Hi!
I'm sure this is a NAT issue, but I'm not sure what options to use.
I have a Linksys SPA3102, NAT'd on the internet (remotely) and
connected to my FS on the otherside of the world, which is also
natted. A PAP2T is connected on the same subnet as the FS. The 3102
registers successfully and
Thanks Brian and Gad,
I have stun set and if I do a 'sofia status profile internal', I see
the external IP address of the 3102 ATA, so I assume that stun is
working correctly on the SPA3102.
These are the options that I have set (according to the 3102 manual).
• Handle VIA received: yes
•
Thanks Yehavi...
I posted a question on the Cisco Forum and am waiting a response from
'engineering' to tell us if they plan to implement standard SRTP
support in the Linksys ATA's.
TLS is working fine.
On Thu, Dec 17, 2009 at 4:39 PM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
An
Pennytel.com
On Sat, Dec 12, 2009 at 12:52 AM, Phillip Jones pjinthe...@gmail.com wrote:
Hi - sorry to go off topic - but we are looking for Voip supplier with SMS
capability. Would you mind telling me which Voip supplier you use?
On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith
I use a dectop by Data Evolution... Its cheap at ~$100. I have it
running debian lenny and FS... works well for me.
http://www.dataevolution.com/dectop%20info%202.htm
http://www.gadgettastic.com/2007/08/18/dectop-the-100-pc/
On Thu, Dec 10, 2009 at 10:45 PM, Fred-145 codecompl...@free.fr
Hi!
My voip provider provides a SOAP interface to be able to send SMS's,
so after a voicemail is left, I want to execute a 'send sms' script.
I don't want a separate statement in the dialplan after the voicemail
statement because I only want to send sms's when a voicemail is
actually left.
The
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber
Collins m...@freeswitch.org wrote:
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000
the user so that the vars become available on the
leg you're processing.
-MC
On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
That's exactly what I want to do and that was the first thing I tried,
but nothing is passed to the script.
In a case like
, eman e...@chabotel.com wrote:
Check out the Linksys SPA2102
On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
The only ATA mentioned on the WIKI that supports TLS/SRTP is the
Grandstream HandyTone 503. But, again according to the wiki, that
doesn't seem
, not sure if Cisco plans on
upgrading the firmware to ever support SDES on the ATAs. They added
support for SDES to their IP Phones about 1 year ago, but nothing has
happened with the ATAs as of yet.
Gabe
On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi
it (so says their marketing material and docs).
I'd see if Cisco has any plans to add support for it to the ATAs. Next
time I see our Cisco SE, I'll try to poke him about it.
Gabe
On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Quote: Cisco/Linksys SPA
; they
should support TLS also (will try it next week; up to now I preffered to not
use TLS so I can sniff the traffic and debug things).
Regards, __Yehavi:
2009/12/4 Mark Campbell-Smith mcampbellsm...@gmail.com
Cheers Gabriel.. thanks for the information.
I'll look
The only ATA mentioned on the WIKI that supports TLS/SRTP is the
Grandstream HandyTone 503. But, again according to the wiki, that
doesn't seem to behave to well with TLS ...
On Wed, Nov 25, 2009 at 7:14 PM, Jason White ja...@jasonjgw.net wrote:
Mark Campbell-Smith mcampbellsm...@gmail.com
Didn't Michael already answer this? Best read the FS wiki and the
softphone user guide for help with this.
http://wiki.freeswitch.org/wiki/Getting_Started_Guide
http://wiki.freeswitch.org/wiki/Interop_List
On Wed, Nov 25, 2009 at 7:29 PM, ovvenkat ovvenkate...@gmail.com wrote:
Hi to All,
at 8:41 PM, Itamar Reis Peixoto
ita...@ispbrasil.com.br wrote:
it's support SRTP
On Sun, Nov 22, 2009 at 7:21 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Do LInksys devices support TLS and SRTP that FS supports? 3102 at
least doesn't according to this post
.
On Sun, Nov 22, 2009 at 5:35 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
HI All,
Has anyone got some recommendations on which ATA to buy that supports
TLS and SRTP?
Thanks!
--
Itamar Reis Peixoto
e-mail/msn/google talk/sip: ita...@ispbrasil.com.br
skype
HI All,
Has anyone got some recommendations on which ATA to buy that supports
TLS and SRTP?
Thanks!
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Hi!
How do I setup FS so that placing a call from an extension that only
support SRTP (1002) to an extension that only supports RTP (1000)?
I put this dialstring, from the wiki
http://wiki.freeswitch.org/wiki/Tls, into the users xml file under
directory/default
param name=dial-string
with the responce you have mentioned.
Sent from my iPhone
On 19/11/2009, at 1:36 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How do I setup FS so that placing a call from an extension that only
support SRTP (1002) to an extension that only supports RTP (1000)?
I put
HI All,
pretty basic question and I feel a bit stupid asking this, but what
are the prerequisites for the INFO to be displayed when action
application=info/ is called in a dialplan?
ie are there requirements on the loglevel, does the INFO command have
to be put at a certain place in the dialplan
) 664-1044 x200
mr...@avgs.ca
On 17-Nov-09, at 9:53 PM, Mark Campbell-Smith wrote:
HI All,
pretty basic question and I feel a bit stupid asking this, but what
are the prerequisites for the INFO to be displayed when action
application=info/ is called in a dialplan?
ie
, if its at warning, you
wont see anything below warning ANYWHERE (console/event socket log
files)
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 17-Nov-09, at 10:05 PM, Mark Campbell-Smith wrote:
I had console
Hi!
I am trying to enable SSL support in FS. I have followed the wiki at
http://wiki.freeswitch.org/wiki/SIP_TLS
I already had libssl-dev installed, so I thought support should
already have been compiled into FS, however enabling
Internal_ssl_enable=true in vars.xml results in FS internal
it appears in.)
make[6]: *** [mod_dingaling.lo] Error 1
Anyone know which package should be installed so that TLS works on Debian?
On Tue, Nov 17, 2009 at 10:05 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I am trying to enable SSL support in FS. I have followed the wiki at
http
Hi!
I have FS natted and am connecting with an 'external' extension that
is registered to FS. ie the extension 2000 is registered on the
internet with a public IP through my router to FS (192.168.1.120 IP
address). uPnP works and I see that the extension is registered
successfully.
The problem
Campbell-Smith wrote:
Hi!
I have FS natted and am connecting with an 'external' extension that
is registered to FS. ie the extension 2000 is registered on the
internet with a public IP through my router to FS (192.168.1.120 IP
address). uPnP works and I see that the extension is registered
,
MIke
On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:
Hi Mike,
I should have put that in also.
I do have external_rtp_ip set in my config. I have it set to my
domain name:
X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/
I should also mention that if I use flaphone.com
t=0 0
m=audio 21234 RTP/AVP 0 2 9 8 101 13
Why would this be? I thought auto-nat was meant to solve these issues?
However, I still do not see the TRYING or RINGING messages ideas
appreciated.
Thanks!
On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote
too?
Regards,
JM
On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi again,
Actually, changing the param name=ext-rtp-ip value=auto-nat/ to
param name=ext-rtp-ip value=$${external_sip_ip}/ means that I
now see the IP address in the INVITE message
else that caused this
issue? I am using FreeSWITCH Version 1.0.trunk (15126)
On Mon, Nov 9, 2009 at 2:40 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Is there a way to determine if FS has detected nat? I am behind UPnP
and I can see on the router the mappings for Freeswitch.
2009/11
I think the (exported) means you don't have the latest svn, but probably the
officially released build 1.0.4 that can be downloaded from the FS page.
I think you should see something like (the latest trunk is 15203):
freeswi...@internal version
FreeSWITCH Version 1.0.trunk (15126)
I guess you
Hi!
How do I do a NOT equal to in a dialplan expression
Normaly in regex I would use the ! character. This doesn't seem to work in FS..
ie
condition field=${variable} expression=!^1
Shouldn't that match when the variable is not starting with one?
Can't you use the inline statement to set a variable so that it can be
used directly in a condition?
http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
On Thu, Oct 22, 2009 at 3:08 PM, Ahmed Munir ahmedmunir...@gmail.com wrote:
Hi,
Thanks for reply, it really helped me. One more
templates should be changed to handle this?
On Fri, Jul 10, 2009 at 8:57 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
1. Can I email the voicemail message to multiple email addresses?
A comma separated list does not work in the extensions.xml file
(1000.xml), but it does work if I
recent trunk. That
being said, read is not being run inline, so the set is actually being
run before digits_dialed is set. You will most likely need to use
transfer in this situation.
Mike
On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:
Hi!
I simply want to validate the dtmf
Hi!
I simply want to validate the dtmf digits I read from a user.From
the wiki, it appears I need to use inline=true when setting the
variable so it can be used directly within the same extension.
What have I done wrong below? I have tried many different
alternatives, but the second
On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:
Thanks Brian. Is this something that is planned to be implemented?
The workaround is to set the stun server also in the dingaling
configuration, but as I said, for some reason the stun times for me
out occasionally with dingaling
time.
as a workaround in googletalk any string that starts with + typed in the
chat box is treated as dtmf by FS
e.g. +1
Once the jingle spec stops being a moving target we will re-investigate
making sure its supported properly.
On Thu, Oct 15, 2009 at 5:19 PM, Mark Campbell-Smith
Hi!
I was wondering if the Dingaling implementation in FS supports DTMF?
This is now supported in the jingle specs (
http://www.jabberforum.org/showthread.php?t=2709 ), even though Google
Talk client does not currently support DTMF.
If not, are there plans to implement this?
Thanks!
...@freeswitch.org wrote:
I don't think mod_dingaling will do a lookup for host: like sofia will
as it doesn't have the code for that last I checked... I could be
wrong but I don't recall it doing that.
/b
On Oct 13, 2009, at 3:02 PM, Mark Campbell-Smith wrote:
I have a hostname set in vars.conf.xml
Hi!
I am trying to call from FS to gtalk. This used to work, so not sure
if there is a problem with my build (FreeSWITCH Version 1.0.trunk
(15126))
freeswi...@internal dingaling status
--DingaLing status--
login | connected
mygmai...@gmail.com/gtalk| AUTHORIZED
It looks okay
/internal_nat/1...@192.168.1.120) Ended
2009-10-13 21:33:05.807529 [NOTICE] switch_core_session.c:1089 Close
Channel sofia/internal_nat/1...@192.168.1.120 [CS_DESTROY]
On Tue, Oct 13, 2009 at 9:16 PM, Jason White ja...@jasonjgw.net wrote:
Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
When I dial
I've fixed the problem.
My dialplan for outbound calling had a typo:
action application=bridge
data=dingaling/gtalk/mygmai...@gmail.com/
The gtalk was gtallk somehow .
On Tue, Oct 13, 2009 at 8:56 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I am trying to call from
Hi!
I have a hostname set in vars.conf.xml for the parameters
external_rtp_ip and the external_sip_ip instead of the usual stun. I
found that stun was timing out and was causing some problems. And as
I have a hostname, it makes sense to use that instead of relying on
stun.
However, when I use
This is what I have in my dialplan and the fax is detected
beautifully. Note that in my case, extension 1000 will ring for a
second or two before the fax is detected. So in your example, the fax
does not have time to be detected, the dialplan exists and the call is
hungup.
When the fax is
on below link,
http://www.voip-info.org/wiki/view/STUN
Thank you.
On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Anyone have this issue?
On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I have just started
HI all,
I just tried to update to the latest svn and I get these errors right
at the end after issuing a 'make current'. I am using Debian Lenny.
making all mod_fax
make[5]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax'
make[6]: Entering directory
of modules
wouldn't load. You'll want to get it resolved before installing. I
ended up moving the existing source aside and re-checked out the
trunk, which compiled fine.
On Oct 8, 2009, at 10:06 PM, Mark Campbell-Smith wrote:
HI all,
I just tried to update to the latest svn and I get
will just get hung up.
Mike
On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
Hi
I was hoping someone could help me to setup the fax detection / tone
detection application.
I want to be able to transfer an incoming fax to a specific extension.
In my default.xml file, I have
is
the timeout, try making that longer, or doing the tone_detect in
execute_on_answer
Mike
On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
Thanks for the response Mike,
I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm
Further playing around and everything is working fine (even the
emailing). I'm not sure what I changed though to document it.
cheers
/M
On Mon, Oct 5, 2009 at 12:03 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi
I was hoping someone could help me to setup the fax detection
Hi
I was hoping someone could help me to setup the fax detection / tone
detection application.
I want to be able to transfer an incoming fax to a specific extension.
In my default.xml file, I have the following (extracted):
extension name=1000
condition field=destination_number
Anyone have this issue?
On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I have just started to use dingaling again, and noticed I constantly
get a stun error.
2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
stun.fwdnet.net:3478
Hi!
I have just started to use dingaling again, and noticed I constantly
get a stun error.
2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
stun.fwdnet.net:3478 [Remote Address Error!]
I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers
and keep getting this
What about this one for Debian...
http://wiki.freeswitch.org/wiki/Freeswitch_init
On Mon, Sep 28, 2009 at 9:15 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Yes. I have seen the scripts. But I could not find a suitable one for
Ubuntu.
Thank you.
LLoyd
2009/9/27 João
this script ... but it is not working. Did you try ?
On Sun, Sep 27, 2009 at 8:12 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
What about this one for Debian...
http://wiki.freeswitch.org/wiki/Freeswitch_init
On Mon, Sep 28, 2009 at 9:15 AM, Aloysius Thevarajah Lloyd
lloyd.aloys
trunk
and re-test. Get a pcap of the traffic (SIP and RTP) for review and then
report back.
Thanks,
MC
On Sun, Sep 13, 2009 at 2:34 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I have just experienced some call drops and each time the sequence is
the same
dialplan look at 5900 for park and 5901 for unpark.
/b
On Sep 17, 2009, at 7:58 AM, Mark Campbell-Smith wrote:
I am trying to create a simple call waiting dialplan as my phone does
not have Recall button.
___
FreeSWITCH-users mailing list
Hi!
I have just experienced some call drops and each time the sequence is
the same in the freeswitch.log file. Both parties are sure that they
did not accidentally hit the 3 button to send the DTMF tone (and the
same thing has happened four times already after ~5 minutes).
2009-09-13
Hi!
Where in the dialplan does FS decide which context is used for
processing.. I am dialing an outbound call but the call is being
processed in context public and not default?
mod_dialplan_xml.c:315 Processing Extension1000-dest number in context public
Why is FS choosing the public context
cmd=set data=external_ssl_enable=false/
X-PRE-PROCESS cmd=set data=external_ssl_dir=$${base_dir}/conf/ssl/
It is simple :P
On Thu, Sep 10, 2009 at 1:39 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
Where in the dialplan does FS decide which context is used
Hi Alan,
I hope you find your answers here as these are the sort of things that
are hard to find on the wiki, which is somewhat outdated in areas. If
you do find your answers, please post them back here for everyone
else.
I am new to FS also, so my comments below may not be 100% correct!
1.
Hi All,
I know this question has come up before but I couldn't find the answer
that I could understand! Sorry in advance.
My setup is:
Freeswtch NAT'd (192.168.x.x) - Router - Internet - Softphone with public IP
I can easily get the softphones to register, but when I try to call
from the
to achieve this?
On Sat, Jul 11, 2009 at 2:04 AM, Michael Jerrism...@jerris.com wrote:
could you post how you tired to do it in dialplan that didn't work?
Mike
On Jul 10, 2009, at 5:57 AM, Mark Campbell-Smith wrote:
Hi!
1. Can I email the voicemail message to multiple email addresses?
A comma
a voicemail is left?
api Hangup hook?
I g
From: Brian West br...@freeswitch.org
On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote:
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
Thanks!
Hi!
My call dropped and I saw this error in the syslog:
Jun 24 17:05:04 freeswitch kernel: [157531.309017] freeswitch[4621]:
segfault at c ip b73b2a42 sp b72a3840 error 4 in
mod_sofia.so[b7369000+16c000]
How can I get more information on this fault to file a bug report?
Thanks!
Hi!
I am trying to call from my corporate network (firewalled) using Gtalk
to Freeswitch. I am not getting any audio.
In the logs I see that mod_dingaling is using my internal corporate IP
address which is not publically addressable.
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634
internally as wan.auto
so you can use that
instead of making one in your config.
On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I am trying to call from my corporate network (firewalled) using Gtalk
to Freeswitch. I am not getting any audio
To: freeswitch-users@lists.freeswitch.org
Message-ID: 7c7a8ed9-eced-4100-87f6-0875c054e...@freeswitch.org
Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings
/b
On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote:
Hi
Hi!
I am trying to email from
2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore
original codec.
2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to
1...@192.168.0.20
/bin/cat: write error: Broken pipe
sh: line 1: 11975 Done(1) /bin/cat
Hi!
I have a problem with voicemail in that freeswitch fails to let users
leave their message. Something wrong in the config I guess. I see
this in the logs:
2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No
language specified - Using [en]
2009-06-21 11:28:32.238744 [DEBUG]
Hi!
Does freeswitch support extracting the billing data (PortaBilling) in
SIP messages? If so, is there anyway I can get that information to an
extension?
03:36:00.245: //-1//SIP/Msg/ccsipDis playMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4
Hi!
How can I configure voicemail so that I do not get the options such as
record your message at the tone and mark this message as urgent
Thanks!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Hi!
Is it possible to configure freeswitch and mod_dingaling so that it
sends a text chat to a gtalk client?
I would plan to use this for certain debugging purposes.
Thanks!
/Mark
___
Freeswitch-users mailing list
Hi!
I need to enable debug mode in dingaling as I can't see that freeswitch is
coming online in gtalk.
I have changed the following:
changed the loglevel to debug in console.conf.xml
changed the debug level to 1 in dingaling.conf to 1
I do not see any xmpp logs in the console or in
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