"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
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y and
>> default context it sucessfully calls other internal phones
>> with bridge to
>> "sofia/internal/sip:exters...@public-ip:translated-port";
>> however when I log into "Company1" with the phones, it tries
>> "sofia/internal/dialed-
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
Still having this issue. Do separate domains need to be real fully
qualified domains, or can they just be added as in Company1, 2, 3, etc?
On 12/22/2009 9:16 AM, John wrote:
> Thanks Brian. I did have both force-register-domain and
> force-register-db-domain commented in both the intern
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 b
file extensions
including 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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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.freesw
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
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/[EM
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
fault.xml?
>
> It needs to be above this line:
>
>
>
>
> /b
>
>
> On Oct 10, 2008, at 1:22 PM, John Millican wrote:
>
>> I currently have this in default.xml in the context default:
>>
>>
>>
>>mailto:sofia/external/[
ault.xml?r=9935
>
> /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 ou
voicemail
path (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 PROTECTE
rs that d is empty. Though:
var d = apiExecute ("status", "");
console_log ("err", "D " + d + "\n");
works fine.
What's the proper way to invoke xml_locate from javascript?
-- John
-----
ate.c:1067 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 / poi
openzap.conf.xml contains:
in each of the analog_spans / analog_em_spans. Is something else needed
to specify the domain for processing inbound openzap calls?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email:
re the source isn't mod_sofia.
What JIRA category should I file this under?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| F
ck
glance and wasn't sure what to use.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle| F
audio from one 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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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
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 a
dge.
On Wed, Dec 10, 2008 at 8:36 AM, John Rutherford <[EMAIL PROTECTED]>
wrote:
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: Monda
nsfer
would be most helpful to capture a pcap of the entire thing by itself
start to finish.
/b
On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
No. I realize that's it's a B2BUA and that's exactly what we want.
Everything with the transfer seems to work fine,
, 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 pca
so the OS 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]
t: Re: [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
ca
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: frees
[defa...@192.168.14.10]
where 192.168.14.10 is the number assigned to the logical interface,
however the call goes through / everything seems to work. I'm able
to place a call to the VoIP phone from openzap without any complaints.
Suggestions / pointers regarding the warning and
ets it from sip_auth_realm. 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 domai
est of FreeSWITCH shouldn't know or even care that
force-register-domain is in use ... it should be as if the
VoIP phone had in fact registered using the domain specified
by force-register-domain.
-- John
-
| Feith Systems
I think the problem with changing the CPU's is off your going from a single
to multi, or multi to single CPU and not adjusting the kernel from SMP to
normal or vise versa?
I've got a nice ESXi farm at the moment (32 cores, 48 gigs of ram) thats
running at under 5% usage. Haven't noticed any timing
?
-- john
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e calls switched on the DID.
I'm struggling to know where to start - can someone point me in the
right direction?
Regards,
John
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http://lists.freeswitch.org/mailm
Hi;
We're looking to set up a CP which will interact with BT's 21CN network
using the IPX gateway.
We're running through the test scenarios (which, unfortunately, we have
under NDA) now.
Just wondering if anyone out there has already passed the test suite
with Freeswitch
re, click "new message" input
> 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.
>
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 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 xlite or a
a Sangoma 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
---
eived from FreeSWITCH. At this point
the Cisco gets unhappy and marks Layer 2 as down.
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
eBSD).
See:
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: 1-2
7;t seem to be one
fired for this condition.
Thanks in advance.
{P^/
John
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author 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
Virtualization has issues with timing in my experiance.
Sent from my iPhone
On May 25, 2009, at 6:43 PM, Peter P GMX wrote:
> We have used FS on ec2 for testing purposes only. It was ok. We havn't
> done any performance test though.
>
> Best regards
> Peter
>
> Ing. Edwin Villarreal schrieb:
>>
call at the front of the fifo "fifo".
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax:
sip:(@[^:]*):.*$/g;
new_name = name.replace (re, "/$1");
}
return new_name;
}
Suggestions 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
ndle 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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oxy lite that can 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.
whe
t;
>> How are you handling your FS box crashing?
>>
>> -Original Message-
>> From: freeswitch-users-boun...@lists.freeswitch.org
>> [mailto:freeswitch-users-boun...@lists.freeswitch.org
>> ] On Behalf Of John Dalgliesh
>> Sent: Wednesday, June 10, 20
On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote:
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 not stuck.
That is,
ferred-By variable. E.g.:
allows the station id making the transfer to be known when a call is
transfered to *5. The station id can then be used to park the call in
the proper fifo.
-- John
-
| Feit
the upgrade.
> -Michael
{P^/
> -Original Message-
> From: freeswitch-users-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.
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
>>>
ar what is said 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
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
o 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
t this default changed ... the older FS came
with these commented out and worked fine in the simple configuration
where the server and phones are on the same network segment.
In any case my config has been adjusted, things are working, it's
Friday, and I get to go home so
ua originate method seems to require timeout to be specified
even though the documentation implies it's optional.
c) Using this approach causes the message:
[WARNING] mod_limit.c:576 USAGE: hash [insert|delete]///
which I have yet to track down.
Thoughts?
-- John
-
pt fails.
Explicitly calling session.originate seems to allow you to check if
the call 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
-
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 Sy
> 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-
");
apiExecute ("voicemail", "check default " + domain + " " + from);
however the apiExecute doesn't seem to invoke voicemail. Probably
because there's a difference between invoking an application and
an api. So, how do you invoke an ap
omeone else which
means that based 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 tags later in the dialplan
to control how to call the voicemail application?
-- John
---
just setting variables
and handling the rest in the dialplan since I can't use conditionals in
the dialplan after the application to parse the DTMF has been called.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 |
> if you do
> session.execute("transfer", "");
> 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
... not dialing?
I.e. dialing 551 just gets me a PBX line with dialtone.
3) What condition would I use in my dialplan to match an FXO line ringing?
I.e. when the FXO line rings I want to invoke javascript.
-- John
-
|
dialing 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
andled 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: 1-215-54
declaration of function `select'
src/zap_ss7_boost.c:878: warning: implicit declaration of function `FD_ISSET'
make: *** [src/zap_ss7_boost.o] Error 1
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email:
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: 1-21
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.
--
s two tones, 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
I currently have:
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
es. In my particular application the System 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 th
the A port (aka any available port on span 2)?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
| John Wehle| Fax: 1-215-540-5
>
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 n
angup
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
oicemail application is notified of the hangup when ##99 is received on
the line?
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehle|
ng openzap span 4 (a System 25 loopstart line) to extension #90482
which, if successful, causes 482's message waiting light to go on (the
DTMF signals from the System 25 indicated to the JavaScript program that
the voice mail message was for ext 482).
So what's the correct, rel
sage?
I'm not adverse to doing some scripting / web page development,
however I'd prefer a pointer in the right direction and I'm also
not looking to reinvent the wheel.
-- John
-
| Feith Systems | Voice:
The short version (to answer my own question) is:
a) Enable mod_xml_rpc.
b) Enable mod_shout.
c) Change file-extension in voicemail.conf.xml to mp3.
d) The URL is:
http://my_ip:8080/api/voicemail/web
-- John
what needs to happen in order to implement it? I may be
up for contributing code if someone points me in the right direction.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: [EMAIL PROTECTED] |
|John Wehl
;m winkstart uses MF, * not *
DTMF to send the phone number / extension ... at least
my PBX doesn't respond when I send DTMF.
Is there support in FreeSWITCH for MF?
If not, then any pointers to how to go about ad
> 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???
-- J
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 ther
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 the
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:
I have the following configured in /usr/local/freesw
> What rev exactly?
>
> /b
>
> 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
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 s
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 c
[Freeswitch-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 P
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?
br...@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
from there.
>
> Mike
>
> On Nov 24, 2009, at 5:04 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
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: "
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 was
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
__
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.
_
Windows
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]
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
>
>
>
>
>
>
>
>
> J
However, I think this assume that I have an autoconf file in /usr/bin.
Needless to say, I still get the error messages that autoconf 2.59 or
above is required.
Can anyone help me on how to solve this configuration error.
/john
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