Re: [SR-Users] how can I use WITH_IPAUTH when the IP may be stored in the DB as a FQDN?
Authentication by IP address must be done by IP address only, DNS names must
not be allowed. Period. By definition.
What you want can be achieved with dns_int_match_ip() function provided by
ipops module. But keep in mind it is slow because of DNS lookup.
On Sunday 08 March 2015 13:38:52 canuck15 wrote:
Here is is the relevant section of kamailio.cfg
$var(tempfU) = $fU;
#!ifdef WITH_IPAUTH
if((!is_method(REGISTER)) allow_source_address() $au == )
{
# Loading $fU from database using IP
sql_pvquery(elxpbx, SELECT name FROM sip WHERE host = '$si'
AND sippasswd IS NULL, $var(tempfU));
# source IP allowed
return;
}
The problem is that when host= somefqdn.com the above will fail since
$si will always be an IP address as far as I can tell. More often than
not host= is a fqdn and requiring it to always be an IP address is not
an option. Converting it to IP before storing it in the DB is also not
an option because it needs to be able to work of the IP address changes.
So how can the above be done to accomodate the possibility that host=
somefqdn.com or an IP address. Preferably in such a way that it can
scale to hundreds/thousands of rows in the database without slowing
things down or crashing.
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[SR-Users] how can I use WITH_IPAUTH when the IP may be stored in the DB as a FQDN?
Here is is the relevant section of kamailio.cfg
$var(tempfU) = $fU;
#!ifdef WITH_IPAUTH
if((!is_method(REGISTER)) allow_source_address() $au == )
{
# Loading $fU from database using IP
sql_pvquery(elxpbx, SELECT name FROM sip WHERE host = '$si'
AND sippasswd IS NULL, $var(tempfU));
# source IP allowed
return;
}
The problem is that when host= somefqdn.com the above will fail since
$si will always be an IP address as far as I can tell. More often than
not host= is a fqdn and requiring it to always be an IP address is not
an option. Converting it to IP before storing it in the DB is also not
an option because it needs to be able to work of the IP address changes.
So how can the above be done to accomodate the possibility that host=
somefqdn.com or an IP address. Preferably in such a way that it can
scale to hundreds/thousands of rows in the database without slowing
things down or crashing.
___
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sr-users@lists.sip-router.org
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Re: [SR-Users] Dialog order of operations
Well, I can say this: it seems that dlg_set_property(ka-{src,dst})
must be set after dlg_manage(), or it has no effect.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/
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Re: [SR-Users] kamailio asterisk
Hello Everyone,
I set up tcp and udp for right now to test all config and dialog is not
established correctly. Call between extensions not working. Call from
extesnsion to extension going right a way to voicemail. Any help thank you.
Here one call debug.
http://fpaste.org/195572/14258553/
Slava.
From: Slava Bendersky volga...@networklab.ca
To: mico...@gmail.com
Cc: sr-users sr-users@lists.sip-router.org
Sent: Tuesday, March 3, 2015 9:08:19 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello Daniel,
Is notify should follow subscribe routes ?
Here SUBSCRIBE completely look normal to me.
--- SIP read from UDP:10.18.130.46:5060 ---
SUBSCRIBE sip:10...@networklab.ca SIP/2.0
Record-Route:
sip:kamailio_pub_ip:5081;transport=tls;lr=on;ftag=08c99307d3;nat=yes
Accept: application/simple-message-summary
Via: SIP/2.0/UDP
10.18.130.46;branch=z9hG4bKdee2.b83c2028dc57c641d2b8cad968347d6a.0;i=2
Via: SIP/2.0/TLS
192.168.0.16:5063;rport=5063;received=client_pub_ip;branch=z9hG4bK0c54032bd8905d262
Max-Forwards: 69
From: 10102 sips:10...@networklab.ca;tag=08c99307d3
To: sips:10...@networklab.ca
Call-ID: cea703eb8c21f709
CSeq: 1096440084 SUBSCRIBE
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE,
UPDATE
Allow-Events: message-summary
Authorization: Digest
username=10102,realm=networklab.ca,nonce=29ac01f1,uri=sips:10...@networklab.ca,response=0298c2a93bdd4969a3cd0b757d671dbb,algorithm=MD5
Contact: sips:10102@client_pub_ip:5063;audio
Event: message-summary
Expires: 3600
Supported: eventlist, replaces, timer
User-Agent: Media5-fone/4.1.6.3283 Android/4.1.2
Content-Length: 0
Path:
sip:outbound@10.18.130.46;lr;received=sip:client_pub_ip:5063%3Btransport%3Dtls
P-hint: outbound
Slava
From: Daniel-Constantin Mierla mico...@gmail.com
To: Slava Bendersky volga...@networklab.ca
Cc: sr-users sr-users@lists.sip-router.org
Sent: Tuesday, March 3, 2015 1:50:03 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello,
I pushed a fix for it -- the issue was in a LM_DBG() trying to print te address
for local socket, but it was not set yet, it was affecting only when running in
debug mode with debug=3. I pushed the patch to branch 4.2, so if you want the
fix, then you have to install from git branch 4.2.
For REGISTER, you need to add the Path header (see path module), otherwise the
OPTIONS and INVITE sent out to the phone are not going via Kamailio. You have
to check the version of your Asterisk to be one that supports Path.
Cheers,
Daniel
On 27/02/15 14:40, Slava Bendersky wrote:
Hello Daniel,
Here paste for gdb
http://fpaste.org/191338/25043949/
I got REGISTER and SUBSCRIBE start working correctly I see on asterisk correct
record routes and sip traffic flow, but when asterisk or client ( soft phone)
send OPTIONS or NOTIFY can't get properly relay it.
This is SUBSCRIBE route.
--- Transmitting (NAT) to 10.18.130.46:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060
Via: SIP/2.0/TLS 192.168.88.252:5062;received=Client public
ip;branch=z9hG4bK0bbe1f7d27257bba9;rport=5062
Record-Route: sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes
Record-Route:
sip:PUBLIC_KAMAILIO_IP:5081;transport=tls;r2=on;lr=on;ftag=a185d974ec;nat=yes
From: Slava Bendersky sips:10...@networklab.ca ;tag=a185d974ec
To: sips:10...@networklab.ca ;tag=as00757d3e
Call-ID: b08adb1ad1804a83
CSeq: 236711034 SUBSCRIBE
Server: FPBX-2.11.0(11.15.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: sip:10101@10.18.130.50:5060 ;expires=3600
Content-Length: 0
Slava.
From: Daniel-Constantin Mierla mico...@gmail.com
To: Slava Bendersky volga...@networklab.ca
Cc: sr-users sr-users@lists.sip-router.org
Sent: Friday, February 27, 2015 6:42:33 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello,
I asked for the wrong command, as the bt full was already sent before -- I
wanted the output from gdb for:
p *tcpconn
Daniel
On 27/02/15 04:10, Slava Bendersky wrote:
BQ_BEGIN
Hello Daniel,
Here bt full from back trace.
http://fpaste.org/191207/50064491/
Slava.
From: volga629 volga...@skillsearch.ca
To: mico...@gmail.com , Slava Bendersky volga...@networklab.ca
Cc: sr-users sr-users@lists.sip-router.org
Sent: Thursday, February 26, 2015 9:56:48 PM
Subject: Re: [SR-Users] kamailio asterisk
Hello Daniel,
I tried $rz option on top of request route and that where I see wrong request
uri like sip:sips : . And as far I can see it happenes only for SUBSCRIBE
INVITE and NOTIFY.
if($rz==sips) {
$ru = sip + $(ru{s.substr,4,0});
}
Slava.
Sent from mobile device typos are expected.
From: Daniel-Constantin Mierla mico...@gmail.com
Sent: Feb 25, 2015 1:04 PM
To: Slava Bendersky
Cc: sr-users
Subject: Re: [SR-Users] kamailio asterisk
Hello,
On
Re: [SR-Users] RTPProxy issue?
Hello Maxim, I'm running legacy 1.2 or 1.4, not sure. I see in the latest code that the function is still there. Do you suggest to upgrade or there's a patch to make? Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Maxim Sobolev Envoyé : samedi 7 mars 2015 09:14 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] RTPProxy issue? Ah, ok, I see now. I did not realize you guys are using resizer. Which version of the software are you actually using? I.e. is it latest rel_2_0 / master, or some legacy 1.x code? We've done quite some revamping down there, so that it needs to be checked against the very latest code to make sure. Let us know. Thanks! On Mar 6, 2015 12:31 AM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: Hi Maxim, Hard to do because it's in production. I have a serious finding since yesterday on how this happened. My understanding is that the function ts_less returns FALSE into rtp_resizer.c because the timestamp between the two packets is (1 31) [for example: 3740425320]. That's result in a drop of any following packets as I can see it into a capture. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org mailto:sr-users-boun...@lists.sip-router.org ] De la part de Maxim Sobolev Envoyé : vendredi 6 mars 2015 07:44 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] RTPProxy issue? Hi Igor, that's bit strange, since the rtpproxy is not checking any of the rtp flags including marker bit. It would help if you can post a tcpdump capture of the streams in question along with the log output of the rtpproxy running at the dbug level. Thanks! On Mar 5, 2015 5:54 AM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: I reviewed again a call trace and I can be more precise: a RTP packet comes with a new SSRC and the Marker bit set to True. This packet is properly forwarded. Then, just after this packet, another RTP packet containing a new SSRC with the huge timestamp and the Marker bit set to True is coming from the UA. The RTPProxy stops forward since this packet. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com ] Envoyé : jeudi 5 mars 2015 11:34 À : mico...@gmail.com mailto:mico...@gmail.com ; 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] RTPProxy issue? Hello, Thank you. Just to let you know, the RTPProxy is running in bridging mode. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Daniel-Constantin Mierla Envoyé : jeudi 5 mars 2015 09:33 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] RTPProxy issue? Hello, maybe Maxim (cc-ed) will be able to provide more insights. Cheers, DAniel On 04/03/15 16:59, Igor Potjevlesch wrote: Hello, I discovered an issue related to the handling of timestamp and/or Marker bit with rtpproxy (I use the latest Extension 20081224). The call-flow is the following: one UA places a call to A and put this call on hold. Then, the same UA call another number B. Individual streams are ok. When the UA tries to transfer A with B, the RTPProxy receive a RTP packet with a huge timestamp and the Marker bit set to True. Just after this RTP packet, RTPProxy stop forward the RTP packets from A to B. B to C is still working. Anyone have an idea? Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] 4.2.3 textops: flag f doesn't work in subst_hf(hf, subexp, flags)
Hello, I received INVITE with 3 Diversion headers. I try to make substitutions in the body of a first Diversion header field by using flag f, command *subst_hf(Diversion, /sip:(.*)@/sip:$var(dU)@/, f) ;* substitutes *all headers *Diversion instead of only first header, command *subst_hf(Diversion, /sip:(.*)@/sip:$var(dU)@/, *l*) ;* substitutes only *last* Diversion header. Thanks for help, Julia. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
