...@public-ip:translated-port;
however when I log into Company1 with the phones, it tries
sofia/internal/dialed-extens...@company1 ... I also get
User not Registered. The dialplans are the same either
way.
Any ideas?
Thanks
John
___
FreeSWITCH
One point of clarification, currently all the phones are behind NAT, so
it appears that when the phones are in a Non-multitenant scenario, they
use SIP:dialed_num...@ip-address-of-their-gateway.
On 12/22/2009 9:16 AM, John wrote:
Thanks Brian. I did have both force-register-domain
How can I perform click-to-call or click-to-dial in FreeSWITCH?
Do you have any recommendations on programs capable of click-to-call or
click-to-dial from Microsoft Outlook or Microsoft Excel?
You've made my day.
From: jpitc...@nuvio.com
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 16 Dec 2009 08:11:05 -0800
Subject: Re: [Freeswitch-users] Click-to-call and click-to-dial
John,
To do a click to call in FS you need to have some
I have uploaded the dialplan and JavaScript files used to process calls to
MODENDP-272. I have even done a make current to revision 15755, and the blind
transfer is still failing.
_
I am new to FS having ditched Asterisk a few weeks ago. I have iptel
registered but I cannot get Inphonex to work. I am using the settings from
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no
avail.
The error displayed in the console is 2009-12-02 21:32:55.243917 [ERR]
I attempted to do a make current with revision 15739, but some of the Sofia
source files will not compile with revision 15739. Those source files were not
changed between revisions 15738 and 15739. I am using GCC 4.1.2 to compile
FreeSWITCH. I used the following to get revision 15738, which
I have tried to do a blind transfer from a phone that is registered with
FreeSWITCH, and it will fail, even when proxying and media bypass are enabled.
Details about this issue can be found here:
http://jira.freeswitch.org/browse/MODENDP-272
To clarify the problem, the invite message is incorrect because comfort noise
is being negotiated in the re-invite instead of G.711 or G.729:
INVITE sip:19729831...@168.75.202.246:5060 SIP/2.0
Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj
Max-Forwards: 69
From: John
:04 PM, John Platts wrote:
I have considered writing JavaScript code to bridge two calls together.
However, I would like to perform custom handling of the 302 Moved
Temporarily response. How do I handle the 302 Moved Temporarily response if
I use JavaScript
I have modified sofia.c in mod_sofia so that I can define gateways without
having to specify the password parameter. This is because I am using a SIP
gateway that does not require SIP registration. The modified version still
requires the password to be set on any gateway for which register is
I was having trouble doing call forwarding from my SIP phone that is connected
to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved
Temporarily responses, but my SIP gateway does not support 302 Moved
Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward
-users] Problems with proxy media and bypass media in
FreeSWITCH
This was fixed in trunk yesterday about 8 hrs before you sent this message.
(15619). Please update and try again.
Mike
On Nov 23, 2009, at 11:33 PM, John Platts wrote:
I was using revision 15586
I have considered writing JavaScript code to bridge two calls together.
However, I would like to perform custom handling of the 302 Moved Temporarily
response. How do I handle the 302 Moved Temporarily response if I use
JavaScript?
...@freeswitch.org
Date: Tue, 24 Nov 2009 15:32:44 -0600
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call forwarding problem
You'll have to hairpin the media thru your machine usually if they
won't accept either of those.
/b
On Nov 24, 2009, at 3:05 PM, John Platts wrote
I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in
/usr/local/freeswitch/conf/dialplan/default.xml:
extension name=setup_media continue=true
condition field=${sip_nat_detected} expression=true
action
On Nov 23, 2009, at 6:19 PM, John Platts wrote:
I actually checked out the latest version of FreeSWITCH in the SVN
repository.
I have the following configured in /usr/local/freeswitch/conf/
dialplan/default.xml:
___
FreeSWITCH
I have installed FreeSWITCH on our server, and need some help configuring our
FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance
are in the format: 1NPANXX (where NPA is the area code, and NXX are the
last 7 digits of the phone number).
I need
Brian West wrote:
I looked out my window... but I didn't see pigs flying... did I miss
something! :P
/b
On Nov 4, 2009, at 11:22 AM, Giovanni Maruzzelli wrote:
...and will get more people using the x64 version of Windows! ;)
-gm
When their own commercials say that there old
How do you get a system variable from within a lua startup script?
Specifically I want domain_name from vars.xml ... normally I'd use
session:getVariable, however there is no session in this case.
-- John
-
| Feith Systems
You can execute global_getvar api call.
Thanks ... I've updated the wiki.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495
was successful ... is there a particular reason it's discouraged?
I'm happy to avoid it if a better approach is available, however I'm
having trouble finding one.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j
On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote:
On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote:
On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
Well, if you're running multiple machines, waiting for it to drainstop
isn't that big of a deal unless you're in some sort
on the openzap
side, however openzap hears silence from the Grandstream.
Calling from Grandstream to Grandstream doesn't work ... call goes
through however both sides hear silence.
Suggestions on how to proceed?
-- John
BTW: in all cases show channels says PCMU 8000 is being used
for the read and well as write codec.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540
to the Grandstream
in order for the phone to send audio?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495
get to go home so life is good. :-)
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495
be enabled on the server machine when
needed. Anyway that lost out as it's more work and even less portable.
{P^/
John
On Thu, 11 Jun 2009 at 11:54 -0500, Anthony Minessale wrote:
or you can put a sip proxy in front of 2 boxes where you can control the
flow of traffic.
when you want to upgrade one
-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of John Dalgliesh
Sent: Wednesday, June 10, 2009 9:04 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Live Upgrade Techniques
Hi,
I am slowly
On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo m...@giagnocavo.netwrote:
Exactly. You probably want to have something like this anyways, so that
when someone accidentally unplugs the system, or the disks/CPU/RAM crash,
you’re
to park the call in
the proper fifo.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495
-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of John
Dalgliesh
Sent: Thursday, June 11, 2009 12:14 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Live Upgrade Techniques
I assume he's talking about hardware
for a better approach? Keep in mind that my existing user
population expects (for better or worse) to use *5 to park the call on
their phone so I'm somewhat limited in what I can do.
-- John
-
| Feith Systems | Voice: 1
these upgrade
windows. It seems like a bit of a waste.)
So how are you handling your FS software upgrades?
{P^/
John
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for this condition.
Thanks in advance.
{P^/
John
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of sofia said it would be a big job to bring that up to the even
callback.
Someone may be able to persuade him to allow you to pass a global timeout
waiting for 100
or something but no solution exists atm
On Tue, Apr 21, 2009 at 3:32 AM, John Dalgliesh jo...@defyne.org wrote:
Hi,
I am trying
:
http://wiki.sangoma.com/wanpipe-freebsd-drivers
for futher information.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495
.
If nothing obvious comes to anyone's mind, then I'll
simply need to trace through the FreeSWITCH ISDN code
and see what's going on.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle
A104d on FreeBSD 6.4.
I unfortunately don't currently speak ISDN (though I'm starting to pick
up a little as a result of this exercise) ... suggestions / hints regarding
what's going on and how to resolve it would be welcomed.
-- John
-- wanpipe2.conf
settings and restart the phone.
Or in cfg files for aastra set:
sip use basic codecs: 1
regards- John
On Wed, Feb 4, 2009 at 10:06 PM, John Hyde jacre...@gmail.com wrote:
I am having problems getting an Aastra 57i to make calls through FS. the
phone registers fine, but all calls fail. If i use
freeswitch-users@lists.freeswitch.org
then type your subject and message then click send. Your email
client echo's back the headers that causes the mailing list server and
many email clients to thread the message properly.
Whoops, sorry!
User IQ Error.
jd
--
John Daragon
.
I'm struggling to know where to start - can someone point me in the
right direction?
Regards,
John
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Michael Collins wrote:
If anyone figures this out please post it to this thread. I am working
on a wiki page for the VMWare appliance and I would like to be able to
inform people on how to handle this situation.
I had some issues under vmware fusion. They were resolved by adding
clock=pit [1]
openldap to provide these information..
but these about these binding settings... where should i set them?
best regards
John Skopis (Lists) wrote:
vinicius wrote:
hi ppl.. i tried to find something at google, but i couldnt manage to find
anything.
i still dont know what to do to make
... it should be as if the
VoIP phone had in fact registered using the domain specified
by force-register-domain.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax
. I guess the question is if
force-register-domain is being used then:
a) Should sip_auth_realm be set by FreeSWITCH to the value associated
with force-register-domain
b) or should set_domain in default.xml simply check for force-register-domain
when setting domain?
-- John
://172.16.75.129; /
param name=binddn value=cn=admin,dc=example /
param name=bindpass value=secret /
/binding
/bindings
which should/probably/might work with ldap objects like these:
dn: cn=John Skopis,ou=people,dc=example
objectClass: person
objectClass: inetOrgPerson
objectClass
: [Freeswitch-users] No audio after transfer
I smell a NAT... is there any NAT involved?
On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford [EMAIL PROTECTED]
wrote:
Okay. I just tried this.
Now we're getting the audio going one way, but not the other. So, I
can
hear the person that I just
Sent.
Let me know if you see anything. I'm not able to see anything wrong.
Thanks,
John
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Raymond Chandler
Sent: Thursday, December 11, 2008 9:39 AM
To: freeswitch
Sorry to repost, but I haven't heard anything back on this in a little
while.
I checked out the trunk last week. I'm on revision 10597.
Thanks,
John
From: John Rutherford
Sent: Monday, December 08, 2008 4:36 PM
To: freeswitch-users@lists.freeswitch.org
Subject: No audio after
, 2008 at 10:01 AM, John Rutherford [EMAIL PROTECTED]
wrote:
I have a pcap, but I'm not able to see anything obviously wrong with
it.
We find that some equipment (in fact a lot of equipment) have features
that cause issues to be quite non-obvious, so perhaps you could give
the pcap to Brian
catches the port not open and returns an ICMP 3:3
back to the MSS which in turn chokes on the queued up RTP and
refuses to send anymore...
-Ray
John Rutherford wrote:
I just emailed it to him.
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
call to the other call and vice-versa.
Any help would be greatly appreciated. I have a pcap and the freeSWITCH
logs, and I can easily reproduce this.
Thanks!
John
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Freeswitch-users@lists.freeswitch.org
http
Sorry. I forgot to mention that.
I checked out the trunk last week. I have revision 10597.
John
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian West
Sent: Monday, December 08, 2008 4:48 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch
switch_ivr_originate() Can not create
outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
It appears openzap sees the request from the PBX fine ... somehow
FreeSWITCH can't connect the openzap inbound call to 1003 with the
VoIP phone on ext 1003.
Suggestions / pointers?
-- John
of the
server's interface.
openzap.conf.xml contains:
param name=context value=default/
in each of the analog_spans / analog_em_spans. Is something else needed
to specify the domain for processing inbound openzap calls?
-- John
figure it out.
--
However it appears that the logic is wrong. It fails to handle
cases where the source isn't mod_sofia.
What JIRA category should I file this under?
-- John
-
| Feith Systems | Voice: 1
and wasn't sure what to use.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495
, );
console_log (err, D + d + \n);
works fine.
What's the proper way to invoke xml_locate from javascript?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215
(currently it is also hardcoded)? I use it in order to play the
recorded_name (if present) for an extension when doing the lookup.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John
Channel sofia/internal/[EMAIL PROTECTED]
[521c96a2-5205-bf46-9f9f-31124757b0ef]
2008-10-10 13:33:24 [INFO] mod_dialplan_xml.c:228 dialplan_hunt()
Processing John Millican-2002 in context default
2008-10-10 13:33:24 [NOTICE] switch_ivr.c:1098
switch_ivr_session_transfer() Transfer
sofia/internal/[EMAIL
Brian West wrote:
Its looking for extension 2002 in context default on FreeSWITCH and
one doesn't exist so you get the NO ROUTE message.
Add a route to map 2002 so that it points at the Asterisk box.
/b
On Oct 10, 2008, at 1:00 PM, John Millican wrote:
Processing John Millican-2002
to be above this line:
X-PRE-PROCESS cmd=include data=default/*.xml/
/b
On Oct 10, 2008, at 1:22 PM, John Millican wrote:
I currently have this in default.xml in the context default:
extension name=ast_extens
condition field=destination_number expression=^(2\d{3})$
action
/b
On Oct 10, 2008, at 1:46 PM, John Millican wrote:
Yep that was it! Now I just need to add a matching gateway, as I am
getting the error no matching gateway found, which I think I can
figure out.
Thank you for such quick and accurate help.
JohnM
wav, gsm, au, etc., and I get this same error with all of
them. Has anyone run into this before?
I installed from the openSUSE 10.3 rpm.
Thanks!
John
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http
No. I have 'inbound-proxy-mode' set to true and I have
'inbound-late-negotiation' set to true as well.
John
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian West
Sent: Friday, October 03, 2008 12:27 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re
That makes sense. I disabled those options and it's working now.
Thanks!
John
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian West
Sent: Friday, October 03, 2008 1:21 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] help
Anyone had any luck getting the Sangoma A 102 card into a 1 U box?
I'm looking to build 2 freeswitch servers for redundancy, and was
wondering if anyone has had any luck getting these cards to work with
1U riser cards and if so with what brand of case/mobo?
Any recommended bare bones kits would
Is there support in FreeSWITCH for MF?
To answer my own question ...
Perhaps I can add a ZAP_COMMAND_SEND_MF case to zap_channel_command
patterned after ZAP_COMMAND_SEND_DTMF and just use teletone_set_tone
to set the proper frequencies???
-- John
Brian West wrote:
http://www.tunnelbroker.net If you need/want ipv6 tunnels.
sixxs also
http://www.sixxs.net/pops/occaid/
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Michael Collins wrote:
That begs the question… is there a mechanism in sqlite or Linux that
allows for the RAM drive to be backed up periodically? That would be a
cool feature to get documented for those power users like Ken! ;)
Interesting thought:
Grey Man wrote:
[snip]
One suggestion I'd have for another row is Security Fix Rate. For
example while the Asterisk community's approach to handling security
releases is commendable the rate at which they happen is a real pain
when you have to potentially upgrade a production system for each
/curl.spec
./libs/libsndfile/libsndfile.spec
./libs/js/nsprpub/pkg/linux/sun-nspr.spec
./freeswitch.spec
I have never actually used freeswitch packages. I think it would be
great if they would offer daily debug/prod builds.
Also, irc is a great resource.
HTH
-john
Birgit Arkesteijn wrote:
Hi all,
We've got an older version of FreeSWITCH (Trunk 7948) running on a Linux
x86_64 machine. At the moment it's crashing few times a day, making our
services very unreliable.
At the moment we don't have the time to rebuild this version, so I'm
looking for
Adrian Gschwend wrote:
Adrian Gschwend wrote:
I can't compile latest trunk on FreeBSD 7.0-STABLE, bootstrap works,
configure fails here:
[...]
I could solve that problem at least:
My BSD has the following autoconf installed:
autoconf-2.61_2
autoconf-2.62
- autoconf-2.62 is the
Adrian Gschwend wrote:
Hi group,
I can't compile latest trunk on FreeBSD 7.0-STABLE, bootstrap works,
configure fails here:
=== configuring in libs/libsndfile
(/usr/home/ktk/freeswitch.trunk/libs/libsndfile)
configure: running /bin/sh ./configure.gnu --disable-option-checking
25 PBX connects to FreeSWITCH
using OpenZAP running on a Sangoma A204DX card. At the end of a call sent
to voicemail I need FreeSWITCH to send some DTMF tones to the PBX in order
to turn on / off the message waiting indicator.
-- John
action application=gentones data=1234567890/
Yep, this works through the A204d FXO openzap lines, though sometimes
there's little odd click / hiccup in the middle of playing the tones.
I'm a little confused as to why using send_dtmf didn't seem to
work well, however no matter.
-- John
until 33 seconds have elapsed.
I've emailed Sangoma however thought to post it here in case it
rings a bell.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax
, and then disconnects. How do I
configure FreeSWITCH to send all the dtmf tones?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495
}/
/condition
/extension
which routes calls to extension 4XX out the first openzap line.
How do I set up a pool of openzap lines and route calls to extension
4XX to any available openzap line?
-- John
-
| Feith Systems
Just curious as to the state of / plan for T1 RBS support.
Our System 25 PBX doesn't support ISDN, however it does support
RBS. It would be nice to be able to run more lines into our
FreeSWITCH box using a T1 instead of analog lines.
-- John
: warning: implicit declaration of function `FD_ISSET'
make: *** [src/zap_ss7_boost.o] Error 1
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495
an extension on the PBX.
-- John
---8---8---
*** libs/openzap/src/include/openzap.h.ORIGINAL Tue Jul 1 19:07:52 2008
--- libs/openzap/src/include/openzap.h Tue Jul 1 19:20:12 2008
***
*** 127,132
--- 127,133
#include
event 17
Any idea what they mean? Should I be concern?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495
to match an FXO line ringing?
I.e. when the FXO line rings I want to invoke javascript.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| Fax: 1-215-540-5495
on the DTMF I need to call the voicemail application
with different parameters. How can I have the script just set variables
if I can't use those variables with condition tags later in the dialplan
to control how to call the voicemail application?
-- John
if you do
session.execute(transfer, some ext);
and exit the script.
That goes back to the dialplan for a *new* lookup so now your variables can
be used in conditions.
Thanks,
-- John
-
| Feith Systems | Voice: 1
Brian West wrote:
On Jun 18, 2008, at 1:11 AM, Aadilkhan Maniyar wrote:
Thanks for the reply Brian.
So what you mean to say is that I need not configure
mod_spidermonkey_odbc at all in order to store registration data in
the MySQL database?.
Correct this is just a way to access odbc
Nicola wrote:
Thanks ... now works ...
the problem was that the server ldap lacked the parameter: dial-string.
Last problem:
How come when I call some internal, the voice can not go?
Could it be a firewall issue? Do you see anything in your logs?
: default
variablevalue: test
variablevalue: 0
hth,
john
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there is no sofia domain configured of that name (there shouldn't
be at least) the default config is gone. Of course you could also just
rm -rf conf/directory ;]
-john
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Nicola wrote:
thanks for your answer the problem is that FreeSwitch NOT MADE no
query the LDAP server. Consequently, I can not know whether the query is
wrong or not. I tried to follow the instructions you have given me kindly
... but freeswitch not going to make any queries on LDAP: In
now... sip phones != mail clients. but if they were... Thanks
for reminding me Mike.
I am not exactly sure how difficult it is to do digest auth in openldap
or AD. If I manage to find some time I will look into it though,
-john
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Freeswitch-users
this so ldap schema didn't
have to be extended everytime the fs xml schema is extended. The reason
would be rather than requesting 100 attrs and iterating through them all
just request the ~4 required to generate the xml and iterate through them.
Also startls/ssl support might be nice ;]
-john
PS
uses) though.
-John
Anthony Minessale wrote:
We have a concept called the directory interface not to be confused
with the user directory.
The directory interface is a pluggable abstract API that looks and feels
like LDAP only you can plug in anything you want to implement the
functions
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