Kamailio SIP Proxy solution

List overview All Threads
Download

newer

older

4.4.5 Shared memory leak with...

Warnings at apply_changes()

Hai Bui Duc Ha

18 Apr 2017 18 Apr '17

3:25 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

Attachments:

attachment.html (text/html — 1.4 KB)
Kamailio-ProxyServer.jpg (image/jpeg — 38.8 KB)

Show replies by date

ycaner

18 Apr 18 Apr

5:56 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-tp157815... Sent from the Users mailing list archive at Nabble.com.

Hai Bui Duc Ha

7:32 a.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-request... 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@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.

Kamailio (SER) - Users Mailing List sr-users@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

gh＠heissa.de

25 Apr 25 Apr

6:29 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.

Alex Balashov

6:32 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@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@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/

gh＠heissa.de

7:16 a.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@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@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

gh＠heissa.de

11:57 a.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@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@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Daniel-Constantin Mierla

19 Apr 19 Apr

9:30 a.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

Hai Bui Duc Ha

12:37 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@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@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

2857

Age (days ago)

2864

Last active (days ago)

sr-users@lists.kamailio.org

8 comments

5 participants

tags (0)

participants (5)

Alex Balashov
Daniel-Constantin Mierla
gh＠heissa.de
Hai Bui Duc Ha
ycaner