Hi,
The documentation states the following.
https://www.kamailio.org/wiki/cookbooks/4.2.x/core#myself
if(uri==myself) {
log("the request is for local processing\n");
};
Now this could also be used to do something like this: (also used in
some examples)
if(from_uri==myself) {
log("the request is for local processing\n");
};
Using this the from_uri could be: sip:sip.voipexample.net:5061
myself in this case only has sip.voipexample.net:5060 as an Aias
Now the equation wouldn't match, because the port 5061 is non-default.
Would it be possible to match the from_uri to myself ignoring the port
in from_uri? The only thing I can think of is using a regular
expression. somthing like this:
from_uri=~"sip:.+@sip\.voipexample\.net(:([0-9]{1,4}|[1-5][0-9]{4}|6[0-4][0-9]{3}|65[0-4][0-9]{2}|655[0-2][0-9]|6553[0-5])|)"
But this skips the use of myself entirely. Any insight on this would be
greatly appreciated!
Cheers,
Dirk
Hello,
anyone going to Fosdem?
If no unexpected weather conditions messing the transport system, I am
going to be there. It would be good to know who else we can meet there.
If there are enough people interested, we can try organizing a social
event (dinner/drinks) on Saturday evening. Last year we did it together
with Jitsi team, but they are not coming this year. Perhaps there are
other folks around the RTC dev room we can ask.
On the other hand, it is harder to find a place if it is a large group,
even harder to put all together if there are different
groups/coordinators. Also, it is not easy to discuss withing a large
group, as typically for a dinner we were seated to tables. Therefore
organizing for a smaller group just for Kamailio friends could be a good
alternative. Let's see first the number of participant and the level of
interest for various options.
This being an announcement, was crossposted to few mailing lists, but as
most of the people are on sr-users, I think it would be good to reply
and follow up on this discussion only there.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
Hi guys,
I have tried to integrate kamailio and ldap.It seems like both are taking
to each other.
Here the purpose of ldap is for back-end support for sip user
authentication rather than kamailio database.However i am able to log in
but could not make call between sip users.I could not figure out what i am
missing.
Here is the routing block for ldap in my kamailio.cfg
> request_route {
> route(LDAPAUTH);
>
> }
---
route[LDAPAUTH]
> {
> if(is_method("REGISTER"))
> {
> if(is_present_hf("Authorization"))
> {
> if
> (!ldap_search("ldap://sipaccounts/ou=MyUsers,dc=example,dc=com?cn,userPassword?one?(cn=$fU)"))
> {
> switch ($retcode)
> {
> case -1:
> sl_send_reply("404", "User Not Found");
> exit;
> case -2:
> sl_send_reply("500", "Internal server error");
> exit;
> default:
> exit;
> }
> }
> ldap_result("cn/$avp(username)");
> ldap_result("userPassword/$avp(password)");
> route(REGISTRAR);
> if (!pv_www_authenticate("$td", "$avp(password)", "4"))
> {
> www_challenge("$td", "1");
> exit;
> }
> sl_send_reply("200", "ok");
> exit;
> }
> else
> {
> www_challenge("$td", "1");
> exit;
> }
> }
> }
Hello,
Is SIP trunking possible with the free version of Kamailio or we need to
upgrade
to business license?
Kindly get back to us with the details of using SIP trunking using
Kamailio. We have
a requirement in our organisation regarding the same which uses IMS and SIP
trunking.
In this setup we would initially like to test with 2 LTE EPCs connected to
two Kamailio IMS with a SIP trunk between them.
Thanks and Regards,
Indranil
Hi,
I'm using topoh and am running into lots of these errors:
Jan 19 20:34:52 csrpus1 /usr/local/sbin/kamailio[3641]: ERROR:
[R-DEFAULT-STATELESS-REPLY:nmhfrms2ib9Cnm9kS00Qtbtot5PMiQ0Ni1QKS5iKibuotel14Ygc]
!> Received stateless reply 200 (CANCEL) from xxx.xxx.xxx.xxx:5060
Jan 19 20:34:53 csrpus1 /usr/local/sbin/kamailio[3639]: ERROR: tm
[t_lookup.c:932]: t_reply_matching(): matching transaction found but
callids don't match (received:
nmhfrms2ib9Cnm9kS00Qtbtot5PMiQ0Ni1QKS5iKibuotel14Ygc stored:
51899BC6-569EE4330005CE1B-7D47C700_bleg)
The call topology is:
UAC --> Kamailio (+ topoh) --> PSTN GW
Topoh is behaving precisely as expected, but it looks like we might have
hit a TM case that topoh does not accommodate somehow:
1. A CANCEL comes in from the UAC and Kamailio sends a CANCEL for its
branch to the PSTN GW.
2. Kamailio sends a '200 cancelling' for the UAC's branch backward.
3. The PSTN GW replies with a 200 OK for the CANCEL (PSTN GW branch)
that contains the topoh-spoofed Call-ID, but for some reason Kamailio
does not match this reply, as indicated by the above error.
4. Because the 200 OK reply was stateless and not matched to a CANCEL
transaction, this causes Kamailio to retransmit the CANCEL to the PSTN
GW, at which point it receives: 481 Call Leg/Transaction Does Not Exist
5. #4 repeats once more.
I don't see any real harm from this situation, but it seems like
something that needs fixing. The 487 Request Terminated final reply is
processed and relayed correctly to the UAC, and the dialog ends
appropriately everywhere.
FWIW, I thought this might have to do with async processing of the
initial INVITE (through mqueue + rtimer), so I turned it off, but it
didn't make any difference.
My CANCEL handling in the main request route looks like this:
if(is_method("CANCEL")) {
set_rtpengine_set("1");
rtpengine_delete();
if(!t_relay_cancel()) {
sl_send_reply("500", "Internal Server Error");
exit;
}
xlog("L_ERR", "[R-MAIN:$ci] !> "
"Corresponding INVITE transaction for CANCEL "
"does not (or, no longer) exists\n");
exit;
}
So, it just looks like TM is not matching, and therefore not absorbing
the 200 OK reply to its CANCEL to the upstream gateway -- when topoh is
enabled.
Any insights would be appreciated!
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hello,
wondering if anyone is using fork=no -- some old docs suggest it is
suitable for debugging, but actually kamailio doesn't work properly in
this mode, leading to more troubles than benefits (e.g., having reports
of invalid issues, like tcp not working in this mode).
In first phase I would disable setting this value, with a warning if set
to no, because most of the configs out there have fork=yes. Removing it
could be considered in the future.
Note that this fork=no is different than don't daemonize controlled with
-D, which will stay being useful for some init.d systems.
Comments or other suggestions?
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
I am thinking that I cat get the port during the first call and then use
that port for the rest of calls.
Maybe that first call will fail but after that, all calls will go fine.
Example of a Trying showing received port: 52548
¿Do you know what pseudovariable represent that value?
I tested with all variables I could find but I could not find anyone.
Thank you
---------------------------------------------------------------------------------------------------------------------------------------------------------
2016/01/19 17:38:38.981987 VENDOR-IP:5060 -> KAMAILIO-IP:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
PUBLIC-IP:52548;branch=z9hG4bK8b5c.b122371ac0ac2f3ef3204b0f192cb16c.1;rport=52548
Via: SIP/2.0/UDP
ASTERISK-IP:5060;received=ASTERISK-IP;branch=z9hG4bK4082d124;rport=5060
From: 888888888 <sip:888888888@ASTERISK-IP>;tag=as34f971fe
To: <sip:999999999@VENDOR-DOMAIN>
Call-ID: 0b5307fa290674a97b47970643fce42a@ASTERISK-IP:5060
CSeq: 102 INVITE
Server: kamailio
Content-Length: 0
2016-01-15 9:50 GMT+01:00 Nelson Migliaro <eng.migliaro(a)gmail.com>:
> Hello Daniel,
>
> Yes, I am registered to the vendor.
>
> Regards,
>
> Nelson.-
>
> 2016-01-15 7:58 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>:
>
>> Ahh, I thought Asterisk is in the public internet, but actually you
>> connect to a provider (vendor), which seems to run Kamailio as well.
>>
>> Using information from 100 trying is too late, as the INVITE was already
>> sent... so one more question before trying to propose a solution. Do you
>> have to register to the provider?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 14/01/16 18:51, Nelson Migliaro wrote:
>>
>> Yes it is possible, but is there an easy way to workaround the issue
>> using Kamailio.
>>
>> Because I have the port because vendor is sending that info in Trying:
>>
>> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
>> SIP/2.0 100 trying -- your call is important to us
>> Via: SIP/2.0/UDP PUBLIC-IP:52548;branch=
>> z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=*52548*
>>
>> I need to find the way to discover the port used by firewall (maybe
>> getting that info from Trying) and then advertise that port.
>>
>>
>>
>> 2016-01-14 18:32 GMT+01:00 Daniel-Constantin Mierla < <miconda(a)gmail.com>
>> miconda(a)gmail.com>:
>>
>>> Not really up to date with all Asterisk features -- do you know if you
>>> can append a custom header to a SIP response that is going to be generated
>>> by Asterisk? Eventually the reply for an OPTIONS request.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 14/01/16 17:19, Nelson Migliaro wrote:
>>>
>>> Yes, I manage all devices, even the internet router but it does not
>>> allow static pat.
>>>
>>> 2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>:
>>>
>>>> Do you control the Asterisk? If yes, depending on Asterisk capabilities
>>>> of building replies, you may be able to do some automation to detect the
>>>> external port.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro <
>>>> <eng.migliaro(a)gmail.com>eng.migliaro(a)gmail.com> wrote:
>>>>
>>>>> There is not a public Kamailio, only one Kamailio behind NAT,
>>>>>
>>>>> Right now the configuration is:
>>>>>
>>>>> Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy )
>>>>> <-> Internet router (public IP + symmetric na) <-> Internet
>>>>>
>>>>> Regards,
>>>>>
>>>>> 2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla <
>>>>> <miconda(a)gmail.com>miconda(a)gmail.com>:
>>>>>
>>>>>> Is the kamailio behind nat communicating with another kamailio on a
>>>>>> public IP?
>>>>>>
>>>>>> Cheers,
>>>>>> DAniel
>>>>>>
>>>>>> On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro <
>>>>>> <eng.migliaro(a)gmail.com>eng.migliaro(a)gmail.com> wrote:
>>>>>>
>>>>>>> Thank you Daniel for your answer,
>>>>>>>
>>>>>>> As you mention, there is a symmetric nat and router does not allow a
>>>>>>> static NAT.
>>>>>>>
>>>>>>> By sniffing traffic I can see the port is using new but in case it
>>>>>>> change, how can automate the process of advertising the correct port?
>>>>>>>
>>>>>>> Cheers!
>>>>>>>
>>>>>>>
>>>>>>> ---------- Forwarded message ----------
>>>>>>> From: Daniel-Constantin Mierla < <miconda(a)gmail.com>
>>>>>>> miconda(a)gmail.com>
>>>>>>> Date: 2016-01-13 23:28 GMT+01:00
>>>>>>> Subject: Re: [SR-Users] Kamailio and NAT
>>>>>>> To: "Kamailio (SER) - Users Mailing List" <
>>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org>
>>>>>>>
>>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> it looks like you have a symmetric nat router, so the allocated port
>>>>>>> is randomly selected.
>>>>>>>
>>>>>>> If you don't control the nat router to set a static forwarding rule
>>>>>>> or it doesn't provide the option to set static forwarding, then you are
>>>>>>> pretty much left with sniffing the traffic to discover the external port
>>>>>>> and advertise it.
>>>>>>>
>>>>>>> Cheers,
>>>>>>> Daniel
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On 13/01/16 20:31, Nelson Migliaro wrote:
>>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> I finally were able to run my Kamailio behind NAT but in order to
>>>>>>> accomplish that I included:
>>>>>>>
>>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
>>>>>>>
>>>>>>> 52548 is the port my internet router change when doing NAT
>>>>>>> (5060->52548). I found this port sniffing traffic
>>>>>>>
>>>>>>> Conclusions at this point are:
>>>>>>>
>>>>>>>
>>>>>>> ---------------------------------------------1--------------------------------------------------------------------------------------------------
>>>>>>> If I use this line:
>>>>>>>
>>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work
>>>>>>> :(
>>>>>>>
>>>>>>> When I dial a call, INVITE / ACK / Trying / OK goes fine because
>>>>>>> they are part of the same transaction
>>>>>>> When remote party disconnects the call, BYE goes to PUBLIC-IP port
>>>>>>> 5060 and router blocks de request. I assume vendor sends BYE to 5060
>>>>>>> because it is a new transaction
>>>>>>>
>>>>>>> -----------------------------------------------2--------------------------------------------------------------------------------------------------
>>>>>>>
>>>>>>> If I use this line:
>>>>>>>
>>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!!
>>>>>>>
>>>>>>> When I dial a call, INVITE / ACK / Trying / OK goes fine because
>>>>>>> they are part of the same transaction
>>>>>>> When remote party disconnects the call, BYE goes to PUBLIC-IP port
>>>>>>> 52548 and router forward the request to Kamailio. Since there is an open
>>>>>>> connection.
>>>>>>>
>>>>>>> I need to find the way to find the way to advertise the public port
>>>>>>> internet router is doing NAT (PAT).
>>>>>>>
>>>>>>>
>>>>>>> ---------------------------------------------------------------------------------------------------------------------------------------------------
>>>>>>> This trace is a call that worked fine because I included line:
>>>>>>>
>>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548
>>>>>>>
>>>>>>>
>>>>>>> This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060
>>>>>>> advertise PUBLIC-IP:52548
>>>>>>> 2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060
>>>>>>> INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0
>>>>>>> Record-Route: <
>>>>>>> sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB
>>>>>>> A-;nat=yes>
>>>>>>> Via: SIP/2.0/UDP
>>>>>>> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0
>>>>>>> Via: SIP/2.0/UDP
>>>>>>> PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
>>>>>>> Max-Forwards: 69
>>>>>>> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
>>>>>>> To: <sip:NUM-DESTINATION@sip.VENDOR-IP>
>>>>>>> Contact:
>>>>>>> <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1>
>>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
>>>>>>> CSeq: 102 INVITE
>>>>>>> User-Agent: Kamailio
>>>>>>> Date: Wed, 13 Jan 2016 19:10:15 GMT
>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>> INFO, PUBLISH, MESSAGE
>>>>>>> Supported: replaces, timer
>>>>>>> Content-Type: application/sdp
>>>>>>> Content-Length: 255
>>>>>>>
>>>>>>>
>>>>>>> Trying.....
>>>>>>>
>>>>>>> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
>>>>>>> SIP/2.0 100 trying -- your call is important to us
>>>>>>> Via: SIP/2.0/UDP
>>>>>>> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548
>>>>>>> Via: SIP/2.0/UDP
>>>>>>> PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060
>>>>>>> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453
>>>>>>> To: <sip:NUM-DESTINATION@VENDOR-IP>
>>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
>>>>>>> CSeq: 102 INVITE
>>>>>>> Server: kamailio
>>>>>>> Content-Length: 0
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> And finally a BYE
>>>>>>>
>>>>>>> 2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
>>>>>>> BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1
>>>>>>> SIP/2.0
>>>>>>> Via: SIP/2.0/UDP
>>>>>>> VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0
>>>>>>> From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93
>>>>>>> To: "NUM-SOURCE" <sip:NUM-SOURCE@VENDOR-IP>;tag=as3b72a453
>>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060
>>>>>>> CSeq: 28731 BYE
>>>>>>> Max-Forwards: 69
>>>>>>> Route: <
>>>>>>> sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na
>>>>>>> yes>
>>>>>>> Reason: Q.850;cause=16
>>>>>>> Content-Length: 0
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> -----------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>>>>>>
>>>>>>> -----------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>>>>>>
>>>>>>> Finally, It is finally working because I hardcoded NAT´d port.
>>>>>>> I would like to find a way to avoid setting the port in "hard".
>>>>>>>
>>>>>>> Thank you
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>>>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>>>>> list
>>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org
>>>>>>> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>>>>> list
>>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org
>>>>>>> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Daniel-Constantin Mierla - <http://www.asipto.com>
>>>>>> http://www.asipto.com
>>>>>> <http://twitter.com/#%21/miconda>http://twitter.com/#!/miconda -
>>>>>> <http://www.linkedin.com/in/miconda>http://www.linkedin.com/in/micond
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Daniel-Constantin Mierla - <http://www.asipto.com>http://www.asipto.com
>>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond
>>>> <http://www.linkedin.com/in/miconda>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
>>
>>
>
Hi,
I am a newbie. As I read I understand I need to use UAC module for
registration but cannot figure how to trigger the registration request. I
enable the UAC module and it seem to be working.
How can I trigger the REGISTER to a SIP provider account?
And is it possible to use multiple accounts for different provider
registrations?
Can someone help about UAC usage please.
Regards
Evren Akbulut
Hi,
We have a media server and (opensips + rtpengine).
How can I configure opensips+rtpengine to just do DTLS handshake with the
client, but the media should not go through RTPEngine. The media should go
to the Media server.
Thanks.
Riko
