Daniel hi,
Now it works!
I'm adding in the rpid avp the value that the RFC it says $avp(s:rpid) =
$fu; and then
append_rpid_hf($fU,
;party=calling;id-type=subscriber;privacy=off;screen=no);
In this way I see the headers added correctly in sip message. Considering
what you said about the from
Hello,
On 2/3/10 10:43 AM, alex pappas wrote:
Daniel hi,
Now it works!
I'm adding in the rpid avp the value that the RFC it says $avp(s:rpid)
= $fu; and then
append_rpid_hf($fU,
;party=calling;id-type=subscriber;privacy=off;screen=no);
In this way I see the headers added correctly in sip
Hello,
what version are you using? Do you have big traffic on it?
Usual delays are coming from:
- slow dns
- slow database
- lot of debugging with non-asynchronous syslog
- bad config file
You can use benchmark module to get executing time of parts in
configuration file. If you can share
Hello all,
two questions regarding alias_db:
1) How do I make use of the append_branches function of alias_db? Trying to
insert two or more aliases is not possible in the current db layout (
UNIQUE KEY `alias_idx` (`alias_username`,`alias_domain`) ). The idea was to
enable simple db-based
Hello all,
I've run into an issue of Mar 2009:
http://lists.kamailio.org/pipermail/users/2009-March/022143.html
This seems to be unresolved in v1.5.4, the then presented work-around via
modparam(pua_dialoginfo, override_lifetime, 600)
still works fine. The question is whether this should work
Hello,
On 2/3/10 2:55 PM, Martin Koenig wrote:
Hello all,
I've run into an issue of Mar 2009:
http://lists.kamailio.org/pipermail/users/2009-March/022143.html
This seems to be unresolved in v1.5.4, the then presented work-around via
modparam(pua_dialoginfo, override_lifetime, 600)
still
On 2/3/10 5:11 PM, Daniel-Constantin Mierla wrote:
Hello,
On 2/3/10 2:55 PM, Martin Koenig wrote:
Hello all,
I've run into an issue of Mar 2009:
http://lists.kamailio.org/pipermail/users/2009-March/022143.html
This seems to be unresolved in v1.5.4, the then presented work-around
via
Hello all,
what is the best edit tool to see quickly the right syntax ... ?
which language to choose?
___
Kamailio (OpenSER) - Users mailing list
Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
On 02/03/2010 11:19 AM, BERGANZ Francois wrote:
what is the best edit tool to see quickly the right syntax ... ?
What syntax?
which language to choose?
For what?
--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1 678-954-0670
Direct : +1 678-954-0671
Web:
Hello,
On 2/3/10 2:46 PM, Martin Koenig wrote:
Hello all,
two questions regarding alias_db:
1) How do I make use of the append_branches function of alias_db? Trying to
insert two or more aliases is not possible in the current db layout (
UNIQUE KEY `alias_idx`
To see quickly where an 'if' start and stop, to see if I have a double
which isn't close ...
-Message d'origine-
De : users-boun...@lists.kamailio.org
[mailto:users-boun...@lists.kamailio.org] De la part de Alex Balashov
Envoyé : mercredi 3 février 2010 17:22
À :
In which language?
On 02/03/2010 11:25 AM, BERGANZ Francois wrote:
To see quickly where an 'if' start and stop, to see if I have a double
which isn't close ...
-Message d'origine-
De : users-boun...@lists.kamailio.org
[mailto:users-boun...@lists.kamailio.org] De la part de Alex
Francois-
To see quickly where an 'if' start and stop, to see if I have a double
which isn't close ...
You might try Programmer's Notepad (pnotepad.org,
http://en.wikipedia.org/wiki/Programmer's_Notepad)... the source
language syntax highlighting and checking is really outstanding. It's a
I have set it up the Radius for Kamailio, it works fine, until the
Radius server goes down it did not go down for a while, when tis
happes the Kamailio server stops processing any calls.
I do not think is a normal behaviour, probably a bug, can anybody
please tell me if this is expected
This may be an unavoidable problem with radiusclient-ng.
On 02/03/2010 11:29 AM, Omar wrote:
I have set it up the Radius for Kamailio, it works fine, until the
Radius server goes down it did not go down for a while, when tis happes
the Kamailio server stops processing any calls.
I do not think
On Wednesday 03 February 2010, Omar wrote:
I have set it up the Radius for Kamailio, it works fine, until the
Radius server goes down it did not go down for a while, when tis
happes the Kamailio server stops processing any calls.
I do not think is a normal behaviour, probably a bug, can
El Miércoles, 3 de Febrero de 2010, BERGANZ Francois escribió:
Hello all,
what is the best edit tool to see quickly the right syntax ... ?
which language to choose?
I assume you mean for kamailio.cfg syntax, right?
Then AFAIK there is a vim syntax.
In my case I use Kate editor (KDE)
Hello,
Does anyone knows if Kamalito supports MESSAGE method where I can find
the documentation or API's ?
Regards,
Abdul Hakeem
attachment: winmail.dat___
Kamailio (OpenSER) - Users mailing list
Users@lists.kamailio.org
Kamailio supports any SIP method, standard or invented.
On 02/03/2010 12:16 PM, Abdul Hakeem wrote:
Hello,
Does anyone knows if Kamalito supports MESSAGE method where I can find
the documentation or API's ?
Regards,
Abdul Hakeem
___
Kamailio
El Miércoles, 3 de Febrero de 2010, Abdul Hakeem escribió:
Hello,
Does anyone knows if Kamalito supports MESSAGE method
Kamailio is a proxy. A proxy can handle any kind of method, as CHICKEN.
--
Iñaki Baz Castillo i...@aliax.net
___
Kamailio
Thanks for the reply, is there anywhere I can check for this provision (
RFC3428 )in the source code ?
Cheers,
AH
-Original Message-
From: users-boun...@lists.kamailio.org
[mailto:users-boun...@lists.kamailio.org] On Behalf Of Iñaki Baz Castillo
Sent: Wednesday, February 03, 2010 5:22
There is no provision.
The provision is:
route {
...
if(uri == myself) {
if(is_method(MESSAGE)) {
# Handle however you want.
}
}
}
On 02/03/2010 12:35 PM, Abdul Hakeem wrote:
Thanks for the reply, is there anywhere I can check for this provision (
RFC3428
El Miércoles, 3 de Febrero de 2010, Abdul Hakeem escribió:
Thanks for the reply, is there anywhere I can check for this provision (
RFC3428 )in the source code ? Cheers,
Not sure what you watn to achieve. MESSAGE is just a method to send text/html.
It's supported out of the box by Kamailio
On 02/03/2010 12:39 PM, Iñaki Baz Castillo wrote:
El Miércoles, 3 de Febrero de 2010, Abdul Hakeem escribió:
Thanks for the reply, is there anywhere I can check for this provision (
RFC3428 )in the source code ? Cheers,
Not sure what you watn to achieve. MESSAGE is just a method to send
Abdul-
Thanks for the reply, is there anywhere I can check for this
provision ( RFC3428 )in the source code ?
I've been in the telecom industry for 20+ years and there is I've never seen a
situation where you can check the
source code to verify support/ interoperability / compatibility with
Actually I had Kamalito Sip or Kamalito SMPP in mind. I understand there
are some people working with Asterisk SMPP.
I was thinking this might be achievable with Kamalito to route sms/text
messages in sip to sip or sip to smpp.
Cheers,
AH
-Original Message-
From:
I'm not sure I understand your scenario, but, it doesn't sound like
something you'd want to do with Kamailio.
On 02/03/2010 12:51 PM, Abdul Hakeem wrote:
Actually I had Kamalito Sip or Kamalito SMPP in mind. I understand there are
some people working with Asterisk SMPP.
I was thinking
P.S. What the hell is Kamalito, and why is it in the subject line?
On 02/03/2010 12:53 PM, Alex Balashov wrote:
I'm not sure I understand your scenario, but, it doesn't sound like
something you'd want to do with Kamailio.
On 02/03/2010 12:51 PM, Abdul Hakeem wrote:
Actually I had Kamalito
On Tuesday 02 February 2010, Ovidiu Sas wrote:
Having both method of initialization will just complicate the code.
When the module parameters are parsed, the shared memory is not
available and therefor all the module rate limit parameters will need
to be cached into the memory until the shared
No need to swear ! it was my outlook playing up.
-Original Message-
From: users-boun...@lists.kamailio.org
[mailto:users-boun...@lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Wednesday, February 03, 2010 5:55 PM
To: users@lists.kamailio.org
Subject: Re: [Kamailio-Users]
Hello,
On 2/3/10 5:41 PM, Iñaki Baz Castillo wrote:
El Miércoles, 3 de Febrero de 2010, BERGANZ Francois escribió:
Hello all,
what is the best edit tool to see quickly the right syntax ... ?
which language to choose?
I assume you mean for kamailio.cfg syntax, right?
Then AFAIK
Kamailio is a proxy. A proxy can handle any kind of method, as CHICKEN.
Dear Sir Castillo,
Can you please REFER me to the correct RFC and ABNF for the CHICKEN method?
Thank you in advance,
Your truly
___
Kamailio (OpenSER) - Users mailing list
3 feb 2010 kl. 18.39 skrev Iñaki Baz Castillo:
El Miércoles, 3 de Febrero de 2010, Abdul Hakeem escribió:
Thanks for the reply, is there anywhere I can check for this provision (
RFC3428 )in the source code ? Cheers,
Not sure what you watn to achieve. MESSAGE is just a method to send
El Miércoles, 3 de Febrero de 2010, Olle E. Johansson escribió:
3 feb 2010 kl. 18.39 skrev Iñaki Baz Castillo:
El Miércoles, 3 de Febrero de 2010, Abdul Hakeem escribió:
Thanks for the reply, is there anywhere I can check for this provision (
RFC3428 )in the source code ? Cheers,
Not
El Miércoles, 3 de Febrero de 2010, Olle E. Johansson escribió:
Kamailio is a proxy. A proxy can handle any kind of method, as CHICKEN.
Dear Sir Castillo,
Can you please REFER me to the correct RFC and ABNF for the CHICKEN method?
Thank you in advance,
It's still a draft that updates
Hello everyone,
I'm just curious as to see what some of you guys do in regards to
handling a Re-Invite that comes back downstream to a NATTED UAC.
For example, call scenario:
UAC - Kamailio (Fix Nated Contact) - PSTN
Re-Invite Occurs:
PSTN - Kamailio - UAC
UAC (200 OK w/ NAT RFC1918
Handle the request and the reply with far-end NAT traversal detection
routines in the same way you would handle a non-sequential INVITE.
On 02/03/2010 03:05 PM, Brandon Armstead wrote:
Hello everyone,
I'm just curious as to see what some of you guys do in regards to
handling a Re-Invite
3 feb 2010 kl. 20.14 skrev Iñaki Baz Castillo:
El Miércoles, 3 de Febrero de 2010, Olle E. Johansson escribió:
Kamailio is a proxy. A proxy can handle any kind of method, as CHICKEN.
Dear Sir Castillo,
Can you please REFER me to the correct RFC and ABNF for the CHICKEN method?
Thank
El Miércoles, 3 de Febrero de 2010, Brandon Armstead escribió:
Hello everyone,
I'm just curious as to see what some of you guys do in regards to
handling a Re-Invite that comes back downstream to a NATTED UAC.
For example, call scenario:
UAC - Kamailio (Fix Nated Contact) - PSTN
El Miércoles, 3 de Febrero de 2010, Olle E. Johansson escribió:
And does the ROOSTER has an additional event type audio/wakeup?
Unfortunatelly IANA won't decide the exact mime-type name within the following
5 years so it's expected that each vendor will use its own name (i.e. Cisco
names it
Daniel,
Thank you for your response.
Daniel-Constantin Mierla wrote:
Hello,
what version are you using? Do you have big traffic on it?
I am using version 1.5, and there isn't much traffic.
Usual delays are coming from:
- lot of debugging with non-asynchronous syslog
This was a very useful
Hello,
I do see all the behavior as referenced, however the actual problem is upon
receipt of the invite to the UAC, in which it responds with 200 OK and Contact
of RFC1918 address, in which is not being fix natted contact because at this
point kamailio is not aware of the UAC being behind
On 02/03/2010 03:39 PM, bran...@cryy.com wrote:
the actual problem is upon receipt of the invite to the UAC, in which it
responds with 200 OK and Contact of RFC1918 address, in which is not being fix
natted contact because at this point kamailio is not aware of the UAC being
behind nat due to
Oh, wait. I think I understand what you're asking. Set a flag when the
initial INVITE is being processed and NAT is detected; this will show
up in the
When handling the initial INVITE from the customer set a flag. This
will show back up on re-INVITE. Works for me.
On 02/03/2010 03:43
El Miércoles, 3 de Febrero de 2010, Alex Balashov escribió:
Oh, wait. I think I understand what you're asking. Set a flag when the
initial INVITE is being processed and NAT is detected; this will show
up in the
When handling the initial INVITE from the customer set a flag. This
will
El Miércoles, 3 de Febrero de 2010, Alex Balashov escribió:
On 02/03/2010 03:39 PM, bran...@cryy.com wrote:
the actual problem is upon receipt of the invite to the UAC, in which it
responds with 200 OK and Contact of RFC1918 address, in which is not
being fix natted contact because at this
El Miércoles, 3 de Febrero de 2010, bran...@cryy.com escribió:
I do see all the behavior as referenced, however the actual problem is
upon receipt of the invite to the UAC, in which it responds with 200 OK
and Contact of RFC1918 address, in which is not being fix natted contact
because at
On 02/03/2010 04:04 PM, Iñaki Baz Castillo wrote:
El Miércoles, 3 de Febrero de 2010, Alex Balashov escribió:
Oh, wait. I think I understand what you're asking. Set a flag when the
initial INVITE is being processed and NAT is detected; this will show
up in the
When handling the initial
Thinking about this a little more, I am not sure I understand the issue,
perhaps because the original post is confusing.
Correct me if I'm wrong:
1) If initial INVITE is from NAT'd user to PSTN, when INVITE is received
by proxy its Contact is fixed up and it is passed along.
So, replies and
El Miércoles, 3 de Febrero de 2010, Alex Balashov escribió:
What about using add_rr_param() to indicate NAT somehow so that this
flag can be fished later out of the Record-Route header / Route set on
sequential requests and replies?
This is exactly what I use to determine if rtpproxy must be
El Miércoles, 3 de Febrero de 2010, Alex Balashov escribió:
Thinking about this a little more, I am not sure I understand the issue,
perhaps because the original post is confusing.
Correct me if I'm wrong:
1) If initial INVITE is from NAT'd user to PSTN, when INVITE is received
by proxy
El Miércoles, 3 de Febrero de 2010, Iñaki Baz Castillo escribió:
It could, but it's not required as target uri cannot change after the
dialog is established.
This is: upon dialog creation both UA's must send in-dialog requests to
the proxy (if it did loose routing) with the RURI pointing
On 02/03/2010 04:25 PM, Iñaki Baz Castillo wrote:
This is: upon dialog creation both UA's must send in-dialog requests to the
proxy (if it did loose routing) with the RURI pointing to the target uri (set
in the initial INVITE/200 by inspecting Contact headers). The Contact in a re-
INVITE or
El Miércoles, 3 de Febrero de 2010, Alex Balashov escribió:
On 02/03/2010 04:22 PM, Iñaki Baz Castillo wrote:
However this trick is not needed at all to fix just the signalling as the
Contact URI (so the dialog target uri) was already replaced in the
initial INVITE/200, and it cannot change
El Miércoles, 3 de Febrero de 2010, Brandon Armstead escribió:
Hello,
Please refer to sip trace: http://pastebin.org/86010.
On LINE 294, PSTN sends the Re-INVITE
And there is the error (in PSTN_PROXY server):
U +0.000104 MY_SBC_PROXY:5060 - PSTN_PROXY_IP:5060
SIP/2.0 200 OK.
To:
El Miércoles, 3 de Febrero de 2010, Alex Balashov escribió:
On 02/03/2010 04:44 PM, Iñaki Baz Castillo wrote:
This is a re-INVITE and the UA (MY_SBC_PROXY) replies a 200 with the
natted Contact (as usual).
Then the PSTN_PROXY server creates the ACK by setting such private SIP
URI as RURI.
Hi Brandon!
This is my pragmatic approach:
During dialog-creating transaction (INVITE, SUBSCRIBE) I decide if a
clients gets NAT-traversal* or not. Thus result will be stored in a
record-route cookie (add_rr_param()).Therefore, when dooing loose_route
I will check the content of this
Iñaki Baz Castillo wrote:
El Miércoles, 3 de Febrero de 2010, Alex Balashov escribió:
What about using add_rr_param() to indicate NAT somehow so that this
flag can be fished later out of the Record-Route header / Route set on
sequential requests and replies?
This is exactly what I use to
Iñaki Baz Castillo wrote:
El Miércoles, 3 de Febrero de 2010, Brandon Armstead escribió:
Hello,
Please refer to sip trace: http://pastebin.org/86010.
On LINE 294, PSTN sends the Re-INVITE
And there is the error (in PSTN_PROXY server):
U +0.000104 MY_SBC_PROXY:5060 - PSTN_PROXY_IP:5060
El Miércoles, 3 de Febrero de 2010, Klaus Darilion escribió:
Are you sure? IIRC the contact may change (the Route set must not change).
They call them target-refreshing requests:
12.2: ...
Requests within a dialog MAY contain Record-Route and Contact header
fields. However,
On 02/03/2010 05:48 PM, Iñaki Baz Castillo wrote:
El Miércoles, 3 de Febrero de 2010, Klaus Darilion escribió:
Are you sure? IIRC the contact may change (the Route set must not change).
They call them target-refreshing requests:
12.2: ...
Requests within a dialog MAY contain
Hello,
I was able to fix the problem, by enabling AHCI in the BIOS and
rebuilding initrd with the ahci driver. Instructions in link below
http://serverfault.com/questions/68793/enabling-ahci-in-rhel5-post-install
Current hdparm -t /dev/sda output
Timing buffered disk reads: 182 MB in 3.02
El Miércoles, 3 de Febrero de 2010, Klaus Darilion escribió:
This is a re-INVITE and the UA (MY_SBC_PROXY) replies a 200 with the
natted Contact (as usual).
Then the PSTN_PROXY server creates the ACK by setting such private SIP
URI as RURI. This is incorrect, this RURI *MUST* be the
El Miércoles, 3 de Febrero de 2010, Klaus Darilion escribió:
Are you sure? IIRC the contact may change (the Route set must not change).
They call them target-refreshing requests:
12.2: ...
Requests within a dialog MAY contain Record-Route and Contact header
fields. However,
On Feb 02, 2010 at 13:41, Ovidiu Sas o...@voipembedded.com wrote:
Hello Henning,
Having both method of initialization will just complicate the code.
When the module parameters are parsed, the shared memory is not
available and therefor all the module rate limit parameters will need
to be
This guy lives in my town. I've googled him, so I'll try to call him and
give him a heads-up on the matter.
If we get no news by tomorrow, maybe he can be un-subscribed.
On Wed, Feb 3, 2010 at 8:18 PM, Iñaki Baz Castillo i...@aliax.net wrote:
El Lunes, 1 de Febrero de 2010, Daniel-Constantin
The major disadvantage that I encounter with modparam (in the current
ratelimit module) is related to changes done via rpc/fifo/mi: on
server restart, all these changes are lost (unless the config is also
updated).
I much prefer to have the ability to change the db and then perform a
reload. This
Hello,
I know that we can break a call with sl_reply().
But, in the on_reply route, how can I break a call if I receive a 3XX (I
cant use a sl_reply() function!) ?
Thank you
___
Kamailio (OpenSER) - Users mailing list
e.g.:
failure_route[1] {
if(t_check_status(302)) {
xlog(L_INFO,302 redirect received\n);
t_reply(403,received 302 from downstream client);
exit;
}
}
klaus
Am 04.02.2010 08:37, schrieb BERGANZ Francois:
Hello,
I know that we can
69 matches
Mail list logo
