Re: [Freeswitch-users] Local call uses public context?

[Lars Zeb]

Thanks for the reply, Michael.

 

I tried the digest authentication using the cidr and copying the
conf/sip_profiles/internal.xml from the distribution, where



As a result, one endpoint could not register and another was unauthorized.

 

http://pastebin.freeswitch.org/11634

 

Then I went changed the context in internal.xml from public to default and

http://192.168.0.0/24>
192.168.10.0/24"/> http://192.168.0.0/24> 192.168.10.0/24"/>

 

And the phones registered OK. So my confusion persists.

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, December 24, 2009 11:00 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Local call uses public context?

 

Lars,

Since this question has come up a few times I'm going to write up a nice
wiki article on it explaining the differences between letting someone in via
an ACL and actually doing digest authentication. In a nutshell, though, it's
this: if the user does digest authentication (with the whole REGISTER, 401,
REGISTER, 200 OK exchange) then whatever value is in user_context is the
context for the calls made by that user. In conf/directory/default/1000.xml
(and 1001.xml, etc.) they all have user_context = "default" so when those
users register the calls they make are handled in the default context. OTOH,
if you let a user in via an ACL they aren't really registered, you've simply
opened the door for anyone coming from a particular IP address or IP address
range. In that case the calls are handled in the context specified by the
context parameter of the sip profile where the calls come in. By default the
internal sip profile uses the public context. This is for security reasons.
"Paranoid by default" is how you might describe it. You are welcome to
change that value to "default" so that calls let in by the ACL are handled
just like auth'd calls.

Play around with it and let us know how it goes. I think you'll get it once
you start modifying settings and making test calls.

-MC

On Thu, Dec 24, 2009 at 8:16 AM, Lars Zeb  wrote:

Brian,

 

Please forgive my slowness, but I'm still having problems with this. When
you say that I "really didn't auth the user", did you mean the
endpoint/extension?

 

If you did, I upped to svn1 16055 and placed a cidr attribute on the
extension and reran the test, resulting in the same output, going to context
public.

 

Further, I'm confused about your response about ACL compared with Billy W in
an email of 12/22/2009.

 

".you could simply put these entries in your internal sofia profile.

 

 

 

In that case, you do not need to include anything in the directory.  The
cidr entries in the directory are for providing additional control for each
user id and what IPs they are allowed to make calls from."

 

http://pastebin.freeswitch.org/11633

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

Thanks Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, December 23, 2009 6:03 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Local call uses public context?

 

2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved
by acl "192.168.10.0/24[]". Access Granted.

 

Because the context is set on the profile as public... and you really didn't
auth the user so user_context was never set.

 

/b

 

On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote:

 

I am trying to setup a second FS box from scratch using v16048.

 

What can cause a local call (81002, or 9996) to use context public? It's a
standard vanilla install.

 

http://pastebin.freeswitch.org/11629

 

Thanks, Lars

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Re: [Freeswitch-users] Local call uses public context?

[Lars Zeb]

Brian,

 

Please forgive my slowness, but I'm still having problems with this. When
you say that I "really didn't auth the user", did you mean the
endpoint/extension?

 

If you did, I upped to svn1 16055 and placed a cidr attribute on the
extension and reran the test, resulting in the same output, going to context
public.

 

Further, I'm confused about your response about ACL compared with Billy W in
an email of 12/22/2009.

 

".you could simply put these entries in your internal sofia profile.

 

 

 

In that case, you do not need to include anything in the directory.  The
cidr entries in the directory are for providing additional control for each
user id and what IPs they are allowed to make calls from."

 

http://pastebin.freeswitch.org/11633

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

Thanks Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, December 23, 2009 6:03 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Local call uses public context?

 

2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved
by acl "192.168.10.0/24[]". Access Granted.

 

Because the context is set on the profile as public... and you really didn't
auth the user so user_context was never set.

 

/b

 

On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote:





I am trying to setup a second FS box from scratch using v16048.

 

What can cause a local call (81002, or 9996) to use context public? It's a
standard vanilla install.

 

http://pastebin.freeswitch.org/11629

 

Thanks, Lars

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[Freeswitch-users] Local call uses public context?

[Lars Zeb]

I am trying to setup a second FS box from scratch using v16048. 

 

What can cause a local call (81002, or 9996) to use context public? It's a
standard vanilla install.

 

http://pastebin.freeswitch.org/11629

 

Thanks, Lars

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Re: [Freeswitch-users] Freeswitch not seeing Register requests

[Lars Zeb]

Mike,

 

You were right. I turned iptables off and the phone registered.

 

Thanks so much, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, December 22, 2009 8:39 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch not seeing Register requests

 

If your seeing the trafic in ngrep bit not in sip trace in Sofia when
enabled, your firewall is blocking the traffic

 

Mike 


On Dec 22, 2009, at 5:20 PM, Michael Collins  wrote:

 

On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb  wrote:

Yes, the internal profile exists.

 

 Name  Type   Data
State


=

 internal   profile   sip:mod_so...@192.168.10.25:5060
RUNNING (0)

internal-ipv6   profile   sip:mod_so...@[::1]:5060
RUNNING (0)

 external   profile   sip:mod_so...@192.168.10.25:5080
RUNNING (0)

  example.com   gatewaysip:joeu...@example.com
<mailto:sip%3ajoeu...@example.com>   NOREG

192.168.10.25 alias   internal
ALIASED


=

3 profiles 1 alias

 

 

I would do a sanity check at this point: put this box and one phone on a
completely separate network with nothing else and see what happens. 
-MC

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Re: [Freeswitch-users] Freeswitch not seeing Register requests

[Lars Zeb]

Yes, the internal profile exists.

 

 Name  Type   Data
State


=

 internal   profile   sip:mod_so...@192.168.10.25:5060
RUNNING (0)

internal-ipv6   profile   sip:mod_so...@[::1]:5060
RUNNING (0)

 external   profile   sip:mod_so...@192.168.10.25:5080
RUNNING (0)

  example.com   gatewaysip:joeu...@example.com
NOREG

192.168.10.25 alias   internal
ALIASED


=

3 profiles 1 alias

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, December 22, 2009 11:15 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch not seeing Register requests

 

 

On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall  wrote:

I have set up a second FreeSWITCH box on the same LAN. I have v16018
installed on it and have changed nothing.

 

I configured a Polycom phone to register one of its four lines to this
second box, but it does not register. When looking at the console, there is
no activity. However, there is SIP activity on the box which I have captured
via ngrep. It looks like the phone is sending out REGISTER requests but
there is no response. The request on the pastebin repeats forever, with only
the timestamp varying.

On the new box do "sofia status" - does the internal profile exist? 

 

Is the problem that there are two FreeSWITCHes? Any suggestions on how I can
make it work?

 

On the original and the new box in vars.xml
"external_sip_ip=stun:stun.freeswitch.org"

On the original box in vars.xml "external_sip_port=5090" but in the new it
is 5080.

 

Do I need to hardcode the external_sip_ip addresses in both boxes?

 

http://pastebin.freeswitch.org/11600

 

Thanks Lars

 

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Re: [Freeswitch-users] Authenticating end points by IP

[Lars Zeb]

Bill,

Thanks for your ACL Overview. Perhaps you can help me understand more
clearly.

If you include the "local-network-acl" and "apply-inbound-acl" params in the
sip_profiles and setup the list for "localnet.auto" in acl.conf.xml, does
this mean you do not have to include the cidr attribute for individual
extensions in the directory/default folder?

Is "apply-inbound-acl" supposed to exist in both internal and external
profiles while "apply-inbound-acl" is only in the internal?

Thanks, Lars

> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-
> boun...@lists.freeswitch.org] On Behalf Of Bill W
> Sent: Monday, December 21, 2009 9:03 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Authenticating end points by IP
> 
> I recently added an overview to this wiki page to help make things more
> clear as to which ACL you need for different purposes.
> 
> http://wiki.freeswitch.org/wiki/ACL#Overview
> 
> Thanks,
> Bill W.
> 
> 
> Mathieu Rene wrote:
> > Check out: http://wiki.freeswitch.org/wiki/ACL#Users
> >
> > It'll automatically add users with a cidr= attribute to the ACL list.
> > This way you can set channel variables in the users and use them through
> > your dialplan, all authenticated by ip address.
> >
> > Cheers,
> >
> > Mathieu Rene
> > Avant-Garde Solutions Inc
> > Office: + 1 (514) 664-1044 x100
> > Cell: +1 (514) 664-1044 x200
> > mr...@avgs.ca 
> 
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Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP

[Lars Zeb]

Can you copy the address of the pastebin so that people can see it? After
you hit the Send button, the address is posted back at the top of your
browser, like:

 

http://pastebin.freeswitch.org/11441

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jerry
Richards
Sent: Tuesday, December 08, 2009 12:35 PM
To: 'Michael Jerris'; freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
UNREACHABLE When Gateway Sends RTP

 

Anthony and Michael,

 

I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs
that Anthony told me to turn on.  I put the results up in the PasteBin.

 

Best Regards,

Jerry

 

  _  

From: Jerry Richards [mailto:jerry.richa...@teotech.com] 
Sent: Monday, December 07, 2009 10:49 AM
To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
UNREACHABLE When Gateway Sends RTP

When I got the latest trunk the make gets an error.  Should I perhaps
disable the mod_amr?

 

making all mod_amr

make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'.
Stop

 

The method I used to get the latest trunk follows:

 

svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch

 

Best Regards,

Jerry

 

  _  

From: Jerry Richards [mailto:jerry.richa...@teotech.com] 
Sent: Monday, December 07, 2009 7:44 AM
To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
UNREACHABLE When Gateway Sends RTP

I am changing the 3pcc setting because one of my gateways sends INVITEs
without SDP.  I will try to update to the latest trunk today and capture
traces as Anthony described.  If I can't do it today, it might be at the end
of the week.

 

Best Regards,

Jerry

 

 

  _  

From: Michael Jerris [mailto:m...@jerris.com] 
Sent: Saturday, December 05, 2009 7:30 PM
To: Jerry Richards
Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION
UNREACHABLE When Gateway Sends RTP

Jerry- 

 

Any update on this?

 

Mike

 

On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote:





Why are you changing the 3pcc setting, is this an invite with no sdp?
you need to take a trace from FS.

1) update to latest trunk first so line number match up.
2) issue these commands

sofia profile internal siptrace on
console loglevel debug

save the output and put it on pastebin http://pastebin.freeswitch.org
 





On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards 
wrote:


I have  Mediant 1000 gateway, and for some reason, when I make an outbound
call, FS enters the CS_CONSUME_MEDIA state and never connects the call.  A
Wireshark trace shows that FS is replying to the gateway's inbound RTP
packets with ICMP DESTINATION UNREACHABLE.  But the gateway is sending RTP
packets to the same port that FS specified in the outbound INVITE.  It
appears in the log that FS is discarding the 200 OK from the gateway.

I disabled the Firewall and SELinux on the Freeswitch machine.  I tried
changing  to "true" and also "proxy", but it has no effect.

Anyone know what could be the issue?  I posted the Freeswitch log in the
pastebin.

Best Regards,
Jerry


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Re: [Freeswitch-users] Trapping dtmf on bridged call

[Lars Zeb]

It can. I use it like:

 

   session:execute("bind_meta_app", "1 b s execute_extension::dx XML
features");

   session:execute("bind_meta_app", "2 b s
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft
ime(%Y-%m-%d-%H-%M-%S)}.wav");

   session:execute("bind_meta_app", "3 b s execute_extension::cf XML
features");

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: Monday, December 07, 2009 2:59 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call

 

Can this be done in an lua script?

 

Regards,

 

  _  

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: 07 December 2009 22:18
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call

 

 

On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton
 wrote:

Hi

 

Is it possible to trap on DTMF on a bridged call within an LUA script?  I've
tried setting the gateway to use inband, but no joy.  It looks like I could
use start_dtmf, but I can't see how to launch this within LUA

Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever
you want to have happen. Check it out:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app

The Local_Extension in the default.xml dialplan file has a few examples of
using this tool.
-MC

 

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Re: [Freeswitch-users] Eavesdrop error?

[Lars Zeb]

Is this reasonable given it was the only call in FreeSwitch at the time? How
can this situation be corrected in the future?

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 02, 2009 3:35 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop error?

it probably just means the uuid was not retrieved from the db when you
called the eavesdrop exten which does the lookup on the uuid for the hash
key based on what ext you hit to retrieve the most recent uuid that called
that ext.


On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb  wrote:
Sorry, svn 15753

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lars Zeb
Sent: Wednesday, December 02, 2009 2:08 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Eavesdrop error?

I tried to use eavesdrop today and it did not work. The error message in the
log is:

[ERR] mod_dptools.c:334 Usage: [all | ]

I simply dialed 881010, trying to eavesdrop on extension 1010. Is this
incorrect?

http://pastebin.freeswitch.org/11363

Thanks Lars


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Re: [Freeswitch-users] Eavesdrop error?

[Lars Zeb]

Sorry, svn 15753

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lars Zeb
Sent: Wednesday, December 02, 2009 2:08 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Eavesdrop error?

I tried to use eavesdrop today and it did not work. The error message in the
log is:

[ERR] mod_dptools.c:334 Usage: [all | ]

I simply dialed 881010, trying to eavesdrop on extension 1010. Is this
incorrect?

http://pastebin.freeswitch.org/11363

Thanks Lars


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[Freeswitch-users] Eavesdrop error?

[Lars Zeb]

I tried to use eavesdrop today and it did not work. The error message in the
log is:

[ERR] mod_dptools.c:334 Usage: [all | ]

I simply dialed 881010, trying to eavesdrop on extension 1010. Is this
incorrect?

http://pastebin.freeswitch.org/11363

Thanks Lars


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Re: [Freeswitch-users] Copy voicemail greeting

[Lars Zeb]

What tool/GUI do you use to edit the db contents?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, November 04, 2009 12:25 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Copy voicemail greeting

 

copy the wav file and insert the record.

 

/b

 

On Nov 4, 2009, at 2:20 PM, Lars Zeb wrote:





Is it possible to copy an existing wav greeting from one extension to
another? I think something has to be added to db/voicemail_default.db, but
it's not a text file.

 

Is it just easier to re-record the message from the 2nd extension?

 

Thanks, Lars

 

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[Freeswitch-users] Copy voicemail greeting

[Lars Zeb]

Is it possible to copy an existing wav greeting from one extension to
another? I think something has to be added to db/voicemail_default.db, but
it's not a text file.

 

Is it just easier to re-record the message from the 2nd extension?

 

Thanks, Lars

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Re: [Freeswitch-users] Error 488- Incompatible Destination

[Lars Zeb]

I meant "suddenly" after I had been messing with the SNOM's settings. The
firmware is 7.3.14. The SRTP stuff is:

 

Symmetrical RTP:   off 

RTP Encryption:on 

Dynamic G.726 payload: on 

SRTP Auth-tag: AES-32 

RTP/SAVP: off

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, October 29, 2009 4:37 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error 488- Incompatible Destination

 

No its not ALL of the sudden... Please check the SRTP settings on the
identity.  Thats why you're getting that... the phone defaults to sending
crypto in AVP which is invalid... we reject it.

 

This has been this way for some time I suspect you have updated your snom
recently or aren't on 7.1.35 or higher which has the options to fix this on
the phone.  set it to SAVP

 

RTP/SAVP: to off will fix it. Unless you're doing SRTP then set it to
mandatory.

 

/b

 

On Oct 29, 2009, at 6:20 PM, Lars Zeb wrote:





I must have been playing with the settings on a SNOM 320 because suddenly it
cannot connect to FreeSwitch.

 

I have looked at similar posts and it seems like it must be a codec problem
on the SNOM, but I cannot figure it out.

 

Can someone point me in the right direction?

http://pastebin.freeswitch.org/10887

 

Thanks, Lars

 

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Re: [Freeswitch-users] Problem with inbound call answered but no sound

[Lars Zeb]

Updated to v15279 and all is OK now.

 

Thanks, Brian

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 28, 2009 7:13 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

What kind of router are you behind?

 

/b

 

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:





Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

 

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[Freeswitch-users] Error 488- Incompatible Destination

[Lars Zeb]

I must have been playing with the settings on a SNOM 320 because suddenly it
cannot connect to FreeSwitch. 

 

I have looked at similar posts and it seems like it must be a codec problem
on the SNOM, but I cannot figure it out. 

 

Can someone point me in the right direction?

http://pastebin.freeswitch.org/10887

 

Thanks, Lars

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Re: [Freeswitch-users] Problem with inbound call answered but no sound

[Lars Zeb]

BroadXent ADSL 8120 è Netscreen 5XP

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 28, 2009 7:13 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

What kind of router are you behind?

 

/b

 

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:





Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

 

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[Freeswitch-users] Forcing endpoint registration

[Lars Zeb]

Is the following the correct command to force registration of an endpoint
below which is not showing up in 'sofia status profile internal' currently,
but when it does it looks like the following?

 

Call-ID:mjkg8c3zlqwjzfd.hrt5

User:   1...@192.168.10.29

Contact:"user" 

Agent:  snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1)

Status: Registered(UDP)(unknown) EXP(2009-10-27 10:46:07)

Host:   fs

IP: 192.168.10.103

Port:   5060

Auth-User:  1019

Auth-Realm: 192.168.10.29

MWI-Account:1...@192.168.10.29

 

sofia profile internal  flush_inbound_reg  
1...@192.168.10.29

 

Thanks Lars

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Re: [Freeswitch-users] Problem with inbound call answered but no sound

[Lars Zeb]

Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, October 27, 2009 8:26 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

The issue is you're not behind nat-pmp or upnp so it can't figure out what
your public IP is... you'll have to fill that in manually or enable those
options on your router if possible.

 

/b

 

On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:





I haven't changed anything since v15183, where it worked OK.

 

In conf/sip_profiles/external:

 

 

 

 

And

 

 

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Re: [Freeswitch-users] Problem with inbound call answered but no sound

[Lars Zeb]

I haven't changed anything since v15183, where it worked OK.

 

In conf/sip_profiles/external:

 

 

 

 

And

 

 

Nameexternal

Domain Name N/A

DBName  sofia_reg_external

Pres Hosts

DialplanXML

Context public

Challenge Realm auto_to

RTP-IP  192.168.10.29

Ext-RTP-IP  192.168.10.29

SIP-IP  192.168.10.29

Ext-SIP-IP  192.168.10.29

URL sip:mod_so...@192.168.10.29:5090

BIND-URLsip:mod_so...@192.168.10.29:5090

HOLD-MUSIC  local_stream://moh

OUTBOUND-PROXY  N/A

CODECS  PCMU,PCMA,GSM

TEL-EVENT   101

DTMF-MODE   rfc2833

CNG 13

SESSION-TO  0

MAX-DIALOG  0

NOMEDIA false

LATE-NEGfalse

PROXY-MEDIA false

AGGRESSIVENAT   false

STUN-ENABLEDtrue

STUN-AUTO-DISABLE   false

CALLS-IN1

FAILED-CALLS-IN 1

CALLS-OUT   0

FAILED-CALLS-OUT0

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, October 26, 2009 7:46 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip correctly?

 

/b

 

On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote:





I have tried to update (make current) twice since 15183. All inbound calls
are picked up but the caller hears nothing but a couple of clicks. The most
recent version I've tried is 15241.

 

Any ideas on what may be causing this?

 

http://pastebin.freeswitch.org/10843

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

Thanks Lars

 

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[Freeswitch-users] Problem with inbound call answered but no sound

[Lars Zeb]

I have tried to update (make current) twice since 15183. All inbound calls
are picked up but the caller hears nothing but a couple of clicks. The most
recent version I've tried is 15241.

 

Any ideas on what may be causing this? 

 

http://pastebin.freeswitch.org/10843

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

Thanks Lars

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[Freeswitch-users] Setup advice on small LAN

[Lars Zeb]

I am currently running FreeSwitch very successfully, thanks to the help from
many on this list. I am new to Linux so it was a challenge.

 

FS runs on a small LAN with about 5 other computers. The connection to the
internet is DSL with 3M down and 768kb down via Covad. The ITSP is
Flowroute.

 

If one of the computers does a big download, it messes with FS in two ways.
If a connection is made, the voices are broken up, intermittent and
difficult to understand. If the download is long enough, the connection to
Flowroute is no longer usable due to registration failure.

 

Somehow I need to isolate the FS box from the rest of the LAN, or give its
traffic precedence. Covad's suggestion was to place the FS box in the DMZ.
If I have to, I'll get another DSL line and isolate it that way.

 

Is there a write-up anywhere that might help me with this problem, or
lacking that, can anyone offer advice?

 

581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering
flowroute

582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound: unknown
event 8: 101 NAT detected

583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing expire time
to 2243 by request of proxy sip:sip.flowroute.com

 ...

1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302 Socket up
listening on 127.0.0.1:8021

1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound: unknown
event 8: 101 NAT detected

1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing expire
time to 1607 by request of proxy sip:sip.flowroute.c

1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing expire
time to 505 by request of proxy sip:sip.flowroute.co

1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing expire
time to 210 by request of proxy sip:sip.flowroute.co

1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing expire
time to 52 by request of proxy sip:sip.flowroute.com

1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing expire
time to 33 by request of proxy sip:sip.flowroute.com

1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing expire
time to 22 by request of proxy sip:sip.flowroute.com

1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing expire
time to 10 by request of proxy sip:sip.flowroute.com

1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing expire
time to 3 by request of proxy sip:sip.flowroute.com

1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering
flowroute

1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing expire
time to 2 by request of proxy sip:sip.flowroute.com

1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering
flowroute

1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing expire
time to 1 by request of proxy sip:sip.flowroute.com

1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering
flowroute

1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering
flowroute

1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing expire
time to 60 by request of proxy sip:sip.flowroute.com

1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute
Registration Failed with status Operation has no matching challenge  [904].
failure #1

 

Thanks Lars

 

 

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Re: [Freeswitch-users] Outbound sip address call fails

[Lars Zeb]

Thanks, Russell, that did it. Works perfectly!

> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-
> boun...@lists.freeswitch.org] On Behalf Of Russell Mosemann
> Sent: Sunday, October 18, 2009 5:40 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> > No, the checkbox is unchecked.
> 
> Well, check it.
> 
> --
> Russell Mosemann
> 
> 
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Re: [Freeswitch-users] Outbound sip address call fails

[Lars Zeb]

No, the checkbox is unchecked.

> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-
> boun...@lists.freeswitch.org] On Behalf Of russell.mosem...@cune.org
> Sent: Friday, October 16, 2009 10:30 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb  said:
> 
> > I tried dial sip:e...@iptel.org. In the pastebin you can see that this
> > address came over to the cli as "echo". Is it Bria which is messing up,
or
> > my FS configuration?
> 
> It looks like a Bria problem. If you open the account information and go
> to the Advanced tab, is the box checked to "Send outgoing request
> directly to target"?
> 
> 
> --
> Russell Mosemann
> 
> 
> 
> 
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
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Re: [Freeswitch-users] Outbound sip address call fails

[Lars Zeb]

Russell,

My settings in Bria are just like yours.

I captured the SIP trace from the Bria and it says it is using the
FreeSwitch as the Outbound Proxy. I am very confused.

I tried dial sip:e...@iptel.org. In the pastebin you can see that this
address came over to the cli as "echo". Is it Bria which is messing up, or
my FS configuration?

Where did Bria get the Contact info at lines 79 & 82? 192.168.10.11 is the
address of the softphone, so I don't think sip:1...@192.168.10.11 makes
sense.

http://pastebin.freeswitch.org/10723

Lars


> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-
> boun...@lists.freeswitch.org] On Behalf Of russell.mosem...@cune.org
> Sent: Thursday, October 15, 2009 10:33 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb  said:
> 
> > I don't think Bria is set to use a proxy.
> >
> > In the SIP Account/Account tab, it is set to "Register with domain and
> > receive calls" and "Send outbound via" is set to use Domain rather than
> > "Proxy Address".
> >
> > The Account ID is set to 1...@192.168.10.29, where the IP is the
> > address of FreeSwitch.
> 
> That's exactly how I have X-Lite configured, and I am able to call the
> conference directly by entering the conference URL. I'm sitting behind
> NAT, too. Under the Topology tab, it is set to "Discover global address"
> and "Discover server" (for STUN). "Enable ICE" is checked. The next thing
> to check might be if the phone traffic is going directly to the Internet
> and getting past your firewall.
> 
> --
> Russell Mosemann
> 
> 
> 
> 
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
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Re: [Freeswitch-users] Outbound sip address call fails

[Lars Zeb]

I don't think Bria is set to use a proxy.

In the SIP Account/Account tab, it is set to "Register with domain and
receive calls" and "Send outbound via" is set to use Domain rather than
"Proxy Address".

The Account ID is set to 1...@192.168.10.29, where the IP is the address of
FreeSwitch.

> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-
> boun...@lists.freeswitch.org] On Behalf Of russell.mosem...@cune.org
> Sent: Thursday, October 15, 2009 9:48 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound sip address call fails
> 
> Lars Zeb  said:
> 
> > I don't think I have a proxy. It's pretty standard installation.
> 
> Is the phone itself configured to use a proxy?
> 
> --
> Russell Mosemann
> 
> 
> 
> 
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
> 
> 
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Re: [Freeswitch-users] Outbound sip address call fails

[Lars Zeb]

I don't think I have a proxy. It's pretty standard installation.

 

 Name  Type   Data
State


=

 internal   profile   sip:mod_so...@192.168.10.29:5060
RUNNING (0)

internal-ipv6   profile   sip:mod_so...@[::1]:5060
RUNNING (0)

 external   profile   sip:mod_so...@192.168.10.29:5090
RUNNING (0)

flowroute   gatewaysip:55...@sip.flowroute.com
REGED

192.168.10.29 alias   internal
ALIASED


=

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, October 15, 2009 6:55 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound sip address call fails

 

no clue it depends on how you have it configured.  If you have an outbound
proxy set it'll send all calls to the proxy.  If not then most likely it
will go direct to the destination.

 

/b

 

On Oct 15, 2009, at 12:00 AM, Lars Zeb wrote:





Brian,

 

Does this mean that Bria should be sending the invite directly to
sip:8...@conference.freeswitch.org and the From address should be the address
of the Bria itself rather than FS?

 

Lars

 

 

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Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

[Lars Zeb]

Thanks, Michael, using "make current" resolved the issue. I will use only
that command in the future.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb  wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 

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Re: [Freeswitch-users] Outbound sip address call fails

[Lars Zeb]

Brian,

 

Does this mean that Bria should be sending the invite directly to
sip:8...@conference.freeswitch.org and the From address should be the address
of the Bria itself rather than FS?

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 14, 2009 8:55 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound sip address call fails

 

Sounds like bria is sending the invite via the proxy which is your local FS
box and NOT direct.

 

/b

 

On Oct 14, 2009, at 4:26 PM, Lars Zeb wrote:





I am trying to make a test call to the FreeSwitch conference address,
sip:8...@conference.freeswitch.org on a Bria softphone. However, it fails
with a 404 Not found message.

 

I have struggled with this for a while. Is there something special I must do
on the Bria to make this happen correctly?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/10715

 

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[Freeswitch-users] Outbound sip address call fails

[Lars Zeb]

I am trying to make a test call to the FreeSwitch conference address,
sip:8...@conference.freeswitch.org on a Bria softphone. However, it fails
with a 404 Not found message.

 

I have struggled with this for a while. Is there something special I must do
on the Bria to make this happen correctly?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/10715

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Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

[Lars Zeb]

I did svn up, ./configure, make and make install.

 

Do you want me to do make current before proceeding, or just try to make it
stop with the current build from the above commands?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, October 14, 2009 9:02 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

By which process did you upgrade?
Did you use "make current"
if so, run freeswitch from the directory you built it from then reporduce
the problem and watch for it to crash and type

./support-d/fscore_pb

at your shell to upload a crash report to pastebin and tell us the url it
generated.



On Wed, Oct 14, 2009 at 10:28 AM, Lars Zeb  wrote:

Michael,

 

I upgraded to v15152. There is no apparent change in the behavior. When I
call into a number handled by the lua script, I may get a fast busy signal,
or a recording saying the connection cannot be made.

 

But FreeSwitch is removed from memory as a result. I cannot see from the log
that the FreeSwitch ended.

 

Is there anything I can try on my side?

 

http://pastebin.freeswitch.org/10707

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb  wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 


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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com
<mailto:msn%3aanthony_miness...@hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
<mailto:paypal%3aanthony.miness...@gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
<mailto:sip%3a...@conference.freeswitch.org> 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
<mailto:googletalk%3aconf%2b...@conference.freeswitch.org> 
pstn:213-799-1400

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Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

[Lars Zeb]

Michael,

 

I upgraded to v15152. There is no apparent change in the behavior. When I
call into a number handled by the lua script, I may get a fast busy signal,
or a recording saying the connection cannot be made.

 

But FreeSwitch is removed from memory as a result. I cannot see from the log
that the FreeSwitch ended.

 

Is there anything I can try on my side?

 

http://pastebin.freeswitch.org/10707

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

 

 

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb  wrote:

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

We sent out a notice about a bug in event socket between revs 15119 and
15134. Update to latest trunk. (FYI, the fix was in r15135, so you just
barely missed it.)

After you update, let us know if the problem persists. 
-MC

 

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[Freeswitch-users] lua script causing FreeSwitch to crash?

[Lars Zeb]

http://pastebin.freeswitch.org/10686

 

I am trying to use a lua script for inbound calls. The caller hears a busy
signal. After this call fails to go through, FreeSwitch is no longer in
memory - it does not appear in  'ps -ef' output.

 

I don't have a clue what I might be doing incorrectly in the script in order
to cause this to happen. There is no indication in the log that FreeSwitch
has terminated. But it is not resident in memory.

 

I am using v15134. I had been using this same script without incident as
late as v14929. Until I can resolve the issue, I have removed the call to
lua from the dialplan and done all I could in it instead.

 

I have included the dialplan, the script and the cli log in the pastebin. I
would appreciate any help or suggestions.

 

Thanks, Lars

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[Freeswitch-users] Upgrading causes no answer

[Lars Zeb]

http://pastebin.freeswitch.org/10612

 

I having been running v14996 OK for a while. I have upgraded a couple of
times after, but every time, an inbound call is hung up on. The only thing
that has changed is the upgrade. This morning I upgraded to v15098 and the
problem persists.

 

I believe it has to do with a lua script I use for inbound calls. Reading
from the log, just after the script is launched, the following two lines
appear:

 

switch_cpp.cpp:1116  session not ready

switch_cpp.cpp:925 destroy/unlink session from object

 

Has something changed recently with lua processing? Is there something in
the lua script which is causing the problem?

 

I would appreciate any help.

 

Thanks, Lars

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Re: [Freeswitch-users] restart when convenient

[Lars Zeb]

Try:

/usr/local/freeswitch/bin/fs_cli -x 'fsctl shutdown elegant restart'

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Christian Löschenkohl
Sent: Thursday, September 03, 2009 11:03 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] restart when convenient

hello

i'm looking for a possibility to restart freeswitch like it is possible with
asterisk.
for me i tried to created a script that looks for open channels and if no
channel
is open it restarts freeswitch with the init script (not the most efficient
way).

i think i would be great if we would have a buildin function for this, i
think such
command would help with maintenance and not only for me.

br

-- 
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenk...@xpirio.com

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[Freeswitch-users] Trouble parsing sip:8...@conference.freeswitch.org?

[Lars Zeb]

I tried to dial sip:8...@conference.freeswitch.org via a Bria softphone. Why
does the parsed regex look like:

 

mod_dialplan_xml.c:315 Processing 1009->888 in context default

 

http://pastebin.freeswitch.org/10199

 

It looks like the domain is stripped off. What am I missing?

 

Thanks, Lars

 

 

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[Freeswitch-users] Error building FreeSWITCH

[Lars Zeb]

I just updated using "svn up" which brought the source to 14741. After
running "./configure", I ran "make" and got the following output:

 

making all mod_lua

make[5]: swig: Command not found

make[5]: *** [mod_lua_wrap.cpp] Error 127

make[4]: *** [all] Error 1

make[3]: *** [mod_lua-all] Error 1

make[2]: *** [all-recursive] Error 1

Making all in build

 + FreeSWITCH Build Complete ---+

 + FreeSWITCH has been successfully built.  +

 + Install by running:  +

 +  +

 +   make install   +

 +--+

make[1]: *** [all-recursive] Error 1

make: *** [all] Error 2

 

What did I do wrong?

 

Thanks, Lars

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[Freeswitch-users] Dial phone number + extension?

[Lars Zeb]

Is there a way to dial an external 10-digit phone number, wait a second or
two after connecting, and then dial a 4-digit extension?

 

Thanks, Lars

 

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[Freeswitch-users] Problem with cnam.js?

[Lars Zeb]

I think there's something wrong with the script at
http://wiki.freeswitch.org/wiki/Examples_cnam.js.

 

If you use it as is, it displays "Content-type: text/html" for the
effective_caller_id_name. In cnam.pl, the first two output lines are
generated by:

 

if (!$debug) {print "Content-type: text/html\n\n";}

 

with the actual name in the third line.

 

So I changed:

 

fd.open("read");

buff = fd.readln();

 

if(buff) {

   logger(buff, "info");

   session.setVariable("effective_caller_id_name", buff);

}

 

To:

 

fd.open("read");

buff = fd.readAll();

 

if(buff[2]) {

   logger(buff, "info");

   session.setVariable("effective_caller_id_name", buff[2]);

}

 

Or remove the print statement from cnam.pl.

 

Sorry for the code, but the page was not editable.

 

Lars

 

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Re: [Freeswitch-users] Eavesdrop getting killed after being answered

[Lars Zeb]

Thanks, Michael, it's working at 14548.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, August 17, 2009 2:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop getting killed after being
answered

 

 

On Mon, Aug 17, 2009 at 1:18 PM, Lars Zeb  wrote:

I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong? 

 

I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.

 

http://pastebin.freeswitch.org/10025

 

Hmm, something isn't right. Here's a snippet from a successful eavesdrop:
http://pastebin.freeswitch.org/10027

As you can see, your log shows no media bug operation like mine does.

Can you do another "make current" just to make 100% certain that your build
environment isn't corrupted?
-MC
 

Thanks Lars


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[Freeswitch-users] Eavesdrop getting killed after being answered

[Lars Zeb]

I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong? 

 

I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.

 

http://pastebin.freeswitch.org/10025

 

Thanks Lars

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Re: [Freeswitch-users] Customized voicemail greeting

[Lars Zeb]

OK, sorry, I will try. Thanks for your help, Brian and Mike and Saeed.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, July 31, 2009 1:18 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Customized voicemail greeting

 

NO call and check your voicemail.. go thru the options and record your
greeting

 

/b

 

On Jul 31, 2009, at 3:12 PM, Lars Zeb wrote:





Mike, I believe what you're thinking about is someone calling and leaving a
message. I want to be able to play an outbound greeting message in the
extension owner's voice, like "Hi, this is Lars. Please leave me a message".

 

Thanks, Lars

 

 

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Re: [Freeswitch-users] Customized voicemail greeting

[Lars Zeb]

Mike, I believe what you're thinking about is someone calling and leaving a
message. I want to be able to play an outbound greeting message in the
extension owner's voice, like "Hi, this is Lars. Please leave me a message".

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Friday, July 31, 2009 12:36 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Customized voicemail greeting

 

individual users can create their own greetings by calling it and pressing
the options to record it.  Is this what you are talking about?

 

MIke

 

On Jul 31, 2009, at 12:27 PM, Lars Zeb wrote:





Is it possible to define a custom voicemail greeting (wav file) on an
extension-by-extension basis?

 

I have read the docs on mod_voicemail and searched previous emails, but I
don't know where to begin.

 

Thanks, Lars

 

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[Freeswitch-users] Customized voicemail greeting

[Lars Zeb]

Is it possible to define a custom voicemail greeting (wav file) on an
extension-by-extension basis?

 

I have read the docs on mod_voicemail and searched previous emails, but I
don't know where to begin.

 

Thanks, Lars

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Re: [Freeswitch-users] Asterisk key during message hangs up call

[Lars Zeb]

Your message made me look at the documentation, which was helpful.

 

http://pastebin.freeswitch.org/9838

 

When I press *, I get a busy signal.

 

Please disregard the USER_NOT_REGISTERED error in the log; one of the
endpoints I bridged to is off-line.

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Thursday, July 23, 2009 7:41 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call

 

I meant to pick one based on whichever lang you were using not to literally
write what i said.

anyway, yes so now you solved your autoHangup
make a new debug trace like the one you looked at before now which should be
different.





On Thu, Jul 23, 2009 at 9:23 AM, Lars Zeb  wrote:

Thanks for the reply. This is my first attempt at using a script.

 

I tried:

 

  session:autoHangup(0) or session:autoHangup(false)

 

but got an error:

 

2009-07-23 07:19:09.652238 [ERR] mod_lua.cpp:182
/usr/local/freeswitch/scripts/inbound_calls.lua:29: attempt to call method
'autoHangup' (a nil value)

stack traceback:

/usr/local/freeswitch/scripts/inbound_calls.lua:29: in main chunk

 

I looked at the documentation and tried:

 

  session:setAutoHangup(false)

 

and the script proceeded without error. However, looking at the log, I do
not see the setAutoHangup being called. Also, when pressing *, I get a fast,
busy signal.

 

I have pasted the script and log at http://pastebin.freeswitch.org/9836

 

Thanks again, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, July 22, 2009 4:33 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call

 

you are using a channel created with a script and you did not set 

js
session.autoHangup(0)

lua
session:autoHangup(0)

so when the * makes the call transfer the script kills the channel.

On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb  wrote:

Brian,

 

When calling into FreeSWITCH and pressing * during the greeting, the call
immediately hangs up.

 

It used to ask for the mailbox number to retrieve messages. It no longer
works. I don't know if my dialplan is causing the error or something in
FreeSWITCH has changed.

 

Any ideas? 

 

http://pastebin.freeswitch.org/9803

 

Thanks, Lars

 


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Re: [Freeswitch-users] Asterisk key during message hangs up call

[Lars Zeb]

Thanks for the reply. This is my first attempt at using a script.

 

I tried:

 

  session:autoHangup(0) or session:autoHangup(false)

 

but got an error:

 

2009-07-23 07:19:09.652238 [ERR] mod_lua.cpp:182
/usr/local/freeswitch/scripts/inbound_calls.lua:29: attempt to call method
'autoHangup' (a nil value)

stack traceback:

/usr/local/freeswitch/scripts/inbound_calls.lua:29: in main chunk

 

I looked at the documentation and tried:

 

  session:setAutoHangup(false)

 

and the script proceeded without error. However, looking at the log, I do
not see the setAutoHangup being called. Also, when pressing *, I get a fast,
busy signal.

 

I have pasted the script and log at http://pastebin.freeswitch.org/9836

 

Thanks again, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, July 22, 2009 4:33 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call

 

you are using a channel created with a script and you did not set 

js
session.autoHangup(0)

lua
session:autoHangup(0)

so when the * makes the call transfer the script kills the channel.



On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb  wrote:

Brian,

 

When calling into FreeSWITCH and pressing * during the greeting, the call
immediately hangs up.

 

It used to ask for the mailbox number to retrieve messages. It no longer
works. I don't know if my dialplan is causing the error or something in
FreeSWITCH has changed.

 

Any ideas? 

 

http://pastebin.freeswitch.org/9803

 

Thanks, Lars

 


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[Freeswitch-users] Asterisk key during message hangs up call

[Lars Zeb]

Brian,

 

When calling into FreeSWITCH and pressing * during the greeting, the call
immediately hangs up.

 

It used to ask for the mailbox number to retrieve messages. It no longer
works. I don't know if my dialplan is causing the error or something in
FreeSWITCH has changed.

 

Any ideas? 

 

http://pastebin.freeswitch.org/9803

 

Thanks, Lars

 

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Re: [Freeswitch-users] Inbound call routing help

[Lars Zeb]

Brian,

 

Pressing * no longer works. I don't know if my dialplan is causing the error
or something in FreeSWITCH has changed.

 

When pressing * during the greeting, the call immediately hangs up.

 

Any ideas? 

 

http://pastebin.freeswitch.org/9803

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, May 21, 2009 5:01 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inbound call routing help

 

Try pressing * during the greeting and make sure you have the vmain
extension so you can login.

 

/b

 

On May 21, 2009, at 6:56 PM, Lars Zeb wrote:





I want to setup a dialplan for a single DID. I would like it to go to a
specific extension, and if not picked up in 15 seconds, go to voicemail.

 

I have set this scenario up and it works. But I would also like this person
to be able to call this DID from outside FS via a phone and be able to
retrieve their voicemail. I've seen the example of how to pick up an
extension's voicemail while inside FS by checking to see if the
destination_number is the same as the caller_id_number, and if so, listen to
voicemail, otherwise leave the message with voicemail.

 

But I don't have a clue how to accomplish this from outside, other than
dedicating another DID to solely retrieving voicemail from outside.

 

Any ideas?

 

Thanks, Lars

__

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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Re: [Freeswitch-users] Error in lua script with session:getVariable

[Lars Zeb]

Mathieu,

 

Thanks for the reply. I'm very new with FreeSWITCH and not familiar with api
calls. I tried:

 

 user_data = apiExecute("user_data", "1...@192.168.10.29 var
callgroup");

 

But got a similar error:

 

2009-07-16 17:01:47.212904 [ERR] mod_lua.cpp:182
/usr/local/freeswitch/scripts/helloworld.lua:13: attempt to call global
'apiExecute' (a nil value)

stack traceback:

/usr/local/freeswitch/scripts/helloworld.lua:13: in main chunk

 

I would appreciate an example or a link to the pertinent documentation.

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Mathieu
Rene
Sent: Thursday, July 16, 2009 2:42 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in lua script with session:getVariable

 

You need to make an API call, not get a variable.

 

Mathieu Rene

Avant-Garde Solutions Inc

Office: + 1 (514) 664-1044 x100

Cell: +1 (514) 664-1044 x200

mr...@avgs.ca

 

 

 

 

Am 16-Jul-09 um 5:36 PM schrieb Lars Zeb:





I am getting an error in a lua script which I don't understand. Why is it
returning nil in the script yet something in the cli?

 

lua snippet:

user_data = session:getVariable('user_data 1...@192.168.10.29 var
callgroup');

freeswitch.console_log("INFO", " UserData group " .. user_data .. "\n")

 

log:

2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182
/usr/local/freeswitch/scripts/helloworld.lua:14: attempt to concatenate
global 'user_data' (a nil value)

stack traceback:

/usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk

 

 

cli:

freeswi...@internal> user_data 1...@192.168.10.29 var callgroup

techsupport

 

Thanks, Lars

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[Freeswitch-users] Error in lua script with session:getVariable

[Lars Zeb]

I am getting an error in a lua script which I don't understand. Why is it
returning nil in the script yet something in the cli?

 

lua snippet:

user_data = session:getVariable('user_data 1...@192.168.10.29 var
callgroup');

freeswitch.console_log("INFO", " UserData group " .. user_data .. "\n")

 

log:

2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182
/usr/local/freeswitch/scripts/helloworld.lua:14: attempt to concatenate
global 'user_data' (a nil value)

stack traceback:

/usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk

 

 

cli:

freeswi...@internal> user_data 1...@192.168.10.29 var callgroup

techsupport

 

Thanks, Lars

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[Freeswitch-users] Dialplan with lua - error missing closing angle bracket?

[Lars Zeb]

I copied the action element from http://wiki.freeswitch.org/wiki/Lua,
"Sample Dialplan". 

 

When I try to reloadxml, the cli tells me that there is a missing right
angle bracket. 

+OK [[error near line 3130]: missing >]

 

Do the docs need updating or have I totally blown it?

 

Thanks, Lars

 



   

 

   

 

 

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[Freeswitch-users] contrib directory location

[Lars Zeb]

What is the address of the contrib directory? I would like to download it
and its contents for study.

 

Thanks, Lars

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[Freeswitch-users] fs_cli - display variable values?

[Lars Zeb]

Is it possible to display the value of a variable in fs_cli? I tried "echo
${domain_name}", but it just echoed what I typed (${domain_name}), rather
than its value. 

 

I do not know how to get help on an individual command from the help
facility in fs_cli. I tried fs_cli itself and also the docs but could find
nothing.

 

Thanks, Lars

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Re: [Freeswitch-users] Intercom error with SNOM

[Lars Zeb]

Thanks Michael and Brian and Andrew and Peder. I changed "Enable Intercom"
to off and "'Answer After' Policy" to "only in idle" and it works. The next
firmware will probably change the settings to mean what they should mean.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, July 14, 2009 12:59 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intercom error with SNOM

 

Do what Andrew said.. it has to be a bug :P (in the snom)

 

/b

 

On Jul 14, 2009, at 2:51 PM, Lars Zeb wrote:





2. Do you mean setting "Challenge Response on phone" on the SNOM? It is
already set to no.

3. 7.3.14

 

 

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Re: [Freeswitch-users] Intercom error with SNOM

[Lars Zeb]

2. Do you mean setting "Challenge Response on phone" on the SNOM? It is
already set to no.

3. 7.3.14

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, July 14, 2009 12:41 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intercom error with SNOM

 

Make sure you 1. reboot. 2. make sure the setting is correct to not auth. 3.
what firmware are you on?

 

/b

 

On Jul 14, 2009, at 2:34 PM, Lars Zeb wrote:





Yes.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, July 14, 2009 10:58 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intercom error with SNOM

 

Lars,

Brian pointed out that the challenge is coming from the phone. Is
192.168.10.104 the Snom?

-MC

 

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Re: [Freeswitch-users] Intercom error with SNOM

[Lars Zeb]

Yes.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, July 14, 2009 10:58 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intercom error with SNOM

 

Lars,

Brian pointed out that the challenge is coming from the phone. Is
192.168.10.104 the Snom?

-MC

On Tue, Jul 14, 2009 at 10:46 AM, Lars Zeb  wrote:

Michael,

 

I made the changes you suggested, but the result is the same. If it matters,
I am at 14229. I stopped and restarted FreeSWITCH and then reran the
intercom call.

 

Lars

 

 

   

 

 

   

   

   

 

 

2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by
acl "domains". Falling back to Digest auth.

2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by
acl "domains". Falling back to Digest auth.

2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0]

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, July 14, 2009 10:10 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intercom error with SNOM

 

 

On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb  wrote:

I am getting an error when I try to make an intercom call from a softphone
to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching
gateway found" in the log. I can successfully make an intercom call between
my softphone and a Polycom 501, so it must be something with the SNOM.

 

bkw suggested that the problem was in the challenge/response between
FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set
"Challenge/Response" off,  "Enable intercom" on and "Type of Intercom
Answering" to "Handsfree". The error is still "401 Unauthorized".

 

What more do I need to do?


Lars,
It looks like FreeSWITCH is sending the challenge to the Snom. Note this
line from the debug log:
2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by
acl "domains". Falling back to Digest auth.

For testing purposes you can edit acl.conf.xml and add a new line right
after:


Add:
 

Restart FS or issue this command at the CLI:
reloadacl reloadxml

Then try your call again.

To learn more about how you can have your local users bypass the "domains"
acl without editing acl.conf.xml then look in
conf/directory/default/brian.xml. At the top of that file you will see a
note about how adding a cidr= attribute to your user tag will let you bypass
the domains ACL check.

Enjoy!
-MC

 

Thanks, Lars

 

http://pastebin.freeswitch.org/9709

 


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Re: [Freeswitch-users] Intercom error with SNOM

[Lars Zeb]

Michael,

 

I made the changes you suggested, but the result is the same. If it matters,
I am at 14229. I stopped and restarted FreeSWITCH and then reran the
intercom call.

 

Lars

 

 

   

 

 

   

   

   

 

 

2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by
acl "domains". Falling back to Digest auth.

2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by
acl "domains". Falling back to Digest auth.

2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0]

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, July 14, 2009 10:10 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intercom error with SNOM

 

 

On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb  wrote:

I am getting an error when I try to make an intercom call from a softphone
to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching
gateway found" in the log. I can successfully make an intercom call between
my softphone and a Polycom 501, so it must be something with the SNOM.

 

bkw suggested that the problem was in the challenge/response between
FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set
"Challenge/Response" off,  "Enable intercom" on and "Type of Intercom
Answering" to "Handsfree". The error is still "401 Unauthorized".

 

What more do I need to do?


Lars,
It looks like FreeSWITCH is sending the challenge to the Snom. Note this
line from the debug log:
2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by
acl "domains". Falling back to Digest auth.

For testing purposes you can edit acl.conf.xml and add a new line right
after:


Add:
 

Restart FS or issue this command at the CLI:
reloadacl reloadxml

Then try your call again.

To learn more about how you can have your local users bypass the "domains"
acl without editing acl.conf.xml then look in
conf/directory/default/brian.xml. At the top of that file you will see a
note about how adding a cidr= attribute to your user tag will let you bypass
the domains ACL check.

Enjoy!
-MC

 

Thanks, Lars

 

http://pastebin.freeswitch.org/9709

 


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[Freeswitch-users] Intercom error with SNOM

[Lars Zeb]

I am getting an error when I try to make an intercom call from a softphone
to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching
gateway found" in the log. I can successfully make an intercom call between
my softphone and a Polycom 501, so it must be something with the SNOM.

 

bkw suggested that the problem was in the challenge/response between
FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set
"Challenge/Response" off,  "Enable intercom" on and "Type of Intercom
Answering" to "Handsfree". The error is still "401 Unauthorized".

 

What more do I need to do?

 

Thanks, Lars

 

http://pastebin.freeswitch.org/9709

 

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Re: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error

[Lars Zeb]

Michael,

 

The extension-intercom is from the conf/dialplan/default.xml. I checked the
file in the source tree, and it's the same as I originally used.

 

But I did try your suggestion: . The result was the same after reloadxml.

 

To check if it had been reloaded, I opened conf/freeswitch.xml. I was
surprised to see only 64 lines in the file. Something is hosed in my
configuration. Should I try to rebuild from scratch and move over my changes
to the xml files?

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, July 10, 2009 11:27 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Intercom call failing with "No Matching
gateway found" error

 

Lars,

If I read your dialplan correctly I believe this line is a problem:


Try this:


Let us know if that works...

-MC

On Fri, Jul 10, 2009 at 10:35 AM, Lars Zeb  wrote:

Trying to make an intercom call (8+extension#) gives me an error. I don't
know what I've done wrong, but I think it used to work. I am on Centos 5
with 14196M.

 

Can someone point me in the right direction? The sofia status, dialplan and
log are in http://pastebin.freeswitch.org/9681.

 

Thanks, Lars

 


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[Freeswitch-users] Intercom call failing with "No Matching gateway found" error

[Lars Zeb]

Trying to make an intercom call (8+extension#) gives me an error. I don't
know what I've done wrong, but I think it used to work. I am on Centos 5
with 14196M.

 

Can someone point me in the right direction? The sofia status, dialplan and
log are in http://pastebin.freeswitch.org/9681.

 

Thanks, Lars

 

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Re: [Freeswitch-users] Documentation error?

[Lars Zeb]

Done.

 

Not knowing much, I'm reluctant to make changes without asking.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, July 09, 2009 7:12 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Documentation error?

 

If it makes better sense to you then please login and fix it!  ;)

 

/b

 

On Jul 9, 2009, at 9:08 AM, Lars Zeb wrote:





In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line after
CLI, it states:

 

 which would the options from your config file:

 

Should this be?:

 

 which would use the options from your config file:

 

Thanks, Lars

 

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[Freeswitch-users] Documentation error?

[Lars Zeb]

In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line after
CLI, it states:

 

 which would the options from your config file:

 

Should this be?:

 

 which would use the options from your config file: 

 

Thanks, Lars

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Re: [Freeswitch-users] Error in my dialplan

[Lars Zeb]

Still lost. What is the solution? 1) Remove the ability to login without
password (and the comparison between destination_number and
${caller_id_number}, 2) Create a condition which strips the + sign and
creates a new variable like caller_id_number, or 3) ???

> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-
> users-boun...@lists.freeswitch.org] On Behalf Of Brian West
> Sent: Wednesday, July 08, 2009 2:51 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Error in my dialplan
> 
> Ah yes this line 
> 
> We have since removed that ability to login without password from the
> default configs.
> 
> /b
> 
> On Jul 8, 2009, at 4:48 PM, Rupa Schomaker wrote:
> 
> > the problem is the + is coming from the network...
> >
> > On Wed, Jul 8, 2009 at 4:46 PM, Brian West 
> > wrote:
> > You have to escape the + with \+
> >
> > /b
> >
> > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote:
> >
> >> 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR:
> >> 1 [nothing to repeat][^+13105551212$]
> >
> >
> > ___
> > Freeswitch-users mailing list
> > Freeswitch-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > -Rupa
> > ___
> > Freeswitch-users mailing list
> > Freeswitch-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > http://www.freeswitch.org
> 
> 
> ___
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[Freeswitch-users] Error in my dialplan

[Lars Zeb]

I receive an error on an inbound call from my dialplan. I don't have a clue
what it means. Can someone help?

 

from log:

2009-07-08 09:54:50.172590 [DEBUG] switch_core_state_machine.c:78
sofia/external/+13105551...@66.53.188.187 Standard ROUTING

2009-07-08 09:54:50.172590 [INFO] mod_dialplan_xml.c:310 Processing
+13105551212->1000 in context default



Dialplan: sofia/external/+13105551...@66.53.188.187 parsing
[default->Local_Extension_Lars] continue=false

Dialplan: sofia/external/+13105551...@66.53.188.187 Regex (PASS)
[Local_Extension_Lars] destination_number(1000) =~ /^(100[0-9])$/
break=on-false

Dialplan: sofia/external/+13105551...@66.53.188.187 Action
set(dialed_ext=1000)

Dialplan: sofia/external/+13105551...@66.53.188.187 Action
export(dialed_ext=1000)

2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: 1
[nothing to repeat][^+13105551212$]

Dialplan: sofia/external/+13105551...@66.53.188.187 Regex (FAIL)
[Local_Extension_Lars] destination_number(1000) =~ /^+13105551212$/
break=on-false

Dialplan: sofia/external/+13104647...@66.53.188.187 ANTI-Action
bind_meta_app(1 b s execute_extension::dx XML features)

Dialplan: sofia/external/+13104647...@66.53.188.187 ANTI-Action
bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft
ime(%Y-%m-%d-%H-%M-%S)}.wav)



Caller-Dialplan: [XML]

Caller-Caller-ID-Name: [+13105551212]

Caller-Caller-ID-Number: [+13105551212]

 

My dialplan:

 

   

 

 

   

   

 

 

 

 

 

 

 ...

 

 

 

 

 

 

 



 

 

Thanks, Lars

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[Freeswitch-users] Can't understand documentation

[Lars Zeb]

On http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session, at
the bottom of "Activating via DTMF", it states:

 

The other party doesn't hear the DTMFs but maybe its comfort noisy is
disappearing for a very short time. When that sip client *starts* a call the
above dialplan forbids activating recording.

 

Can someone explain what these two sentences mean?

 

Thanks, Lars

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Re: [Freeswitch-users] Polycom configuration problems?

[Lars Zeb]

Thanks to Rupa and Chris for this help. I didn't know enough to understand
Chris was pointing me to the Polycom phone rather than FS. I would never
have figured this out.

 

Are Polycoms the only SIP phones which have this feature?

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 23, 2009 10:46 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

Ok, most of us configure the polycoms via a provisioning interface.  usually
ftp or http.

Anyway, when using the web interface, you want to look at:

Goto the web interface, Click on SIP.

Scroll down to the Local Settings section and you need to modify digitmap
and digitmap timeout.  the syntax is in the polycom manuals which you can
donwload from polycom.

On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb  wrote:

Via a web browser.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 23, 2009 8:39 AM


To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

How are you configuring your polycom?

On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb  wrote:

I'm sorry Chris, but I don't know where the look for the "global sip.cfg and
mac/phone specific cfg" settings. I also looked for digitmap but could find
nothing.

 

Can you be more specific?

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
Burns
Sent: Monday, June 22, 2009 2:57 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

Sounds like a config issue in the  tag. Check global sip.cfg and
mac/phone specific cfg. When you are dialing on-hook I don't think it will
use your .digitmap or ...digitmap.timer settings. When you dial off-hook it
sure will.

On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb  wrote:

I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines
on the phone. The first two are registered with a SwitchVox, the last with
Freeswitch.

 

When I select the 3rd line and begin to press numbers, pressing the 3rd
digit automatically causes the phone to begin to dial. It does not matter
which three numbers I press, the 3rd one is magic.

 

However, if I do not select a line before dialing and key a 10-digit number
into the phone, then select the 3rd line, it dials out fine.

 

You can see from the debug console output that Processing begins before it
hits any dialplan, so that cannot be the problem. I must have the line
defined incorrectly for Freeswitch.

 

Thanks for any suggestions, Lars.

 

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2]

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel
sofia/internal/1...@192.168.10.29 entering state [received][100]

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:

v=0

o=- 1245682011 1245682011 IN IP4 192.168.10.101

s=Polycom IP Phone

c=IN IP4 192.168.10.101

t=0 0

m=audio 2254 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

 

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403
(sofia/internal/1...@192.168.10.29) State NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:115:32000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:107:16000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G722:9:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[PCMU:0:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec
sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload
to 101

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376
(sofia/internal/1...@192.168.10.29) State Change CS_NEW -> CS_INIT

2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]

2009-06-22 12:32:18.

Re: [Freeswitch-users] Polycom configuration problems?

[Lars Zeb]

Via a web browser.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 23, 2009 8:39 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

How are you configuring your polycom?

On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb  wrote:

I'm sorry Chris, but I don't know where the look for the "global sip.cfg and
mac/phone specific cfg" settings. I also looked for digitmap but could find
nothing.

 

Can you be more specific?

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
Burns
Sent: Monday, June 22, 2009 2:57 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

Sounds like a config issue in the  tag. Check global sip.cfg and
mac/phone specific cfg. When you are dialing on-hook I don't think it will
use your .digitmap or ..digitmap.timer settings. When you dial off-hook it
sure will.

On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb  wrote:

I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines
on the phone. The first two are registered with a SwitchVox, the last with
Freeswitch.

 

When I select the 3rd line and begin to press numbers, pressing the 3rd
digit automatically causes the phone to begin to dial. It does not matter
which three numbers I press, the 3rd one is magic.

 

However, if I do not select a line before dialing and key a 10-digit number
into the phone, then select the 3rd line, it dials out fine.

 

You can see from the debug console output that Processing begins before it
hits any dialplan, so that cannot be the problem. I must have the line
defined incorrectly for Freeswitch.

 

Thanks for any suggestions, Lars.

 

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2]

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel
sofia/internal/1...@192.168.10.29 entering state [received][100]

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:

v=0

o=- 1245682011 1245682011 IN IP4 192.168.10.101

s=Polycom IP Phone

c=IN IP4 192.168.10.101

t=0 0

m=audio 2254 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

 

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403
(sofia/internal/1...@192.168.10.29) State NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:115:32000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:107:16000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G722:9:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[PCMU:0:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec
sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload
to 101

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376
(sofia/internal/1...@192.168.10.29) State Change CS_NEW -> CS_INIT

2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_INIT

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1...@192.168.10.29) State INIT

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83
sofia/internal/1...@192.168.10.29 SOFIA INIT

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111
(sofia/internal/1...@192.168.10.29) State Change CS_INIT -> CS_ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1...@192.168.10.29) State INIT going to sleep

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483
(sofia/internal/1...@192.168.10.29) State ROUTING

Re: [Freeswitch-users] Polycom configuration problems?

[Lars Zeb]

I'm sorry Chris, but I don't know where the look for the "global sip.cfg and
mac/phone specific cfg" settings. I also looked for digitmap but could find
nothing.

 

Can you be more specific?

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
Burns
Sent: Monday, June 22, 2009 2:57 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

Sounds like a config issue in the  tag. Check global sip.cfg and
mac/phone specific cfg. When you are dialing on-hook I don't think it will
use your .digitmap or .digitmap.timer settings. When you dial off-hook it
sure will.



On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb  wrote:

I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines
on the phone. The first two are registered with a SwitchVox, the last with
Freeswitch.

 

When I select the 3rd line and begin to press numbers, pressing the 3rd
digit automatically causes the phone to begin to dial. It does not matter
which three numbers I press, the 3rd one is magic.

 

However, if I do not select a line before dialing and key a 10-digit number
into the phone, then select the 3rd line, it dials out fine.

 

You can see from the debug console output that Processing begins before it
hits any dialplan, so that cannot be the problem. I must have the line
defined incorrectly for Freeswitch.

 

Thanks for any suggestions, Lars.

 

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2]

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel
sofia/internal/1...@192.168.10.29 entering state [received][100]

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:

v=0

o=- 1245682011 1245682011 IN IP4 192.168.10.101

s=Polycom IP Phone

c=IN IP4 192.168.10.101

t=0 0

m=audio 2254 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

 

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403
(sofia/internal/1...@192.168.10.29) State NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:115:32000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:107:16000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G722:9:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[PCMU:0:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec
sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload
to 101

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376
(sofia/internal/1...@192.168.10.29) State Change CS_NEW -> CS_INIT

2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_INIT

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1...@192.168.10.29) State INIT

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83
sofia/internal/1...@192.168.10.29 SOFIA INIT

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111
(sofia/internal/1...@192.168.10.29) State Change CS_INIT -> CS_ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1...@192.168.10.29) State INIT going to sleep

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483
(sofia/internal/1...@192.168.10.29) State ROUTING

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130
sofia/internal/1...@192.168.10.29 SOFIA ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1...@192.168.10.29 Standard ROUTING

2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing
1001->323 in context default

Dialplan: sofia/internal/1...@192.168.10.29 parsing [default->unloop]
co

[Freeswitch-users] Polycom configuration problems?

[Lars Zeb]

I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines
on the phone. The first two are registered with a SwitchVox, the last with
Freeswitch.

 

When I select the 3rd line and begin to press numbers, pressing the 3rd
digit automatically causes the phone to begin to dial. It does not matter
which three numbers I press, the 3rd one is magic.

 

However, if I do not select a line before dialing and key a 10-digit number
into the phone, then select the 3rd line, it dials out fine.

 

You can see from the debug console output that Processing begins before it
hits any dialplan, so that cannot be the problem. I must have the line
defined incorrectly for Freeswitch.

 

Thanks for any suggestions, Lars.

 

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2]

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel
sofia/internal/1...@192.168.10.29 entering state [received][100]

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:

v=0

o=- 1245682011 1245682011 IN IP4 192.168.10.101

s=Polycom IP Phone

c=IN IP4 192.168.10.101

t=0 0

m=audio 2254 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

 

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403
(sofia/internal/1...@192.168.10.29) State NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:115:32000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:107:16000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G722:9:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[PCMU:0:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec
sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload
to 101

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376
(sofia/internal/1...@192.168.10.29) State Change CS_NEW -> CS_INIT

2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_INIT

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1...@192.168.10.29) State INIT

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83
sofia/internal/1...@192.168.10.29 SOFIA INIT

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111
(sofia/internal/1...@192.168.10.29) State Change CS_INIT -> CS_ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1...@192.168.10.29) State INIT going to sleep

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483
(sofia/internal/1...@192.168.10.29) State ROUTING

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130
sofia/internal/1...@192.168.10.29 SOFIA ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1...@192.168.10.29 Standard ROUTING

2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing
1001->323 in context default

Dialplan: sofia/internal/1...@192.168.10.29 parsing [default->unloop]
continue=false

Dialplan: sofia/internal/1...@192.168.10.29 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false

Dialplan: sofia/internal/1...@192.168.10.29 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false

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[Freeswitch-users] event_add_head warning message on cosole

[Lars Zeb]

I just upgraded to version 13886. On the console the following messages
appear every few minutes. I've looked at the code but it's way over my head.

 

Why is it displaying? How can I turn it off? 

 

2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header ->
'toll_allow' = 'domestic,international,local'

2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header ->
'accountcode' = '1000'

2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header ->
'user_context' = 'default'

2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header ->
'effective_caller_id_name' = 'Extension 1000'

2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header ->
'effective_caller_id_number' = '1000'

2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header ->
'outbound_caller_id_name' = 'FreeSWITCH'

2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header ->
'outbound_caller_id_number' = '3235551212'

2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header ->
'callgroup' = 'techsupport'

2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header ->
'record_stereo' = 'true'

2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header ->
'default_gateway' = 'example.com'

2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header ->
'default_areacode' = '323'

2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header ->
'transfer_fallback_extension' = 'operator'

 

 

Thanks, Lars

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[Freeswitch-users] svn update error

[Lars Zeb]

I am currently running 13723 and want to get current. When I issue the "svn
update" command, the following error appears:

 

Asrc/mod/asr_tts/mod_unimrcp

Asrc/mod/asr_tts/mod_unimrcp/mod_unimrcp.2008.vcproj

Asrc/mod/asr_tts/mod_unimrcp/unimrcp.vsprops

Asrc/mod/asr_tts/mod_unimrcp/Makefile.am

Asrc/mod/asr_tts/mod_unimrcp/mod_unimrcp.c

Usrc/mod/asr_tts/mod_pocketsphinx/mod_pocketsphinx.c

Usrc/mod/event_handlers/mod_event_multicast/mod_event_multicast.c

svn: Failed to add file
'src/mod/event_handlers/mod_event_multicast/Makefile': an unversioned file
of the same name already exists

 

What do I need to do? Should I delete the Makefile file?

 

Thanks, Lars

 

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[Freeswitch-users] Some channel variables not written to cdr-csv?

[Lars Zeb]

I have defined the following template in autoload_config/cdr_csv.conf.xml:

 

"${caller_id_name}","${caller_id_number}","${destination_number}"
,"${context}","${start_stamp}","${answer_stamp}",

"${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${ble
g_uuid}","${accountcode}","${read_codec}","${write_codec}",

"${channel_name}","${bridge_channel}","${direction}"

 

 

The resultant Master.csv in logs/1000.csv:

 

"+19495551212","+19495551212","1000","default","2009-06-18
09:59:59","2009-06-18 10:00:06","2009-06-18
10:01:16","77","70","NORMAL_CLEARING","7551138e-5c29-11de-80e6-1b59605a543b"
,"75574754-5c29-11de-80e6-1b59605a543b","","PCMU","PCMU","sofia/external/+19
495551...@66.53.188.187","sofia/internal/sip:1...@192.168.10.101",""

 

Both ${direction} and ${accountcode} do not have any data in the cdr file.
Am I using the wrong variable names? I do see Caller-Direction with a valid
value ([inbound]) in freeswitch.log, but nothing like accountcode. The wiki
at http://wiki.freeswitch.org/wiki/Channel_Variables#variable_ says that
both these variables exist. 

 

Thanks for any help, Lars

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Re: [Freeswitch-users] eavesdrop extension condition in default.xml?

[Lars Zeb]

Michael,

 

The expression is part of version 13723 distribution in
conf/dialplan/default.xml. Shouldn't that be changed?

 



   

 

 

   



 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Wednesday, June 17, 2009 10:23 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] eavesdrop extension condition in
default.xml?

 

 

On Wed, Jun 17, 2009 at 12:06 PM, Lars Zeb  wrote:

In conf/dialplan/default.xml, the eavesdrop extension's condition is -
expression="^88(.*)$|^\*0(.*)$">

 

Is this intended? I thought it was defined to eavesdrop on internal
extensions. Why wouldn't it be something like -
expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial
888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop
extension.

 

Lars, 
"8885551212" should never match regex  expression "^88(\d{3,5})$" . It looks
to me like your eavesdrop is matching on "^88(.*)$" which most definitely
WILL match 8885551212. Please check your dialplan's eavesdrop extension's
regex and make sure it is correct.

-MC

Log:

1030 Dialplan: sofia/internal/1...@192..168.10.29
<mailto:sofia/internal/1...@192.168.10.29>  parsing [default->eavesdrop]
continue=false

1031 Dialplan: sofia/internal/1...@192..168.10.29
<mailto:sofia/internal/1...@192.168.10.29>  Regex (PASS) [eavesdrop]
destination_number(8885819795) =~ /^88(.*)$|^\*0(.*)$/ break=on-false

1032 Dialplan: sofia/internal/1...@192..168.10.29
<mailto:sofia/internal/1...@192.168.10.29>  Action answer()

 

Thanks, Lars

 


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[Freeswitch-users] eavesdrop extension condition in default.xml?

[Lars Zeb]

In conf/dialplan/default.xml, the eavesdrop extension's condition is -
expression="^88(.*)$|^\*0(.*)$">

 

Is this intended? I thought it was defined to eavesdrop on internal
extensions. Why wouldn't it be something like -
expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial
888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop
extension.

 

Log:

1030 Dialplan: sofia/internal/1...@192.168.10.29 parsing
[default->eavesdrop] continue=false

1031 Dialplan: sofia/internal/1...@192.168.10.29 Regex (PASS) [eavesdrop]
destination_number(8885819795) =~ /^88(.*)$|^\*0(.*)$/ break=on-false

1032 Dialplan: sofia/internal/1...@192.168.10.29 Action answer()

 

Thanks, Lars

 

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Re: [Freeswitch-users] Error in Dialplan documentation?

[Lars Zeb]

I'm sorry, but I can't find ${sip_profile} defined in any document in the
conf directory.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Saturday, June 13, 2009 1:08 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in Dialplan documentation?

 

Yes look at the default dialplan... you should note that its in the default
only.

 

/b

 

On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote:





At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, near the
top under From Dialplan, it says:

Bridge the incoming call to extension 100 and 101. The '%' is used instead
of the @ to indicate that the endpoints are registered locally. Separate
multiple endpoints with a comma. The ${sip_profile} variable is defined in
freeswitch.xml.



However, I cannot find ${sip_profile} in freeswitch.xml. Is the
documentation correct?

Thanks, Lars

 

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[Freeswitch-users] Error in Dialplan documentation?

[Lars Zeb]

At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, near the
top under From Dialplan, it says:

Bridge the incoming call to extension 100 and 101. The '%' is used instead
of the @ to indicate that the endpoints are registered locally. Separate
multiple endpoints with a comma. The ${sip_profile} variable is defined in
freeswitch.xml. 



However, I cannot find ${sip_profile} in freeswitch.xml. Is the
documentation correct?

Thanks, Lars

 

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Re: [Freeswitch-users] Unregister extension?

[Lars Zeb]

No, it’s not too harsh, João, but I hope not all of my questions were
answered on the wiki.

 

I do try to go to the wiki first. I think that my total ignorance of the
environment makes it difficult for me to do a search on the wiki or Google.
I did try before asking this list. My query to Google was “Freeswitch
unregister”. That was the best I could do given my limited knowledge.

 

Thank you for the help. I’ll learn eventually.

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of João
Mesquita
Sent: Friday, June 12, 2009 6:47 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Unregister extension?

 

Lars, don't get me wrong but you have been asking questions that are all
answered on the wiki:

http://wiki.freeswitch.org/wiki/Sofia#Flushing.2Frebooting_registered_endpoi
nts

Might be a good idea to value the work of lots of ppl who have been
documenting by actually using the documentation, no?

Sorry if that sounds a bit harsh.

jmesquita

On Fri, Jun 12, 2009 at 10:33 AM, Lars Zeb  wrote:

How can I unregister a softphone’s registration?

 

I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I
changed the second one to 1000. Now when I do ‘sofia status profile
internal’ all three show up. How do I get rid of the 1001 extension? I
shutdown and restarted FS but that didn’t do it.

 

I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is
blocking the Polycom at that same extension and that is the reason the
Polycom is not showing.

 

Thanks, Lars

 

 

Registrations:


=

Call-ID:3c267015ab6b-bd6gioq5ytor

User:   1...@192.168.10.29

Contact:"1010" 

Agent:  snom320/7.3.14

Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24)

Host:   fs

IP: 192.168.10.104

Port:   2048

Auth-User:  1010

Auth-Realm: 192.168.10.29

 

Call-ID:3c267015afa6-6v0sw4o3qei3

User:   1...@192.168.10.29

Contact:"1001" 

Agent:  snom320/7.3.14

Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25)

Host:   fs

IP: 192.168.10.104

Port:   2048

Auth-User:  1001

Auth-Realm: 192.168.10.29

 

Call-ID:OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y.

User:   1...@192.168.10.29

Contact:"1019"


Agent:  Bria Professional release 2.4.3 stamp 50906

Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28)

Host:   fs

IP: 192.168.10.11

Port:   19040

Auth-User:  1019

Auth-Realm: 192.168.10.29

 

Call-ID:3c270d667ff5-47fq2p6n1ou1

User:   1...@192.168.10.29

Contact:"1000" 

Agent:  snom320/7.3.14

Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35)

Host:   fs

IP: 192.168.10.104

Port:   2048

Auth-User:  1000

Auth-Realm: 192.168.10.29

 


=

 


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[Freeswitch-users] Unregister extension?

[Lars Zeb]

How can I unregister a softphone's registration?

 

I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I
changed the second one to 1000. Now when I do 'sofia status profile
internal' all three show up. How do I get rid of the 1001 extension? I
shutdown and restarted FS but that didn't do it.

 

I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is
blocking the Polycom at that same extension and that is the reason the
Polycom is not showing.

 

Thanks, Lars

 

 

Registrations:


=

Call-ID:3c267015ab6b-bd6gioq5ytor

User:   1...@192.168.10.29

Contact:"1010" 

Agent:  snom320/7.3.14

Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24)

Host:   fs

IP: 192.168.10.104

Port:   2048

Auth-User:  1010

Auth-Realm: 192.168.10.29

 

Call-ID:3c267015afa6-6v0sw4o3qei3

User:   1...@192.168.10.29

Contact:"1001" 

Agent:  snom320/7.3.14

Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25)

Host:   fs

IP: 192.168.10.104

Port:   2048

Auth-User:  1001

Auth-Realm: 192.168.10.29

 

Call-ID:OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y.

User:   1...@192.168.10.29

Contact:"1019"


Agent:  Bria Professional release 2.4.3 stamp 50906

Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28)

Host:   fs

IP: 192.168.10.11

Port:   19040

Auth-User:  1019

Auth-Realm: 192.168.10.29

 

Call-ID:3c270d667ff5-47fq2p6n1ou1

User:   1...@192.168.10.29

Contact:"1000" 

Agent:  snom320/7.3.14

Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35)

Host:   fs

IP: 192.168.10.104

Port:   2048

Auth-User:  1000

Auth-Realm: 192.168.10.29

 


=

 

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Re: [Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX- but
delivers XX to FS.

 

Thanks Brian

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 6:41 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match

 

You should be running 7.1.35 or higher.

 

/b

 

On Jun 11, 2009, at 8:34 PM, Lars Zeb wrote:





snom320-SIP 6.5.17.

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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Re: [Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

snom320-SIP 6.5.17.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 5:40 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match

 

What firmware?

 

/b

 

On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote:





It's a SNOM 320.

 

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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Re: [Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

It's a SNOM 320.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 4:05 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match

 

If the phone sends them with dashes in them the phone IS BROKEN and should
be smashed with a hammer.

 

/b

 

On Jun 11, 2009, at 5:57 PM, Lars Zeb wrote:





The users entering numbers into their phonebooks are able to recognize the
number more easily.

 

I will tell them to forget it and make the phone numbers numeric only.

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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Re: [Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

The users entering numbers into their phonebooks are able to recognize the
number more easily.

 

I will tell them to forget it and make the phone numbers numeric only.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, June 11, 2009 2:21 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match

 

 

On Thu, Jun 11, 2009 at 12:49 PM, Lars Zeb  wrote:

I have a match expression for outbound calls as "\d{10}". It's fine for
unformatted numbers. Not knowing any better, I created another extension to
handle numbers formatted like XXX-XXX-, which is easier to read and
exists in one hard phone's phonebook.

 

It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many
extensions for different formats.


Out of curiosity, what benefit does having all these formats get you?
-MC

 

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[Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

I have a match expression for outbound calls as "\d{10}". It's fine for
unformatted numbers. Not knowing any better, I created another extension to
handle numbers formatted like XXX-XXX-, which is easier to read and
exists in one hard phone's phonebook.

 

It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many
extensions for different formats.

 

There's got to be a better way. Any suggestions?

 

Thanks, Lars

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

Michael,

 

Removing everything between the  tag in
sip_profiles/internal/example.xml did the trick - no error message on FS
startup. I'm running 13723.

 

2009-06-11 07:21:03.609317 [INFO] switch_event.c:564 Activate Eventing
Engine.

2009-06-11 07:21:03.612274 [DEBUG] switch_event.c:552 Create event dispatch
thread 0

2009-06-11 07:21:03.995025 [INFO] switch_nat.c:159 Scanning for NAT

2009-06-11 07:21:03.995436 [DEBUG] switch_nat.c:127 Checking for PMP 1/5

2009-06-11 07:21:04.245056 [DEBUG] switch_nat.c:127 Checking for PMP 2/5

2009-06-11 07:21:04.246056 [DEBUG] switch_nat.c:127 Checking for PMP 3/5

2009-06-11 07:21:04.745950 [DEBUG] switch_nat.c:127 Checking for PMP 4/5

2009-06-11 07:21:05.745725 [DEBUG] switch_nat.c:127 Checking for PMP 5/5

2009-06-11 07:21:07.745256 [DEBUG] switch_nat.c:164 Checking for UPnP

2009-06-11 07:21:12.221251 [DEBUG] switch_nat.c:77 No InternetGatewayDevice,
using first entry as default.

2009-06-11 07:21:12.234867 [INFO] switch_nat.c:174 No PMP or UPnP NAT
detected!

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Wednesday, June 10, 2009 5:18 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

 

On Wed, Jun 10, 2009 at 4:54 PM, Lars Zeb  wrote:

Rupa,

 

I think the console log has information in it that log/freeswitch.log does
not.

 

Console:

[r...@fs bin]# ../freeswitch 

2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing
Engine.

2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch
thread 0

2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml
(No such file or directory)

Error including
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml
(Invalid or incomplete multibyte or wide character)

2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT

2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5

2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5

2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5

2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5

2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5

2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP

2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT
detected!

 

log/freeswitch.log:

2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock
interface 'console' to wait for existing references. (from previous
Freeswitch invocation)

2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding
Dialplan 'enum'

2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding
Application 'enum'

2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API
Function 'enum'

2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API
Function 'enum_auto'

2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default
template.

2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql.

2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2.

2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template
example.

2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom.

2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template
linksys.

2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template
asterisk.

 

I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console
log. However, the disk log begins at 16:12:58, whereas the console log
starts at 16:12:50. The console log finishes its NAT and UPnP reporting
before the disk log begins, so I wouldn't see any 0.0.0.0 if it were
present.

 

The [ERR] was due to me removing example.xml from sip_profiles/internal. I
put it back after this. I don't understand the following command in
conf/sofia.conf.xml.

 



I think this is just a cosmetic error. You could probably put an empty xml
file in sip_profiles/internal and be done with it. Or possibly have just an
empty include node, like ""

Try it out and report back - we're dying to know what happens! ;)

-MC 

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 6:05 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

if you haven't changed your logging, then it is probably ok.  The 0.0.0.0
thing is logged at error level, so will show up in the logs.  How did you
search?  Grep?  

grep '0\.0\.0\.0' freeswit

Re: [Freeswitch-users] Help understanding DEBUG and INFO log

[Lars Zeb]

It was. You can see the set at line 2 was done before the bridge at line 4.
What am I missing?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 8:06 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log

 

make sure you set it before the bridge.

 

/b

 

On Jun 11, 2009, at 9:54 AM, Lars Zeb wrote:





Bridge

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 7:24 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log

 

Are you doing an originate or a bridge?

 

/b

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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Re: [Freeswitch-users] Help understanding DEBUG and INFO log

[Lars Zeb]

Bridge

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 7:24 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log

 

Are you doing an originate or a bridge?

 

/b

 

On Jun 11, 2009, at 8:49 AM, Lars Zeb wrote:





In a dialplan, the action sets effective_caller_id_number to a value,
however, in INFO, the displayed value is not the same as the set. Why?

 

http://pastebin.freeswitch.org/9361

 

Thanks, Lars

 

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Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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[Freeswitch-users] Help understanding DEBUG and INFO log

[Lars Zeb]

In a dialplan, the action sets effective_caller_id_number to a value,
however, in INFO, the displayed value is not the same as the set. Why?

 

http://pastebin.freeswitch.org/9361

 

Thanks, Lars

 

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

Rupa,

 

I think the console log has information in it that log/freeswitch.log does
not.

 

Console:

[r...@fs bin]# ./freeswitch 

2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing
Engine.

2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch
thread 0

2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml
(No such file or directory)

Error including
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml
(Invalid or incomplete multibyte or wide character)

2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT

2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5

2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5

2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5

2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5

2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5

2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP

2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT
detected!

 

log/freeswitch.log:

2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock
interface 'console' to wait for existing references. (from previous
Freeswitch invocation)

2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding
Dialplan 'enum'

2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding
Application 'enum'

2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API
Function 'enum'

2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API
Function 'enum_auto'

2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default
template.

2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql.

2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2.

2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template
example.

2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom.

2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template
linksys.

2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template
asterisk.

 

I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console
log. However, the disk log begins at 16:12:58, whereas the console log
starts at 16:12:50. The console log finishes its NAT and UPnP reporting
before the disk log begins, so I wouldn't see any 0.0.0.0 if it were
present.

 

The [ERR] was due to me removing example.xml from sip_profiles/internal. I
put it back after this. I don't understand the following command in
conf/sofia.conf.xml.

 



 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 6:05 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

if you haven't changed your logging, then it is probably ok.  The 0.0.0.0
thing is logged at error level, so will show up in the logs.  How did you
search?  Grep?  

grep '0\.0\.0\.0' freeswitch.log

On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb  wrote:

Rupa,

 

What options do I have for setting up logging? I'm sorry, but I don't know
anything about this.

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 5:19 PM


To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

 

On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb  wrote:

 

It looks like the error message only appears on the console when started
without the nc option; and it does not appear in log/freeswitch.log in any
case.

You might want to review how you have your logging setup then.  The example
I gave you was copied/pasted out of my freeswitch.log file while testing
this fix.  


-- 
-Rupa


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-- 
-Rupa

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[Freeswitch-users] Remove example.com gateway?

[Lars Zeb]

Is it OK to remove the example.com gateway? I removed the example.xml files
in sip_profiles/external and sip_profiles/internal and changed the
default_provider from example.com to myprovider.com.

 

But I still see myprovider.com as a gateway in sofia status. How do I get
rid of this, of course, if it's OK.

 

I want to simplify the configuration. I dialed 18885551212 and it went into
[local.example.com]; and 8885551212 it went into [eavesdrop]. I think I'm
talking more than just the gateway, but I do want to make the dialplan work
in the way I expect.

 

Thanks, Lars

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

There was no 0.0.0.0 anywhere. I used vi. I'll rotate the logs and restart
FS without nc later today and report back.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 6:05 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

if you haven't changed your logging, then it is probably ok.  The 0.0.0.0
thing is logged at error level, so will show up in the logs.  How did you
search?  Grep?  

grep '0\.0\.0\.0' freeswitch.log

On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb  wrote:

Rupa,

 

What options do I have for setting up logging? I'm sorry, but I don't know
anything about this.

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 5:19 PM


To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

 

On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb  wrote:

 

It looks like the error message only appears on the console when started
without the nc option; and it does not appear in log/freeswitch.log in any
case.

You might want to review how you have your logging setup then.  The example
I gave you was copied/pasted out of my freeswitch.log file while testing
this fix.  


-- 
-Rupa


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-- 
-Rupa

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Re: [Freeswitch-users] Documentation error in dialplan XML?

[Lars Zeb]

Done

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, June 09, 2009 7:32 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Documentation error in dialplan XML?

 

Nope shouldn't be there .. if you can update the wiki that would be great.

 

/b

 

On Jun 9, 2009, at 9:25 PM, Lars Zeb wrote:





Is the closing of the condition element correct? I'm new at XML.

 



  



  

   

mailto:sofia/profilename/$%7bdialed_number...@192.168.2.2> "/>

  



 

 

Lars

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Brian West

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-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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[Freeswitch-users] Documentation error in dialplan XML?

[Lars Zeb]

Is the closing of the condition element correct? I'm new at XML.

 



  



  

   



  



 

 

Lars

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

Rupa,

 

What options do I have for setting up logging? I'm sorry, but I don't know
anything about this.

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 5:19 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

 

On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb  wrote:

 

It looks like the error message only appears on the console when started
without the nc option; and it does not appear in log/freeswitch.log in any
case.

You might want to review how you have your logging setup then.  The example
I gave you was copied/pasted out of my freeswitch.log file while testing
this fix.  


-- 
-Rupa

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

Thanks for the explanation, Brian; it was lost on me before.

 

It was a DLink DIR-625 which had UPnP enabled. I turned it off. 

 

It looks like the error message only appears on the console when started
without the nc option; and it does not appear in log/freeswitch.log in any
case.

 

2009-06-09 16:24:32.271913 [INFO] switch_event.c:564 Activate Eventing
Engine.

2009-06-09 16:24:32.274131 [DEBUG] switch_event.c:552 Create event dispatch
thread 0

2009-06-09 16:24:32.663627 [INFO] switch_nat.c:159 Scanning for NAT

2009-06-09 16:24:32.664053 [DEBUG] switch_nat.c:127 Checking for PMP 1/5

2009-06-09 16:24:32.913583 [DEBUG] switch_nat.c:127 Checking for PMP 2/5

2009-06-09 16:24:32.914581 [DEBUG] switch_nat.c:127 Checking for PMP 3/5

2009-06-09 16:24:33.415479 [DEBUG] switch_nat.c:127 Checking for PMP 4/5

2009-06-09 16:24:34.415249 [DEBUG] switch_nat.c:127 Checking for PMP 5/5

2009-06-09 16:24:36.413782 [DEBUG] switch_nat.c:164 Checking for UPnP

2009-06-09 16:24:38.906588 [ERR] switch_nat.c:96 uPNP Device (url:
http://192.168.10.253:/wipconn) returned an invalid external address of
0.0.0.0.  Disabling uPNP

2009-06-09 16:24:38.906633 [INFO] switch_nat.c:174 No PMP or UPnP NAT
detected!

2009-06-09 16:24:38.908650 [INFO] switch_core_sqldb.c:507 Opening DB

2009-06-09 16:24:38.950200 [NOTICE] switch_scheduler.c:166 Starting task
thread

 

Thanks, Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, June 09, 2009 2:33 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

You have to start freeswitch without -nc to see it. Only happens during
start up.

 

/b

 

On Jun 9, 2009, at 4:26 PM, Lars Zeb wrote:





Rupa,

 

Thanks for the detailed response. After upgrading from 13639 to 13732, I see
no log errors. I am accessing Freeswitch vi fs_cli, but I did look in
log/freeswitch.log. Certainly I see nothing that looks like your ERR below.

 

I too have a dlink router. I will look at its configuration and see if upnp
is enabled.

 

Lars

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

Rupa,

 

Thanks for the detailed response. After upgrading from 13639 to 13732, I see
no log errors. I am accessing Freeswitch vi fs_cli, but I did look in
log/freeswitch.log. Certainly I see nothing that looks like your ERR below.

 

I too have a dlink router. I will look at its configuration and see if upnp
is enabled.

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 11:16 AM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

If we get 0.0.0.0 as a public address, upnp support is disabled since we're
getting the info from a gateway that can't route us to the internet.  It is
broken, so we don't trust it.

The message you should see in the logs is something like:

2009-06-08 13:51:44.587812 [ERR] switch_nat.c:126 uPNP Device (url:
http://192.168.1.2:/wipconn) returned an invalid external address of
0.0.0.0.  Disabling uPNP
2009-06-08 13:51:44.587812 [INFO] switch_nat.c:380 No PMP or UPnP NAT
detected!

You'll probably not see this if you start fs in the background and then
connect with fs_cli.  So, look in your log files for it.

The url will give you an idea as to which device is sending you invalid
info.  In my case it is a dlink router setup as a access point but still was
implementing upnp (bad).  I was able to disable upnp on that router.

My printer (Epson Artisan 800) also participates in upnp, but it doesn't
respond to the internet gateway stuff, so it was not the source of a problem
for address discovery.  It is causing me other issues but that is another
story for another day for code that isn't committed yet.

2009/6/9 Lars Zeb 

Thanks, Brian and Mike J and Ken R and Jason W.

 

Outbound calls are now working OK.

 

Brian, I don't know where to look for the 0.0.0.0 addr error message. I
checked the log/freeswitch.log but did not recognize anything.

 

I also noticed that nat_public_addr is not longer displayed in the
global_getvar command. How is this value set?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, June 09, 2009 9:06 AM


To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

Because rupa on IRC is having the same problem.. Check the error message I
print now and the device url will be printed thanks to rupa's patch.

 

/b

 

On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote:

 

Brian, I'm curious, how can you tell that a printer is giving out the
0.0.0.0 addr?

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 


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-- 
-Rupa

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

Thanks, Brian and Mike J and Ken R and Jason W.

 

Outbound calls are now working OK.

 

Brian, I don't know where to look for the 0.0.0.0 addr error message. I
checked the log/freeswitch.log but did not recognize anything.

 

I also noticed that nat_public_addr is not longer displayed in the
global_getvar command. How is this value set?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, June 09, 2009 9:06 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

Because rupa on IRC is having the same problem.. Check the error message I
print now and the device url will be printed thanks to rupa's patch.

 

/b

 

On Jun 9, 2009, at 11:03 AM, Lars Zeb wrote:





Brian, I'm curious, how can you tell that a printer is giving out the
0.0.0.0 addr?

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

Brian, I'm curious, how can you tell that a printer is giving out the
0.0.0.0 addr?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, June 09, 2009 7:41 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

update to 13719, rupa did a patch that fixed this.. also find that printer
that gives out the 0.0.0.0 addr and turn off upnp :P

 

/b

 

On Jun 8, 2009, at 9:57 PM, Lars Zeb wrote:





http://pastebin.freeswitch.org/9319

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, June 08, 2009 7:39 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

pastebin your profile config and the output of global_getvar

 

/b

 

On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote:






I had a working FS installation which I messed up by doing a fresh install.
I tried to integrate all my custom changes, but I'm sure I screwed something
up.

 

The symptom is on an outbound call, sometimes I can hear ringing, other
times I cannot. Finally I can see FS connects via a softphone, but I hear
only silence. The other side of the conversation hears static.

 

I pasted a siptrace of the external profile. The Contact, Via and SDP shows
an address 0.0.0.0.  http://pastebin.freeswitch.org/9318

 

FS exists on a LAN behind a NAT firewall along with all its clients. There
is a SwitchVox system which predates the FS. I had to use an external sip =
5090 for FS. Also I think I had to use a different WAN address
(xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came
up with (xxx.xxx.xxx.82), but I can't figure out where I set this address.

 

I would appreciate any help. Thanks, Lars

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Brian West

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-- Meet us at ClueCon!  http://www.cluecon.com <http://www.cluecon.com/> 

 

 

 

 

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[Freeswitch-users] Error making FS configure

[Lars Zeb]

I tried to make FS current with the following command, which got errors in
the ./configure step. Is it OK to proceed to make install?

 

make clean && svn up && ./configure

configure: creating ./config.status

config.status: creating Makefile

config.status: creating libedit.pc

config.status: creating src/Makefile

config.status: creating doc/Makefile

config.status: creating examples/Makefile

config.status: creating config.h

config.status: executing depfiles commands

configure: configuring in libs/pcre

configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch
--cache-file=/dev/null --srcdir=.

configure: error: cannot find sources (pcre.h.in) in .

configure: error: /bin/sh './configure.gnu' failed for libs/pcre

 

Thanks, Lars

 

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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

http://pastebin.freeswitch.org/9319

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, June 08, 2009 7:39 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

pastebin your profile config and the output of global_getvar

 

/b

 

On Jun 8, 2009, at 9:24 PM, Lars Zeb wrote:





I had a working FS installation which I messed up by doing a fresh install.
I tried to integrate all my custom changes, but I'm sure I screwed something
up.

 

The symptom is on an outbound call, sometimes I can hear ringing, other
times I cannot. Finally I can see FS connects via a softphone, but I hear
only silence. The other side of the conversation hears static.

 

I pasted a siptrace of the external profile. The Contact, Via and SDP shows
an address 0.0.0.0.  http://pastebin.freeswitch.org/9318

 

FS exists on a LAN behind a NAT firewall along with all its clients. There
is a SwitchVox system which predates the FS. I had to use an external sip =
5090 for FS. Also I think I had to use a different WAN address
(xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came
up with (xxx.xxx.xxx.82), but I can't figure out where I set this address.

 

I would appreciate any help. Thanks, Lars

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[Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

I had a working FS installation which I messed up by doing a fresh install.
I tried to integrate all my custom changes, but I'm sure I screwed something
up.

 

The symptom is on an outbound call, sometimes I can hear ringing, other
times I cannot. Finally I can see FS connects via a softphone, but I hear
only silence. The other side of the conversation hears static.

 

I pasted a siptrace of the external profile. The Contact, Via and SDP shows
an address 0.0.0.0.  http://pastebin.freeswitch.org/9318

 

FS exists on a LAN behind a NAT firewall along with all its clients. There
is a SwitchVox system which predates the FS. I had to use an external sip =
5090 for FS. Also I think I had to use a different WAN address
(xxx.xxx.xxx.83, which is the address mapped to 5090) than the one FS came
up with (xxx.xxx.xxx.82), but I can't figure out where I set this address.

 

I would appreciate any help. Thanks, Lars

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