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Hi,
maybe this helps:
http://www.wesip.com/mediawiki/index.php/SipSpy
regards
helmut
On 17.05.2009 20:33, Ron McCarthy wrote:
Kokoska,
Did you ever find a solution for this? I have been working on this as
well, trying to write some perl
Ok, my FS box is on a LAN IP (10.0.0.12), but it is setup as the DMZ
machine on my modem so it can receive incoming connections without any
NAT related problems.
I'm trying to get a user outside on the internet to connect to my FS
box and register as an internal user. He is using X-Lite on
Hello,
I'm in Spain with an analog TDM400 Clone from OpenVox (with 1xFXO
+1xFXS), and am trying first to make the FXO work with Openzap and
Freeswitch (using dahdi 2.2.0-rc4). Openzap perfectly detects and loads
the spans, but I'm currently enable to dial out with the FXO module, it
doesn't dial
David Robinson pawzl...@gmail.com wrote:
Is this correct ? Am I missing something fundamental ?
My suspicion is that the RTP traffic isn't traversing the NAT properly. You
may have to configure the routers at both ends to forward the RTP packets to
the correct destinations. There is a good
Hi Adam,
We had exactly the same problem which we initially believed to be a NAT
firewall issue. However, I changed my firewall to transparent mode and the
problem still persisted. In the end, I solved the problem by changing VOIP
provider. I was using AQL which I couldn't make work and now use
Can you post the INVITE and 200 OK messages from your mates end of the
call. Even if you forward the ports on the router, the RTP will not
traverse correctly if the advertised IP address is an internal one for
both ends.
On Mon, May 18, 2009 at 6:20 PM, David Robinson pawzl...@gmail.com wrote:
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Hello,
does anybody know if and how FS can export its core db to an external
database via odbc like mod_limit or mod_sofia? If not, is such a feature
planned for the near future?
regards
Helmut
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No, you can't.
Math
On 18-May-09, at 2:41 PM, Helmut Kuper wrote:
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Hello,
does anybody know if and how FS can export its core db to an external
database via odbc like mod_limit or mod_sofia? If not, is such a
feature
planned for the near
enable the b leg cdr as well and you will also get cdr from the b leg
perspective.
both xml cdr and cdr csv have params in the config to enable it.
On Sun, May 17, 2009 at 1:26 PM, Ron McCarthy ronmc...@gmail.com wrote:
Hi list,
Ive been trying to find a way to log 503's, 480's and other SIP
Hi there,
Freeswitch looks really interesting. I am trying to connect a softswitch
wich does some strange things. If a call arrives from pots the
softswitch won't send sdp information in the initial invite.
Is this something I can change in the sip-profile with freeswitch?
BR
Uwe
Hi there,
Freeswitch looks really interesting. I am trying to connect a softswitch
wich does some strange things. If a call arrives from pots the
softswitch won't send sdp information in the initial invite.
Is this something I can change in the sip-profile with freeswitch?
BR
Uwe
Hi list,
I just discovered FS (practiced a bit * 2 years ago, but too much unstable)
and find it cool, NOT CPU greedy and (almost) working ouf of the web.
I'd like to know if star codes (such as *98) are normalized or not?
(and if so, where I could find a list)
Also, as I don't use very much my
Hi,
allowing 3cc fixes the problem.
BR
Uwe
Uwe Kastens schrieb:
Hi there,
Freeswitch looks really interesting. I am trying to connect a softswitch
wich does some strange things. If a call arrives from pots the
softswitch won't send sdp information in the initial invite.
Is this
Hello Jean-Yves,
did you ever try a call-trough? (a person dials in (1234567, see below)
types the target number as DTMF and gets connected to this number?
A basic dialplan can be like this:
extension name=Dialthru
condition field=destination_number expression=^(1234567)$
action
Hi,
Is there any reason why the crash-protection parameter in switch.conf.xml
defaults to false and are there any downsides to setting it to true? The
documentation says it helps with certain types of crashes, can anyone tell
me what sort of crashes in particular it helps to prevent as my
Hi,
Crash protection catches segmentation faul signals and try to kill
that thread only. It works for stupid errors like a null pointer
dereference, but in most scenarios, a crash means something in the
process memory was corrupted. Ignoring it will just make it crash
later on, thats why
bridge() appears to be ignoring the absolute_codec_string channel variable
defined in the User Directory even though info shows that it is present.
Other variables, such as effective_caller_id_number seem to behave
correctly which leads me to believe that this may be a very minor bug.
In order
Because by the time it gets here... the codec is already picked..
you'll have to turn on late neg. for this to work.
/b
On May 18, 2009, at 1:08 PM, Metik wrote:
variable name=absolute_codec_string value=PCMU/
Brian West
br...@freeswitch.org
-- Meet us at ClueCon!
Oddly enough, I initially though that was the problem and enabled it without
any success...
freeswi...@noesis.metik.com sofia status profile internal
API CALL [sofia(status profile internal)] output:
Even the b leg cdr is enabled it only remember the final state(channel
vars) on the b leg.
At least there are two possible ways to keep tracking all the gateways:
1) don't use '|' separated dial string, use a lua script like this:
session:execute(bridge, dial_string1);
absolute_codec_string needs to be available from the B-leg too so it
can be used on outbound channels.
Add that to your directory entry and it should work.
variable name=export_vars value=absolute_codec_string /
Math
On 18-May-09, at 9:18 PM, Brian West wrote:
Are you authenticating phone
you for sure need late negotiation:
also:
You are only setting the variable on the inbound leg but not the outbound
leg.
Remember there are 2 separate channels here.
Try this in the same place you are setting the caller id in your broken
example:
!-- this will allow absolute_codec_string to be
On register, sometimes my voip client got SIP/2.0 482 Request merged
sometimes got 200 ok.
482 also means loop detected. my client only has one account logged in
only one place, and no proxy, can I take 482 as 200 OK?
Thanks.
from RFC 3261:
8.2.2.2 Merged Requests
If the request has
=D
That's another way I didn't mention.
There are 2 more but they are more complicated so I will omit them ;)
On Mon, May 18, 2009 at 2:25 PM, Mathieu Rene mrene_li...@avgs.ca wrote:
absolute_codec_string needs to be available from the B-leg too so it can be
used on outbound channels.
Add
Is this in regards to FreeSWITCH or something else you're writing?
/b
On May 18, 2009, at 2:34 PM, dujinfang wrote:
On register, sometimes my voip client got SIP/2.0 482 Request merged
sometimes got 200 ok.
482 also means loop detected. my client only has one account logged in
only one
Math,
That was it--thank you very much!
-Metik
- Original Message -
From: Mathieu Rene
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, May 18, 2009 3:25 PM
Subject: Re: [Freeswitch-users] User Directory and Per-user(Channel)variables
absolute_codec_string needs
Yes, FS(13263) send out 482 request merged to my voip client.
I guess, for some reason, FS doesn't respond to the REGISTER, and when
the client start REGISTER again using another call-id, it merged the
request to one. Anyone ever met this before? See the call-id and cseq
below :
recv 631
Please show me a pcap file to may email address because I can bet you
FS/Sofia is doing it right 99% of the time.
/b
On May 18, 2009, at 6:55 PM, dujinfang wrote:
es, FS(13263) send out 482 request merged to my voip client.
I guess, for some reason, FS doesn't respond to the REGISTER, and
I believe.
capturing on tshark -i eth1 -w register.pcap udp port 5090
do you have further suggestions on the tshark filter?
Thanks.
On May 19, 2009, at 8:06 AM, Brian West wrote:
Please show me a pcap file to may email address because I can bet
you FS/Sofia is doing it right 99% of the
I never use tshark to capture... I use tcpdump -s0 -x port 5090 -w
file.pcap
/b
On May 18, 2009, at 7:39 PM, dujinfang wrote:
I believe.
capturing on tshark -i eth1 -w register.pcap udp port 5090
do you have further suggestions on the tshark filter?
Thanks.
Brian West
My suspicion is that the RTP traffic isn't traversing the NAT
properly. You
may have to configure the routers at both ends to forward the RTP
packets to
the correct destinations. There is a good discussion of NAT on the
wiki.
Situation: FS (10.0.0.12) - DMZ (124.254.81.250) -
I am not receiving any ringback when calling in from an iPhone. I receive
ringback when calling from other cell phones and land lines just not an
iPhone.
2009-05-18 20:28:04 [DEBUG] switch_core_state_machine.c:152
switch_core_standard_on_execute() sofia/external/3153836...@64.24.35.76
Execute
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