1. I think I got sipviewer working so once I find that info, I'll let
you know. but I can't imagine it would be this, since direct calls
follow the rules correctly.
2. I'll check the staged dialing feature on the grandstream.
These users are regular SIP users (from snom phones)
And...you know wa
I've compiled and installed sipx 3.10 from svn on rhel4.
It seems to be mostly working, and when I start it from the init script
it seems happy.
Checking TLS/SSL configuration:[ OK ]
Checking Firewall configuration: [ OK ]
Chec
It's most likely one of two things:
1. There is a problem with your rules and the call is not being forwarded
properly by sipx. A call trace from sipx would be very helpful.
2. The gateway is not handling the forwarding properly (i.e. I think the
grandstream has a strange way of handling "stage
1. subject modified to reflect what the need is.
2. the scenario is this:
Ext 343 has calls forwarded to 555-1234 (example).
Ext 336 dials ext 343. gets failed call. Gateway trace is
listed below.
Every ext has mobile turned ON.
3. version: sipXconfig (3.10
On Wed, 2008-07-02 at 14:44 -0700, IT Services wrote:
>
> So somewhere along the line the call forwarding is not modifying the
> sip
> address from [EMAIL PROTECTED] To [EMAIL PROTECTED]
It should be modified in the request uri - the second token on the first
line of the SIP message (immediatel
X on linux is not the same as a gui on windows. X is an environment. Someone
once posted a way to do this in Windows, I'm not sure how to do that.
The assumption is, and I might be incorrect, that the user has a linux machine
with a GUI installed, with which this can be viewed without using the
1. Please use a subject that is consistent and helps indicate your need. I'm
sure it's an oversight, but it really does help.
2. The gateway is forwarding calls from 336 to 343 (an internal number).
So look at your forwarding rule for user 336, make sure they have call
permissions to MOBILE
Here is a trace from the gateway:
Wed Jul 02 14:28:26 2008: <14>GS_LOG:
[00:0B:82:13:E9:6A][000][FF51][01000102] 687 sip.c Sess: 0 Send SIP
message: 100 To 192.168.1.49:5060
Wed Jul 02 14:28:26 2008: <14>GS_LOG:
[00:0B:82:13:E9:6A][000][FF51][01000102] 568 sip_handle_invite.c Sess: 3
INVITE From="
On Wed, 2008-07-02 at 13:53 -0700, IT Services wrote:
> and I am getting the error below:
>
> sipx-trace
> /usr/bin/sipx-trace: line 133: [: `)' expected, found (
Exactly what revision of the code are you using? Is /usr/bin/sipx-trace
installed from an RPM?
Can you do:
bash -x /usr/bi
On Wed, 2008-07-02 at 13:59 -0700, IT Services wrote:
> External calls via the PSTN gateway are perceptibly lower in audio
> volume than internal calls. I can understand that this may happen since
> calls are routed to an external device.
Since the audio is always in digital form, this is unlikely
Is Bridge Line Appearance on the roadmap? I know there are different
ways to "fake" it with call parking but this has to be one of the most
requested features people ask me about.
Thanks,
Max
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Hi all:
I am using sipx 3.10 with PSTN gateways (grandstream).
External calls via the PSTN gateway are perceptibly lower in audio
volume than internal calls. I can understand that this may happen since
calls are routed to an external device.
Is there a way to increase the audio volume of PSTN ca
Hi all,
I am trying to get sip traces. I have followed
http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Si
pviewer.
and I am getting the error below:
sipx-trace
/usr/bin/sipx-trace: line 133: [: `)' expected, found (
searching / sipXproxy log
searching / sipregistrar log
t
Bandtel is my ITSP, yes. I think the problem is exasperated by the
fact that my server hosting
provider has sophisticated multi-carrier routing as well.
Dale Worley wrote:
> On Tue, 2008-07-01 at 18:02 -0400, Michael Cote wrote:
>
>> If they
>> arrive at the ITSP out of order (which they do
On Wed, 2008-07-02 at 15:57 +0530, VG wrote:
> Can I change the default listning port of SipxProxy from 5060 to 5061.
> Please tell me the file in which i have to make changes.
The listening port is controlled by these lines
in /etc/sipxpbx/config.defs:
PROXY_SERVER_SIP_PORT=5060
PROXY_SERVER_SIP
On Tue, 2008-07-01 at 17:50 -0700, IT Services wrote:
> I'm having a problem with call forwarding.
> I'm stuck right now so any suggestion would be helpful!
You will have to get a call trace of a failed call and see what is going
wrong.
Dale
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On Tue, 2008-07-01 at 18:02 -0400, Michael Cote wrote:
> If they
> arrive at the ITSP out of order (which they do about 50% of the time)
> then the call fails.
If I recall correctly, it is Bandtel that has this problem?
Dale
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If i were you, i would find out where exactly (at which device) "out of
order fragmented packets" occur. And make ISP to fix the problem. Or change
ISP, which is often impossible :(.
> -Original Message-
> From: Michael Cote [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, July 02, 2008 5:01 P
On Tue, 2008-07-01 at 18:02 -0400, Michael Cote wrote:
> I'm running into a packet fragmentation problem on my OK response to an
> INVITE from an ITSP.
> It appears that the OK response is just over the limit and is getting
> split into two IP packets. If they
> arrive at the ITSP out of ord
I agreed that it is unusual for fragmented packets to arrive out of
order until I was told that my ISP (hosting sipX)
has an intelligent network architecture which provides optimum
connectivity to multiple carriers coming out of the
hosting company. Hearing "intelligent" I got suspicious.
Nik
Please:
* don't reply to an existing posting when starting a new subject
* put a subject line on your mail that indicates what your mail is
about
On Tue, 2008-07-01 at 17:50 -0700, IT Services wrote:
> Hey there:
>
> I'm having a problem with call forwarding.
>
> If I list
Is the call forwarding for a user? What version of sipx? Typically you would
dial the PSTN number the same way you would as if it were an internal number
(1xxxyyy) and it should follow the rules of the dial plan, etc.
The message sounds like it's a TELCO recorded message because it didn't li
Hi all,
Can I change the default listning port of SipxProxy from 5060 to 5061.
Please tell me the file in which i have to make changes.
Also Can i configure sipx to learn (for specific user) on 5061?
Regards,
Vikas
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Michael,
Just an idea why fragmented packets may arrive out of order:
It may happen that some device (router or switch) in the transport network
has QOS enabled.
This may result in "small packets first" or something like that...
Nikolay.
> -Original Message-
> From: [EMAIL PROTECTED] [ma
Hi,
someone knows if some company develops blind operator console sipXecs
compliant?
Tnx
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Michael,
> I'm running into a packet fragmentation problem on my OK response to an
> INVITE from an ITSP.
> It appears that the OK response is just over the limit and is getting
> split into two IP packets. If they
Indeed, your 200OK message is 1511 bytes length. So, taking into account
that u
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