calls B (the polycom phone) which tries to
   conference with C is that B does a conference with C and the conference room
   and A is left on hold...

   Thanks! __Yehavi:
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Re: [Freeswitch-users] Translating DTMF from RFC2833 to INFO

[Anthony Minessale]

Some more bad news for you, info dtmf spec has expired and has been
abandoned.  Wait till you see what they did accept instead..

On Dec 6, 2009 1:22 PM, "Metik"  wrote:

Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".

Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by the voip dial peer.

I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833)
previously in the wild.  Unlike some other SIP feature servers,  I have
not had issues (with RFC 2833) between FS and Cisco IOS gateways.

Although unrelated to FS or any other SIP feature server, I have seen
some issues when multple dtmf relay types are left enabled on a voip
dial peer.  Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.

-metik

Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik <
freeswitch-users-l...@metik.com
> >

> > You previously stated that your Cisco gateway has some "bug" that >
prevents you from us...

>  >
> _...
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Re: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360...

[Anthony Minessale]

Or set it to true depending on the case
Also consider using set_profile_var to set the caller id explicitly instead
of using effective.  There is also effective_callee_id name and number you
could set on the a leg.  You'll have to expirement but the one mathieu said
is your best bet.

On Dec 6, 2009 12:47 PM, "Mathieu Rene"  wrote:

Hi Klaus,

Try setting ignore_display_updates=false

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca




On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote:

> Hi, >   > I just checked the SIP traces and it looks like FS sends a
sipfrag message to the phone ...
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Re: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service?

[Anthony Minessale]

Someone else was asking about this too.
I could probably write a dictaction mod in c like the one I made for
asterisk starting at about $3k depending on the featureset required.

On Dec 6, 2009 10:30 AM, "Peter P GMX"  wrote:

Hello,

I would like to offer a dictation service to a secretary.
Means:

   * the boss is dictating some text on a certain phone number
   * the secretary picks up the recording on the phone and types the
 text into the computer

As the secretary is not able to type in as fastly as heir boss is able
to speak, she needs some kind of pause and rewind button.
1st question: Is there any functionality available for example in
uuid_broadcast?
2nd question: How much would be the effort to implement this
(uuid_broadcast_pause, uuid_broadcast_UNpause,
uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this.

Best regards
Peter

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Re: [Freeswitch-users] Translating DTMF from RFC2833 to INFO

[Metik]

Unless the IOS you are running is extremely buggy, "debug voip ccapi" 
commands should not provide you with that detail, what you really want 
to use is "debug voip rtp session named-event".

Normal SIP-to-PSTN calls should use both a pots and voip dial peer but  
DTMF relay type is determined by the voip dial peer.

I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) 
previously in the wild.  Unlike some other SIP feature servers,  I have 
not had issues (with RFC 2833) between FS and Cisco IOS gateways.

Although unrelated to FS or any other SIP feature server, I have seen 
some issues when multple dtmf relay types are left enabled on a voip 
dial peer.  Also, there are some (older) IOS versions that have issues 
with DTMF duration which cause digits to be misinterpreted by the 
far-end (PSTN/POTS) but not ignored altogether. 

-metik


Yehavi Bourvine wrote:
> Hello Metik,
>
>
>  
> 2009/12/6 Metik  >
>
> You previously stated that your Cisco gateway has some "bug" that
> prevents you from using RFC2833, did you enable "dtmf-relay
> rtp-nte" on
> the voip dial-peer that the call is using?
>
>  
> It is a PSTN dialpeer here, and it cannot be defined on it...
>  
>
> Unless you have configured the Cisco to support assymetric SDP or are
> using a non-default "rtp payload-type nte" setting that does not agree
> to well with FS's (default) "rfc2833-pt" setting, you should not
> have to
> use (SIP) INFO unless you want to.
>
> I would recommend doing the following to ensure you are hitting the
> correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):
>
> command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)
>
>  
> Unfortunately this does not work on PSTN dial peers.
>  
>
>
> Also, you can sift through "show sip-ua calls" for the call and ensure
> that the value of "Negotiated Dtmf-relay" is "rtp-nte".
>
>  
> This indeed shows that it has negotiated rtp-nte. Even when I do debug 
> for CCAPI events (I think) I see it decodes the DTMFs; however, it 
> ignores them while it accepts them via INFO. As I said: I guess this 
> is a bug.
>  
> Since the gateway is on a remote site I hesitate on upgrading it until 
> I hae the chance to go there.
>  
>   Thanks, __Yehavi:
> 
>
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Re: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360...

[Mathieu Rene]

Hi Klaus,

Try setting ignore_display_updates=false

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca




On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote:


Hi,

I just checked the SIP traces and it looks like FS sends a sipfrag  
message to the phone with
caller_id_name and caller_id_number instead of  
effective_caller_id_name and effective_caller_id_number values.


Thanks, Klaus


From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Klaus Hochlehnert

Sent: Sunday, December 06, 2009 4:04 AM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] caller_id_name + caller_id_number and  
Snom 360...


Hi,

currently I’m testing the newest FS trunk.
Now I need a hint how to set up an “old” behavior of version 1.0.4.

Here’s the scenario:
- Incoming call from caller_id_name: abc and caller_id_number: 123
- Now I set effective_caller_id_name: xyz and  
effective_caller_id_number: 456

- Leg B (Snom 360) is ringing and displays the new values (xyz + 456)
- After the pickup the Leg B is switching back to the “old” values  
and displays abc + 123


But I would rather see the new values during the call (as it was in  
version 1.0.4).

What do I need to change?

Thanks, Klaus


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Re: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360...

[Klaus Hochlehnert]

Hi,

I just checked the SIP traces and it looks like FS sends a sipfrag message to 
the phone with
caller_id_name and caller_id_number instead of effective_caller_id_name and 
effective_caller_id_number values.

Thanks, Klaus


From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Klaus 
Hochlehnert
Sent: Sunday, December 06, 2009 4:04 AM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360...

Hi,

currently I'm testing the newest FS trunk.
Now I need a hint how to set up an "old" behavior of version 1.0.4.

Here's the scenario:
- Incoming call from caller_id_name: abc and caller_id_number: 123
- Now I set effective_caller_id_name: xyz and effective_caller_id_number: 456
- Leg B (Snom 360) is ringing and displays the new values (xyz + 456)
- After the pickup the Leg B is switching back to the "old" values and displays 
abc + 123

But I would rather see the new values during the call (as it was in version 
1.0.4).
What do I need to change?

Thanks, Klaus


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[Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall???

[mailinglist]

Hi Peter
 
Ok, I got the net changed in acl.conf.xml.
I then tried setting console_loglevel, but I don't see any output on the 
console, it could very well be because it's a FreeBSD, and has very limited 
console.
But after a restart it registers!
So for some reason it needed a nudge there, very interesting.
 
So now I have a local extension registered, and a provider registered, now I 
just need them to communicate.
As I understand it, that's the dialplan I have to look at.
 
I only have one provider hooked up, so a dial should be simple, right?
 
It as Order 001
Condition  ^0(.\d+)$
action bridge
 
As I understand it, it should react on the 0 (for outside dialing) and then 
strip it
And the action bridges the call to the outside.
 
But I guess I'm missing something, because I just get a 'temporarily 
unavailable' shown in the xlite.


>>> 06-12-2009 kl. 15:14 skrev Peter P GMX  i 
>>> meddelelsen <4b1bbc52.3050...@gmx.net>:

Concerning,
> Which I'm kinda confused about, I don't have any 192.168 net here??
I think, this is a default entry in the acl.conf.xml. Please check the
entries there. But normally this shouldn't stop freeswitch from working
and handling requests.

Can you set the console_log_level to "debug" in vars.xml and post you
console output when the phone tries to register? You may also grep the
network traffic on port 5060
(e.g. ngrep -d any port 5060 -W byline) on your machine, to see what's
wrong.

Best regards
Peter

mailinglist schrieb:
> Hi Adam
>  
> Excellent first steps!
> Thankyou for the hint.
> Now I hope somebody can tell me what I'm doing wrong next...
>  
> I've gotten it to register to the testprovider here (musimi.dk), but I
> get an error when I create an account for testing with the X-Lite phone.
>  
> It displays 403 forbidden in the display.
>  
> I've created an account on FreeSwitch
>  
> extension 1001
> password 1001
> mailbox 1001
> voicemail password 1001
> account code 1001
> Effective Caller ID Name Fribert
> Effective Caller ID Number 4692 (the Musimi number)
> Voicemail Mail To 
> Voicemail Attach File true
> User Context default
> Call Group <>
> Enabled true
> Extension Description Test number
>  
> In the X-Lite
> Display Name Fribert
> User name 1001
> Password 1001
> Autorization user name 1001
> Domain LAN-IP-OF-pfSense
>  
> Check in Register with domain and receive incoming calls
> Check in domain.
>  
> That's about it.
>  
> Looking on the status page, I can see these lines in the log:
> 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding
> 192.168.42.0/24 (deny) to list lan
> 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding
> 192.168.42.42/32 (allow) to list lan
> 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up
> listening on 0.0.0.0:8021
> 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up
> listening on 0.0.0.0:8021
>  
> Which I'm kinda confused about, I don't have any 192.168 net here???
> But as it also primarily forbids it, except .42 to allow, I'm
> wondering if it could be something internal?
>
> Best regards
>  
>
> >>> 05-12-2009 kl. 18:52 skrev "Adam Ford"  i
> meddelelsen <0e0013f55e224674a1361329cf7a8...@redbonez>:
>
> I used the pfSense FreeSWITCH for awhile, as it is the only GUI
> FreeSWITCH I have found with a stable release.  It was very easy to
> use, I would recommend it if you just want a quick base system with
> standard features.  Though, I ended up switching to a compiled version
> of FreeSWITCH in order to make the customizations I needed for my office.
>
> http://doc.pfsense.org/index.php/FreeSWITCH
>
> -AF
>
> 
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of
> *mailinglist
> *Sent:* Saturday, December 05, 2009 2:47 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* [Freeswitch-users] Setting up a FreeSwitch system on a
> pfSensefirewall???
>
>  
>
> Has anybody done this?
>
>  
>
> I'm completely at a loss, having tinkered very little with Asterisk,
> and giving up on that, I wonder if there's any help to be found on
> FreeSwitch?
>
> Anybody that can give pointers to a good step-by-step instruction?
>
>  
>
> I want to have it handle my two sip-phones (siemens dect ip and spa
> 901), and handle a sip account at my provider.
>
> Of course transferring calls between the two, as well as group calls
> would be a nice benefit.
>
>  
>
> 
>
> ___
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Re: [Freeswitch-users] Lua and database access to core_db

[Lon Baker]

Jon,

What version of MySQL are you using?
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[Freeswitch-users] uuid_broadcast with pause and rewind for dictation service?

[Peter P GMX]

Hello,

I would like to offer a dictation service to a secretary.
Means:

* the boss is dictating some text on a certain phone number
* the secretary picks up the recording on the phone and types the
  text into the computer

As the secretary is not able to type in as fastly as heir boss is able
to speak, she needs some kind of pause and rewind button.
1st question: Is there any functionality available for example in
uuid_broadcast?
2nd question: How much would be the effort to implement this
(uuid_broadcast_pause, uuid_broadcast_UNpause,
uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this.

Best regards
Peter

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Re: [Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

Dear all

Some feedback regarding using Lua to access core database:

First of all, I did not succeed to get SQLite drivers in Lua or ODBC-drivers in 
Lua to work. The SQLite driver did compile OK, but there was an error when 
loading into Lua. The ODBC driver did also compile OK, did load into Lua, but 
could not connect. Accessing the SQLite from the Linux console using "isql -v 
" worked OK. The problems may be related to the present Linux 
distribution, which is Ubuntu 9.1 server. Unfortunately the public searchable 
information about Lua ODBC driver problems is sparse.

So I continued to try to get the FS to use MySQL as the core db. A number of 
problem occurred, which I did not find solution for in the FS documents. The 
problems and solutions are described below:

1)   The core database is not automatically created by FS, therefore I 
created it manually.
2)   During startup, the FS test for transaction support, and this test 
failed. To achieved transaction support with MySQL and MyODBC, three things had 
to be changed:
a.   A line was added in my.cnf to force innoDB as the default table: under 
the [mysqld] header, the following line was added: set-variable = 
default-table-type=InnoDB.
b.  The a line under the DNS was added to allow for multiple line statement 
support: option = 67108864. (ODBC version is 3.51).
3)   After these changes the transaction worked, but all the tables in the 
core db were not created, therefore I copied the structure from the SQLite 
tables into tables with the same names in the MySQL database. This exercise 
also showed what the problem was: MySQL could not create tables with many 
VARCHAR type files with a size of 4096 (sound very big?). The size was reduced 
to 255, and most of the tables were created OK. One table still gave problems: 
the interface table. One of the fields is called key, which is a reserved word 
in MySQL, and by backticking the word key in the create statement, it worked.
4)   Finally the FS started up using the MySQL, but errors splashed over 
the screen just after startup. There was a problem creating new records in the 
interface table, the problem was the key field. Changing the insert statement 
in switch.core.sqldb.c file by backticking the key field name and recompiling 
the FS solved that problem.

I guess this will be fixed in later releases and I hope this will assist the 
brave programmers!

I would like to argue for the development of SQLite connectivity in Lua. The 
ODBC core solution is not as clean as a direct database connection, and as long 
as this is limited to SQLite, a direct connection from "recommended script 
language" would be the cleanest solution. Further, it would be nice if 
everything works after having compiled the FS package.

Jon Brüel
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882


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Re: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall???

[Peter P GMX]

Concerning,
> Which I'm kinda confused about, I don't have any 192.168 net here??
I think, this is a default entry in the acl.conf.xml. Please check the
entries there. But normally this shouldn't stop freeswitch from working
and handling requests.

Can you set the console_log_level to "debug" in vars.xml and post you
console output when the phone tries to register? You may also grep the
network traffic on port 5060
(e.g. ngrep -d any port 5060 -W byline) on your machine, to see what's
wrong.

Best regards
Peter

mailinglist schrieb:
> Hi Adam
>  
> Excellent first steps!
> Thankyou for the hint.
> Now I hope somebody can tell me what I'm doing wrong next...
>  
> I've gotten it to register to the testprovider here (musimi.dk), but I
> get an error when I create an account for testing with the X-Lite phone.
>  
> It displays 403 forbidden in the display.
>  
> I've created an account on FreeSwitch
>  
> extension 1001
> password 1001
> mailbox 1001
> voicemail password 1001
> account code 1001
> Effective Caller ID Name Fribert
> Effective Caller ID Number 4692 (the Musimi number)
> Voicemail Mail To 
> Voicemail Attach File true
> User Context default
> Call Group <>
> Enabled true
> Extension Description Test number
>  
> In the X-Lite
> Display Name Fribert
> User name 1001
> Password 1001
> Autorization user name 1001
> Domain LAN-IP-OF-pfSense
>  
> Check in Register with domain and receive incoming calls
> Check in domain.
>  
> That's about it.
>  
> Looking on the status page, I can see these lines in the log:
> 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding
> 192.168.42.0/24 (deny) to list lan
> 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding
> 192.168.42.42/32 (allow) to list lan
> 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up
> listening on 0.0.0.0:8021
> 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up
> listening on 0.0.0.0:8021
>  
> Which I'm kinda confused about, I don't have any 192.168 net here???
> But as it also primarily forbids it, except .42 to allow, I'm
> wondering if it could be something internal?
>
> Best regards
>  
>
> >>> 05-12-2009 kl. 18:52 skrev "Adam Ford"  i
> meddelelsen <0e0013f55e224674a1361329cf7a8...@redbonez>:
>
> I used the pfSense FreeSWITCH for awhile, as it is the only GUI
> FreeSWITCH I have found with a stable release.  It was very easy to
> use, I would recommend it if you just want a quick base system with
> standard features.  Though, I ended up switching to a compiled version
> of FreeSWITCH in order to make the customizations I needed for my office.
>
> http://doc.pfsense.org/index.php/FreeSWITCH
>
> -AF
>
> 
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of
> *mailinglist
> *Sent:* Saturday, December 05, 2009 2:47 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* [Freeswitch-users] Setting up a FreeSwitch system on a
> pfSensefirewall???
>
>  
>
> Has anybody done this?
>
>  
>
> I'm completely at a loss, having tinkered very little with Asterisk,
> and giving up on that, I wonder if there's any help to be found on
> FreeSwitch?
>
> Anybody that can give pointers to a good step-by-step instruction?
>
>  
>
> I want to have it handle my two sip-phones (siemens dect ip and spa
> 901), and handle a sip account at my provider.
>
> Of course transferring calls between the two, as well as group calls
> would be a nice benefit.
>
>  
>
> 
>
> ___
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>   

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Re: [Freeswitch-users] Translating DTMF from RFC2833 to INFO

[Yehavi Bourvine]

Hello Metik,



2009/12/6 Metik 

> You previously stated that your Cisco gateway has some "bug" that
> prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
> the voip dial-peer that the call is using?
>
>
It is a PSTN dialpeer here, and it cannot be defined on it...


> Unless you have configured the Cisco to support assymetric SDP or are
> using a non-default "rtp payload-type nte" setting that does not agree
> to well with FS's (default) "rfc2833-pt" setting, you should not have to
> use (SIP) INFO unless you want to.
>
> I would recommend doing the following to ensure you are hitting the
> correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):
>
> command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)
>
>
Unfortunately this does not work on PSTN dial peers.


>
> Also, you can sift through "show sip-ua calls" for the call and ensure
> that the value of "Negotiated Dtmf-relay" is "rtp-nte".
>
>
This indeed shows that it has negotiated rtp-nte. Even when I do debug for
CCAPI events (I think) I see it decodes the DTMFs; however, it ignores them
while it accepts them via INFO. As I said: I guess this is a bug.

Since the gateway is on a remote site I hesitate on upgrading it until I hae
the chance to go there.

  Thanks, __Yehavi:
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Re: [Freeswitch-users] Translating DTMF from RFC2833 to INFO

[Metik]

You previously stated that your Cisco gateway has some "bug" that 
prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on 
the voip dial-peer that the call is using?

Unless you have configured the Cisco to support assymetric SDP or are 
using a non-default "rtp payload-type nte" setting that does not agree 
to well with FS's (default) "rfc2833-pt" setting, you should not have to 
use (SIP) INFO unless you want to.

I would recommend doing the following to ensure you are hitting the 
correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):

command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)

output:
DTMF Relay = enabled,
dtmf-relay = rtp-nte,

example:

show dialplan number 5551212 | i (dtmf-relay|DTMF Relay)
DTMF Relay = enabled,
dtmf-relay = rtp-nte,

Also, you can sift through "show sip-ua calls" for the call and ensure 
that the value of "Negotiated Dtmf-relay" is "rtp-nte". 

-metik


Yehavi Bourvine wrote:
> Hello Ognjen,
>  
>   From the tests I've done it is not so... When I set the profile to 
> use INFO, and a phone calls and asks for RFC2833 (phone-events in the 
> SDP) the FreeSwich ignores it (does not have phone-events field in the 
> reply SDP) which causes the phone to not send RFC2833 events...
>  
>Regards, __Yehavi:
>
> 2009/12/3 Ognjen Seslija mailto:osesl...@gmail.com>>
>
> Bear in mind that FS will accept both 2833 and INFO in any profile
> on an inbound call. Param "dtmf-type" is valid only for outbound
> calls from the profile.
>
> Ognjen
>
> On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine
> mailto:yehavi.bourv...@gmail.com>> wrote:
>
> Hello,
>  
>   I have Polycom phones which send only RFC-2833 (or inband
> which I dislike) and they should go out to the PSTN via a
> Cisco gateway. The Cisco gateway has some bug and accepts only
> INFO.
>  
> I did a few tests:
>
>*
>   Some of the phones are on different profile than the
>   Cisco. On their profile I set 'dtmf-type=rfc2833' and on
>   the Cisco's profile I set  'dtmf-type=info' and
>   Freeswitch did the translation. All works ok...
>*
>   Some of the phones are on the same profile as the Cisco,
>   so I must set dtmf-type to rfc2833; it works with
>   internal applications (like voicemail) but does not work
>   through the Cisco as it misinterprets the rfc2833
>
>  
> Is there a way to set some variable (or a parameter to the
> bridge application) to do the translation?
>  
>  Thanks! __Yehavi:
>
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Re: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall???

[mailinglist]

Hi Adam
 
Excellent first steps!
Thankyou for the hint.
Now I hope somebody can tell me what I'm doing wrong next...
 
I've gotten it to register to the testprovider here (musimi.dk), but I get an 
error when I create an account for testing with the X-Lite phone.
 
It displays 403 forbidden in the display.
 
I've created an account on FreeSwitch
 
extension 1001
password 1001
mailbox 1001
voicemail password 1001
account code 1001
Effective Caller ID Name Fribert
Effective Caller ID Number 4692 (the Musimi number)
Voicemail Mail To 
Voicemail Attach File true
User Context default
Call Group <>
Enabled true
Extension Description Test number
 
In the X-Lite
Display Name Fribert
User name 1001
Password 1001
Autorization user name 1001
Domain LAN-IP-OF-pfSense
 
Check in Register with domain and receive incoming calls
Check in domain.
 
That's about it.
 
Looking on the status page, I can see these lines in the log:
2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding 192.168.42.0/24 
(deny) to list lan
2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding 192.168.42.42/32 
(allow) to list lan
2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening 
on 0.0.0.0:8021
2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening 
on 0.0.0.0:8021
 
Which I'm kinda confused about, I don't have any 192.168 net here???
But as it also primarily forbids it, except .42 to allow, I'm wondering if it 
could be something internal?

Best regards
 

>>> 05-12-2009 kl. 18:52 skrev "Adam Ford"  i meddelelsen 
>>> <0e0013f55e224674a1361329cf7a8...@redbonez>:


I used the pfSense FreeSWITCH for awhile, as it is the only GUI FreeSWITCH I 
have found with a stable release.  It was very easy to use, I would recommend 
it if you just want a quick base system with standard features.  Though, I 
ended up switching to a compiled version of FreeSWITCH in order to make the 
customizations I needed for my office.
http://doc.pfsense.org/index.php/FreeSWITCH
-AF


From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of mailinglist
Sent: Saturday, December 05, 2009 2:47 AM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Setting up a FreeSwitch system on a 
pfSensefirewall???

 

Has anybody done this?

 

I'm completely at a loss, having tinkered very little with Asterisk, and giving 
up on that, I wonder if there's any help to be found on FreeSwitch?

Anybody that can give pointers to a good step-by-step instruction?

 

I want to have it handle my two sip-phones (siemens dect ip and spa 901), and 
handle a sip account at my provider.

Of course transferring calls between the two, as well as group calls would be a 
nice benefit.

 
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Re: [Freeswitch-users] Translating DTMF from RFC2833 to INFO

[Yehavi Bourvine]

Hello Ognjen,

  From the tests I've done it is not so... When I set the profile to use
INFO, and a phone calls and asks for RFC2833 (phone-events in the SDP) the
FreeSwich ignores it (does not have phone-events field in the reply SDP)
which causes the phone to not send RFC2833 events...

   Regards, __Yehavi:

 2009/12/3 Ognjen Seslija 

> Bear in mind that FS will accept both 2833 and INFO in any profile on an
> inbound call. Param "dtmf-type" is valid only for outbound calls from the
> profile.
>
> Ognjen
>
>   On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine <
> yehavi.bourv...@gmail.com> wrote:
>
>>   Hello,
>>
>>   I have Polycom phones which send only RFC-2833 (or inband which I
>> dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco
>> gateway has some bug and accepts only INFO.
>>
>> I did a few tests:
>>
>>- Some of the phones are on different profile than the Cisco. On their
>>profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set
>>'dtmf-type=info' and Freeswitch did the translation. All works ok...
>>- Some of the phones are on the same profile as the Cisco, so I must
>>set dtmf-type to rfc2833; it works with internal applications (like
>>voicemail) but does not work through the Cisco as it misinterprets the
>>rfc2833
>>
>>
>> Is there a way to set some variable (or a parameter to the bridge
>> application) to do the translation?
>>
>>  Thanks! __Yehavi:
>>
>> ___
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>>
>
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