18 Apr
2017
18 Apr
'17
8:25 a.m.
Hi all,
I'm researching the solution on Kamilio like SIP forward message:
- At the moment, the customer has server IP PBX and their clients.
- Then, the client just point the SIP Porxy configure to Kamailio. After
that, Kamailio will forward to IP PBX and send the event to another server.
- Everything will be not changed for customer site.
I found the link do the similar thing but it still authenticate on
Kamailio:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb…
Do you have any advice ?
Regrads,
Hai Bui
--
Hai Bui
VoIP engineer, Cvoice team, HTK-HCM Office
Mobile: +84-165-618-9876
18 Apr
18 Apr
10:56 a.m.
Hello,
i think you want to mirror IP PBX SIP port to Kamailio and all SIP
transactions send to another service with you created events.
is it for CRM project?
it is kinda homer. check it. maybe it helps you.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Kamailio-SIP-Proxy-solution-tp15781…
Sent from the Users mailing list archive at Nabble.com.
12:32 p.m.
Hi Ycaner,
Thanks for your advice !
I'm trying with this configure:
https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-reques…
This work okay, but need to do the event.
I will try with homer.
Regards,
Hai Bui
On Tue, Apr 18, 2017 at 12:56 PM, ycaner <yasin.caner(a)netgsm.com.tr> wrote:
Hello,
i think you want to mirror IP PBX SIP port to Kamailio and all SIP
transactions send to another service with you created events.
is it for CRM project?
it is kinda homer. check it. maybe it helps you.
--
View this message in context: http://sip-router.1086192.n5.
nabble.com/Kamailio-SIP-Proxy-solution-tp157815p157816.html
Sent from the Users mailing list archive at Nabble.com.
_______________________________________________
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Hai Bui
VoIP engineer, Cvoice team, HTK-HCM Office
Mobile: +84-165-618-9876
25 Apr
25 Apr
11:29 a.m.
New subject: Dispatcher Module ds_select_domain("2", "4")
Hi,
after Kamailio did a correct (INVITE sip:089321608@5.9.87.18 SIP/2.0 looks
good to me), to our Sip-Provider we get:
"SIP/2.0 407 Proxy Authentication Required."
Could it be that i have to rewrite also the following line:
To: <sip:089321608@sip.heissa.de>.
To: <sip:089321608@provider_from_dispatcher_table>.
and how can i do that?
Until now i use just ds_select_domain("2", "4")
Thanks,
Georg
U 130.255.76.95:5060 -> 5.9.87.18:5060
INVITE sip:089321608@5.9.87.18 SIP/2.0.
Record-Route: <sip:130.255.76.95;lr;ftag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco>.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
Via: SIP/2.0/UDP
100.99.24.192:53230;received=91.186.0.208;rport=39126;branch=z9hG4bKPjnlNeJmB2Id2F2Tmypfaf5sdh47ir3wg8.
Max-Forwards: 69.
From: "4321"
<sip:080424321@sip.heissa.de>;tag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco.
To: <sip:089321608@sip.heissa.de>.
Contact: <sip:080424321@91.186.0.208:39126;ob>.
Call-ID: -rgvfObl0VN39nEiZTf55yxxn8FemD52.
CSeq: 5545 INVITE.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_acer_z520-19/r2457.
Content-Type: application/sdp.
Content-Length: 366.
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
Via: SIP/2.0/UDP
100.99.24.192:53230;received=91.186.0.208;rport=39126;branch=z9hG4bKPjnlNeJmB2Id2F2Tmypfaf5sdh47ir3wg8.
Record-Route: <sip:130.255.76.95;lr;ftag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco>.
From: "4321"
<sip:080424321@sip.heissa.de>;tag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco.
To: <sip:089321608@sip.heissa.de>.
Call-ID: -rgvfObl0VN39nEiZTf55yxxn8FemD52.
CSeq: 5545 INVITE.
User-Agent: einfachVoIP.de.
Content-Length: 0.
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
11:32 a.m.
New subject: Dispatcher Module ds_select_domain("2", "4")
I'm not sure why you would immediately jump to the conclusion that you
need to rewrite the To header.
This is just a normal 407 challenge. The first and most straightforward
interpretation is that the provider expects you to provide digest
authentication credentials, or that they don't have your IP in their ACL
and are falling back to a digest challenge as a default behaviour for
that reason.
On Tue, Apr 25, 2017 at 08:29:47AM +0200, gh(a)heissa.de wrote:
Hi,
after Kamailio did a correct (INVITE sip:089321608@5.9.87.18 SIP/2.0 looks
good to me), to our Sip-Provider we get:
"SIP/2.0 407 Proxy Authentication Required."
Could it be that i have to rewrite also the following line:
To: <sip:089321608@sip.heissa.de>.
To: <sip:089321608@provider_from_dispatcher_table>.
and how can i do that?
Until now i use just ds_select_domain("2", "4")
Thanks,
Georg
U 130.255.76.95:5060 -> 5.9.87.18:5060
INVITE sip:089321608@5.9.87.18 SIP/2.0.
Record-Route: <sip:130.255.76.95;lr;ftag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco>.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
Via: SIP/2.0/UDP
100.99.24.192:53230;received=91.186.0.208;rport=39126;branch=z9hG4bKPjnlNeJmB2Id2F2Tmypfaf5sdh47ir3wg8.
Max-Forwards: 69.
From: "4321"
<sip:080424321@sip.heissa.de>;tag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco.
To: <sip:089321608@sip.heissa.de>.
Contact: <sip:080424321@91.186.0.208:39126;ob>.
Call-ID: -rgvfObl0VN39nEiZTf55yxxn8FemD52.
CSeq: 5545 INVITE.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_acer_z520-19/r2457.
Content-Type: application/sdp.
Content-Length: 366.
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
Via: SIP/2.0/UDP
100.99.24.192:53230;received=91.186.0.208;rport=39126;branch=z9hG4bKPjnlNeJmB2Id2F2Tmypfaf5sdh47ir3wg8.
Record-Route: <sip:130.255.76.95;lr;ftag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco>.
From: "4321"
<sip:080424321@sip.heissa.de>;tag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco.
To: <sip:089321608@sip.heissa.de>.
Call-ID: -rgvfObl0VN39nEiZTf55yxxn8FemD52.
CSeq: 5545 INVITE.
User-Agent: einfachVoIP.de.
Content-Length: 0.
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
12:16 p.m.
New subject: Dispatcher Module ds_select_domain("2", "4")
Hi Alex,
it's because i can see that kamailio (5.0.1 (x86_64/linux)) is already
successful connected to our Provider via kamailio uac module.
I can also see that kamailio send a refresh auth to 5.9.87.18 from time to
time.
Would the Provider answer with:
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 100 Trying.
in any case if the auth was not successful before?
Cheers,
Georg
I'm not sure why you would immediately jump to the
conclusion that you
need to rewrite the To header.
This is just a normal 407 challenge. The first and most straightforward
interpretation is that the provider expects you to provide digest
authentication credentials, or that they don't have your IP in their ACL
and are falling back to a digest challenge as a default behaviour for
that reason.
On Tue, Apr 25, 2017 at 08:29:47AM +0200, gh(a)heissa.de wrote:
Hi,
after Kamailio did a correct (INVITE sip:089321608@5.9.87.18 SIP/2.0
looks
good to me), to our Sip-Provider we get:
"SIP/2.0 407 Proxy Authentication Required."
Could it be that i have to rewrite also the following line:
To: <sip:089321608@sip.heissa.de>.
To: <sip:089321608@provider_from_dispatcher_table>.
and how can i do that?
Until now i use just ds_select_domain("2", "4")
Thanks,
Georg
U 130.255.76.95:5060 -> 5.9.87.18:5060
INVITE sip:089321608@5.9.87.18 SIP/2.0.
Record-Route:
<sip:130.255.76.95;lr;ftag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco>.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
Via: SIP/2.0/UDP
100.99.24.192:53230;received=91.186.0.208;rport=39126;branch=z9hG4bKPjnlNeJmB2Id2F2Tmypfaf5sdh47ir3wg8.
Max-Forwards: 69.
From: "4321"
<sip:080424321@sip.heissa.de>;tag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco.
To: <sip:089321608@sip.heissa.de>.
Contact: <sip:080424321@91.186.0.208:39126;ob>.
Call-ID: -rgvfObl0VN39nEiZTf55yxxn8FemD52.
CSeq: 5545 INVITE.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_acer_z520-19/r2457.
Content-Type: application/sdp.
Content-Length: 366.
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
Via: SIP/2.0/UDP
100.99.24.192:53230;received=91.186.0.208;rport=39126;branch=z9hG4bKPjnlNeJmB2Id2F2Tmypfaf5sdh47ir3wg8.
Record-Route:
<sip:130.255.76.95;lr;ftag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco>.
From: "4321"
<sip:080424321@sip.heissa.de>;tag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco.
To: <sip:089321608@sip.heissa.de>.
Call-ID: -rgvfObl0VN39nEiZTf55yxxn8FemD52.
CSeq: 5545 INVITE.
User-Agent: einfachVoIP.de.
Content-Length: 0.
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:57 p.m.
New subject: Dispatcher Module ds_select_domain("2", "4")
Hi all,
Solved!
If i create an identical user/password in kamailio which i get from command
kamcmd uac.reg_dump
{
l_uuid: 2
l_username: xyz
}
then it works.
Else result from Provider would be: SIP/2.0 403 Check SIP ID or username.
Is there any way to use avoid a refresh of providing digest authentication
credentials from INVITE?
The first auth from uac module should be enough!
I'm not sure why you would immediately jump to the
conclusion that you
need to rewrite the To header.
This is just a normal 407 challenge. The first and most straightforward
interpretation is that the provider expects you to provide digest
authentication credentials, or that they don't have your IP in their ACL
and are falling back to a digest challenge as a default behaviour for
that reason.
On Tue, Apr 25, 2017 at 08:29:47AM +0200, gh(a)heissa.de wrote:
Hi,
after Kamailio did a correct (INVITE sip:089321608@5.9.87.18 SIP/2.0
looks
good to me), to our Sip-Provider we get:
"SIP/2.0 407 Proxy Authentication Required."
Could it be that i have to rewrite also the following line:
To: <sip:089321608@sip.heissa.de>.
To: <sip:089321608@provider_from_dispatcher_table>.
and how can i do that?
Until now i use just ds_select_domain("2", "4")
Thanks,
Georg
U 130.255.76.95:5060 -> 5.9.87.18:5060
INVITE sip:089321608@5.9.87.18 SIP/2.0.
Record-Route:
<sip:130.255.76.95;lr;ftag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco>.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
Via: SIP/2.0/UDP
100.99.24.192:53230;received=91.186.0.208;rport=39126;branch=z9hG4bKPjnlNeJmB2Id2F2Tmypfaf5sdh47ir3wg8.
Max-Forwards: 69.
From: "4321"
<sip:080424321@sip.heissa.de>;tag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco.
To: <sip:089321608@sip.heissa.de>.
Contact: <sip:080424321@91.186.0.208:39126;ob>.
Call-ID: -rgvfObl0VN39nEiZTf55yxxn8FemD52.
CSeq: 5545 INVITE.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_acer_z520-19/r2457.
Content-Type: application/sdp.
Content-Length: 366.
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
Via: SIP/2.0/UDP
100.99.24.192:53230;received=91.186.0.208;rport=39126;branch=z9hG4bKPjnlNeJmB2Id2F2Tmypfaf5sdh47ir3wg8.
Record-Route:
<sip:130.255.76.95;lr;ftag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco>.
From: "4321"
<sip:080424321@sip.heissa.de>;tag=Wl3D6P2TkS5KuAGzo-9Z1Ezqy7NMRvco.
To: <sip:089321608@sip.heissa.de>.
Call-ID: -rgvfObl0VN39nEiZTf55yxxn8FemD52.
CSeq: 5545 INVITE.
User-Agent: einfachVoIP.de.
Content-Length: 0.
U 5.9.87.18:5060 -> 130.255.76.95:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
130.255.76.95;branch=z9hG4bK7828.51fb03ff0ac02b01b2ba2bb73c1de051.0.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
19 Apr
19 Apr
2:30 p.m.
Hello,
On 18.04.17 05:25, Hai Bui Duc Ha wrote:
Hi all,
I'm researching the solution on Kamilio like SIP forward message:
- At the moment, the customer has server IP PBX and their clients.
- Then, the client just point the SIP Porxy configure to Kamailio.
After that, Kamailio will forward to IP PBX and send the event to
another server.
- Everything will be not changed for customer site.
I found the link do the similar thing but it still authenticate on
Kamailio:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb…
Do you have any advice ?
is the PBX supporting SIP Path extension? If yes, then you have to
enable it in kamailio (see path module) and then forward the traffic to
the pbx.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
5:37 p.m.
Hi Daniel,
Can you explain me "is the PBX supporting SIP Path extension?".
I use the hardware PBX Yeastar S300 (
http://www.yeastar.com/s-series-voip-pbx/).
Actually, I have a problem with the configure ( https://blog.
voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/):
The IP PBX store the IP SIP client, but not store the IP of Kamailio proxy
server.
Regards,
Hai Bui
On Wed, Apr 19, 2017 at 4:30 PM, Daniel-Constantin Mierla <miconda(a)gmail.com
wrote:
Hello,
On 18.04.17 05:25, Hai Bui Duc Ha wrote:
Hi all,
I'm researching the solution on Kamilio like SIP forward message:
- At the moment, the customer has server IP PBX and their clients.
- Then, the client just point the SIP Porxy configure to Kamailio. After
that, Kamailio will forward to IP PBX and send the event to another server.
- Everything will be not changed for customer site.
I found the link do the similar thing but it still authenticate on
Kamailio: http://kb.asipto.com/asterisk:realtime:
kamailio-4.0.x-asterisk-11.3.0-astdb#kamailio_configuration
Do you have any advice ?
is the PBX supporting SIP Path extension? If yes, then you have to enable
it in kamailio (see path module) and then forward the traffic to the pbx.
Cheers,
Daniel
--
Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Hai Bui
VoIP engineer, Cvoice team, HTK-HCM Office
Mobile: +84-165-618-9876
2567
days inactive
2574
days old
8 comments
5 participants
participants (5)
-
Alex Balashov -
Daniel-Constantin Mierla -
gh＠heissa.de -
Hai Bui Duc Ha -
ycaner