Hi
A client of ours needs to insert information into a table at some point into the call. I am in betwwen these 2 options:
1- Using http_query function from the UTILS module to call a web service that will insert the information
passed in the URL in to the table
2- or using the sql_query function from the SQLOPS module.
My priority is to use the fastest one. Since both are TCP they kind of come with some reliability regarding re-tries, etc (vs using a method based on UDP)
I am king of leaning versus the option #1, b/c I am worried about the cost of creating a db connection everytime etc but we dont have experience on none of the methods.
Also please take into account that they are still using version 1.52, and upgrading is not an option at the moment.
Thank you very much, your answer is much appreciated!
john
Hello,
Just wanted to shoot a quick email to the user list and see if anyone
had any 'gotchas' to keep in mind when migrating from 3.2 to 3.3.5.
I know table structure has changed and will need to be updated, but are
there otherwise any syntax and or major configuration differences?
Thank you for your time in advance.
Sincerely,
Brandon Armstead
Hello,
Kamailio SIP Server v4.0.3 stable release is out.
This is a maintenance release of the latest stable branch, 4.0, that
includes fixes since release of v4.0.0. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.0.0. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.0.3.
For more details about version 4.0.3 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2013/08/kamailio-v4-0-3-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
I have SIP proxy (Kamailio) works in conjunction with
rtpproxy<http://www.rtpproxy.org/> to
support client communication. When SIP proxy sends command to rtpproxy to
create new session, rtpproxy will create 2 ports (let's called them port1
and port2). rtpproxy has 1 listen interface
Supposed A and B are 2 clients that use rtpproxy to relay RTP stream, and
works fine.
A <---> port1 [*rtpproxy*] port2 <---> B
Now that A loses his current network, and enter network2 (imagine a network
handover) to become A2. In this case, I see rtpproxy still works fine by
relaying stream between A2 and B
A2 <---> port1 [*rtpproxy*] port2 <---> B
But when A2 lose his network2 and enters network3 to become A3, rtpproxy
stills relay stream between A2 and B. It seems that A can change his
network only once.
A2 <---> port1 [*rtpproxy*] port2 <---> B
A3
Why did the first handover succeed? How can I change rtpproxy behavior to
support many handovers ?
--
Khoa Pham
HCMC University of Science
www.fantageek.com
I am having an issue with the is_subscriber function provided by auth_db module. What I want to accomplish is: if the call is not to a local subscriber then route via LCR. If the call is to a local subscriber then route via LOCATION.
My subscriber table is called sip_accounts and I am using multidomain so domain is populated in subscriber definitions.
Here is my routing block-
if (!is_subscriber("$ru", "sip_accounts", "1")){
route(LCR);
exit;
} else {
route(LOCATION);
exit;
}
Call always routes via LCR despite calling a local subscriber.
INVITE is correct:
INVITE sip:user2@domain.com SIP/2.0.
From: user1 <sip:user1@domain.com>;tag=520c9f9ade1e7026o0.
To: <sip:user2@domain.com>.
Any ideas or direction where my issue could be?
-dan
Alex,
Thanks. It's good to know that it should work.
Daniel,
It's a 2811 running IOS 12.4(25g).
>> [a] ===> [kamailio] ===> [cisco] ===> [b]
This is exactly what I need. I'm trying to modify the request at [kamailio] so that [cisco] forwards to [b].
Possible complication is that [cisco] is also configured as a voice gateway with a SIP trunk to an external provider and a Call manager on the inside. However, this is independent SIP traffic that [cisco] appears to try to route according to voice rules.
I can get the call to proceed by adding a dial-peer to [cisco] config, but if I have understood loose-routing, that shouldn't be necessary.
I have tried several combinations of header and r-URI modifications to test [cisco]'s behaviour, including Route:[cisco];lr but it still seems to respond to the r-uri in the way I initially described.
All the best,
Dave
On 14 Aug 2013, at 5:09 pm, sr-users-request(a)lists.sip-router.org wrote:
>
>
> Message: 3
> Date: Wed, 14 Aug 2013 13:06:19 +0400
> From: Alex Balashov <abalashov(a)evaristesys.com>
> To: miconda(a)gmail.com, "Kamailio (SER) - Users Mailing List"
> <sr-users(a)lists.sip-router.org>
> Subject: Re: [SR-Users] Loose Routing with Cisco router
> Message-ID: <65a98f42-c370-49fe-91c4-d018b3d61249(a)email.android.com>
> Content-Type: text/plain; charset=UTF-8
>
> I use Kamailio extensively with Cisco AS5xxx series gateways and they have never had a problem dealing with Kamailio's RRs and lr values.
>
>
> Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:
>> Hello,
>>
>> what model the cisco router is?
>>
>> From what I understand, you want to go from:
>>
>> [a] ===> [kamailio] ===> [b]
>>
>> to:
>>
>> [a] ===> [kamailio] ===> [cisco] ===> [b]
>>
>> The usual will be that kamailio sets the r-uri to [b] and dsturi
>> (outbound proxy) to [cisco] and then relay. The cisco should have some
>> config options just to forward the traffic based on r-uri, however I
>> never had to deal cisco configs. Maybe you add a Route header with
>> [cisco];lr, then it should match itself in the route header and send to
>>
>> r-uri.
>>
>> On the other hand, when I had to interoperate with cisco
>> gateways/b2bua,
>> they had no problem understanding loose routing added by kamailio.
>>
>> Cheers,
>> Daniel
>>
>> On 8/13/13 10:05 AM, David Wilson wrote:
>>> Hello All,
>>>
>>> I'm running Kamailio 4.0.2 on Ubuntu 12.04 (precise).
>>>
>>> I have SIP messaging flowing nicely, with UACs registering via
>> Kamailio (as a proxy) to a Registrar.
>>>
>>> Now having problems trying to route messages via a Cisco router
>> (12.4(25g)) using loose routing. The plan is to allow the router to
>> see the SIP messaging but forward it to the original location.
>>>
>>> RFC 3261 seems to cover this case in Section 16.6 part 6: "A proxy
>> MAY have a local policy that mandates that a request visit a specific
>> set of proxies before being delivered to the destination."
>>>
>>> Cisco documentation (SIP Configuration Guide, Cisco IOS Release 12.4)
>> states that it implements RFC 3261, including Loose-routing. However,
>> from my observation the router doesn't like a request-URI with anything
>> other than its own IP address (it returns 400 Bad Request - 'Invalid IP
>> address'), but with any attempt to 'decorate' the URI with lr and/or
>> maddr= parameters it returns 400 Bad Request - 'Malformed/Missing' URL.
>> One of these messages is returned regardless of anything I've tried
>> with Via, Record-Route or Route headers.
>>>
>>> Desired behaviour is for the router to remain in the route-set. A
>> different configuration has shown me that this will achieve the
>> required outcome, the tricky part now is just getting that first
>> request to route. I know several ways to make Kamailio send to the
>> router, but the router is not behaving as I expect when I preload a
>> route set (with Route: headers).
>>>
>>> Recognising that this is probably more of a Cisco problem than
>> Kamailio, can anyone confirm whether a Cisco router implements
>> loose-routing when receiving messages, or only when sending?
>>>
>>> Regards,
>>> Dave.
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>> list
>>> sr-users(a)lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Sent from my mobile, and thus lacking in the refinement one might expect from a fully fledged keyboard.
>
> 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/, http://www.alexbalashov.com
>
>
>
All,
I have just setup kamailio as SPI outbound proxy, in front of Asterisk.
I'm novice with Kamailio, it's the first time I use it.
The setup is working but I need your advises:
1) When I type the "sip show peers" command in ASterisk, I see the ip
address of the sip proxy. The qualify (monitoring/keepalive) seems to be
sent to the sip proxy and not to the phone. Is there an alternative to
directly monitor the phone ?
localhost*CLI> sip show peers
phone2a/phone2a * <IP_SIP_PROXY> * D N
53 OK (299 ms)
2) If I use the command "localhost*CLI> sip show peer phone2a"
The ip of the phone is visible in the field "Reg. Contact" only.
In the field " Addr->IP", it's the IP of the SIP Proxy.
* Name : phone2a
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : client2
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 554
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : *<SIP PROXY IP>*
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: phone2a
SIP Options : (none)
Codecs : 0xa (gsm|alaw)
Codec Order : (gsm:20,alaw:20)
Auto-Framing : No
Status : OK (299 ms)
Useragent : LinphoneAndroid/2.1.2 (eXosip2/3.6.0)
Reg. Contact : sip:phone2a@*<IP PHONE>*
line=8b1b24fbaaf794a
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
3) I'm running a fail2ban protection to protect against scanners and my
fail2ban is blocking the SIP Proxy when the threshold is reached, which
means that all the clients behind the sip outbound proxy are blocked.
I think the points 1, 2, 3 are related and if the SIP Proxy could be
"transparent"
Here is a debug of a register request, taken on the kamailio server
REGISTER sip:pbx-qa.mydomain.com SIP/2.0
Via: SIP/2.0/UDP <IP SIP PROXY>;branch=z9hG4bKfea4.34a8fd83.0
Via: SIP/2.0/UDP 100.96.196.103:4294;received=<IP
PHONE>;rport=6738;branch=z9hG4bK1329841729
From: <sip:phone2a@mydomain.com>;tag=1870152222
To: <sip:phone2a@pbx-qa.domain.com>
Call-ID: 1455540209
CSeq: 2 REGISTER
Contact: <sip:phone2a@100.96.196.103:4294;line=c6f956d7cdb0eb5>
Authorization: Digest username="phone2a", realm="asterisk",
nonce="176e8fa1", uri="sip:pbx-qa.domain.com",
response="4c68e98p1ea7cb0ee81674a8384ca6e4", algorithm=MD5
Max-Forwards: 69
User-Agent: LinphoneAndroid/2.1.2 (eXosip2/3.6.0)
Expires: 600
Content-Length: 0
P-hint: outbound
Any ideas to solve my problem, to get a "more" transparent proxy ?
Regards,
Renaud Dubois
I'm installing kamailio with a public IP on a server but for some reason in
can't logging with my sip client. Any help please
DISTRIB_ID=Ubuntu
DISTRIB_RELEASE=10.04
DISTRIB_CODENAME=lucid
DISTRIB_DESCRIPTION="Ubuntu 10.04.2 LTS"
--
Kethzer Docteur
