Hi again,

I used $sel(ruri.hostport) for Kamailio1 after t_relay in Relay route. So I managed to get the request-line URI's IP and port parameters of the outgoing INVITE that is leaving the first Kamailio box.

Thank you very much.

-----Original Message----- From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of CUMHUR KARAHAN Sent: 17 Ocak 2020 Cuma 15:23 To: sr-users@lists.kamailio.org Subject: Re: [SR-Users] Get destination IP from forwarded invite

Hi Joel,

Thank you for your response.

In fact, I want to get Request-line URI's parameters of the outgoing INVITE that is leaving the first Kamailio box, briefly. Unfortunately "du - destination URL" or "sndto(name)" pseudo-variables indicate the next hope's information. And in this case, the next hope's information is not same as the ClientB's information.

As I tried to tell before, our setup is like that:

ClientA -> Kamailio1 -> Kamailio2 -> ClientB

I need the request line URI parameters of the invite message sent by Kamailio1 to Kamailio2. And these parameters contain ClientB's ip and port information such as 10.252.180.51:36945 as you can see below:

INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes Record-Route: sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: sip:905368601286@testsip.tvoip.turkcell.com.tr;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE Call-ID: 1wF-vjzGIX ...

And, your question: no, we don't use the dispatcher mechanism. We just check the DB to find out Kamailio server which the number registered into and send the messages to this server basically.

Thanks for all...

-----Original Message----- From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of sr-users-request@lists.kamailio.org Sent: 17 Ocak 2020 Cuma 14:00 To: sr-users@lists.kamailio.org Subject: sr-users Digest, Vol 176, Issue 17

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To subscribe or unsubscribe via the World Wide Web, visit https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request@lists.kamailio.org

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Today's Topics:

1. Get destination IP from forwarded invite (CUMHUR KARAHAN) 2. Re: Kamalio + RTP Engine + SIP Client behind NAT (Richard Fuchs) 3. Re: Get destination IP from forwarded invite (Joel Serrano) 4. Re: Kamalio + RTP Engine + SIP Client behind NAT (Nuno Miguel Reis) 5. Re: Kamailio/RTC dinner at Fosdem 2020 (Federico Santulli) 6. Re: http_client interface (Daniel-Constantin Mierla) 7. Re: Kamailio stop to process incoming SIP traffic via TCP. (adeykunov) 8. Question about xhttp module kamailio 5.2.x (Jimmy Rodley) 9. Trouble with 302 Redirect (Jim Leahy) 10. Re: Kamalio + RTP Engine + SIP Client behind NAT (Richard Fuchs) 11. Re: http_client interface (Fred Posner) 12. Re: Trouble with 302 Redirect (Alex Balashov) 13. UCaaS Platform (Dario Vargas) 14. Re: [commercial announcement] (Henning Westerholt)

----------------------------------------------------------------------

Message: 1 Date: Thu, 16 Jan 2020 11:45:15 +0000 From: CUMHUR KARAHAN cumhur.karahan@consultant.turkcell.com.tr To: "sr-users@lists.kamailio.org" sr-users@lists.kamailio.org Subject: [SR-Users] Get destination IP from forwarded invite Message-ID: 319901a90d17413292ad249eb56df2ec@consultant.turkcell.com.tr Content-Type: text/plain; charset="utf-8"

Hi all,

I'm a newbie for Kamailio and working on a feature needs end-users' destination IP in the first transferred/forwarded invite message.

Basically, our system/call flow is like: ClientA -> Kamailio1 -> Kamailio2 -> ClientB

Invite from the clientA:

tls:212.252.34.165:32980 tls:172.21.193.100:5061 INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0 Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport From: sip:905368601286@testsip.tvoip.turkcell.com.tr;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 20 INVITE ...

Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message.

Invite to the Kamailio2 :

udp:172.21.193.100:5060 udp:172.21.193.98:5060 INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes Record-Route: sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: sip:905368601286@testsip.tvoip.turkcell.com.tr;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE ...

When the first invite message receives, the dialog variables or sel(msg...) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message's variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts.

Is there any way to get it without diving C code's abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue.

Thanks for all...

S. Cumhur Karahan

[http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2f%2fturkcell.li%2fncfYh&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-bbcefdc4443d64fef548126bd3c5f932d722a0a4

Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir.

TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir

Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir.

________________________________

This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted.

TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL.

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Message: 2 Date: Thu, 16 Jan 2020 08:23:21 -0500 From: Richard Fuchs rfuchs@sipwise.com To: sr-users@lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: 4686712f-d865-c6c9-11ac-649966b867b1@sipwise.com Content-Type: text/plain; charset="utf-8"; Format="flowed"

On 15/01/2020 13.39, Nuno Miguel Reis wrote:

Hi guys.

I'm replacing a environment where I was using kamailio + freeswitch by another where I'm adding rtpengine to the mix. One of the issues I'm having now is when I have a SIP Client behind NAT:

When I send the INVITE from the SIP Client, the SDP is using the private LAN IP address + the advertised media port, let's admit it's 10.10.10.10:5000 http://10.10.10.10:5000 when the call establishes with a public server running kamailio + rtpengine, the actual RTP arrives from the home router public IP on a natted port, let's admit it's 100.100.100.100:65100 https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt p%3a%2f%2f100.100.100.100%3a65100&umid=39581E1E-9C53-E505-96A6-E057F22 BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-193c6dfa2da7d1ac0c 93d1a002e73781a0a22d9e, event though, RTPENGINE assumes that RTP stream is coming from 100.100.100.100 https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt p%3a%2f%2f100.100.100.100&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&au th=cbd2d006abdb2241308511f72807fe0bc6594b4a-9e365df6d282d7bf1e79e91e70 73c5a94aed6d82:5000.I'm using the '--sip-source' with RTPENGINE to make it use the received ip address instead of the private IP coming in the SDP but I'm not finding anything to make RTPENGINE adapt an start sending the RTP FLOW to the port where he starts receiving the RTP flow from, discarding the SDP media port information.

Rtpengine does this automatically, unless the `asymmetric` flag is used. You should see a log message about a `confirmed peer address` as soon as the first RTP is received on a port.

Cheers

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Message: 3 Date: Thu, 16 Jan 2020 07:23:28 -0800 From: Joel Serrano joel@textplus.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org Subject: Re: [SR-Users] Get destination IP from forwarded invite Message-ID: CAMtXxQmmRb-ieA9T+0ofxAxiOd23w_0nHos=nZOy9-JNRahnfA@mail.gmail.com Content-Type: text/plain; charset="utf-8"

Hello,

If I understand correctly your email, you want the IP/Host of the destination of the outgoing INVITE that is leaving a Kamailio box?

First of all, how are you forwarding the request? Are you using dispatcher module?

Here are a couple of options:

1- https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#du_-_destinati... (although this depends on your config, it might not be set)

2- https://www.kamailio.org/wiki/cookbooks/5.3.x/core#onsend_route + https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#sndto_name

If you can describe a little more how you are doing things we might be able to give you more information.

Joel.

On Thu, Jan 16, 2020 at 3:53 AM CUMHUR KARAHAN < cumhur.karahan@consultant.turkcell.com.tr> wrote:

Hi all,

I’m a newbie for Kamailio and working on a feature needs end-users’ destination IP in the first transferred/forwarded invite message.

Basically, our system/call flow is like:

ClientA -> Kamailio1 -> Kamailio2 -> ClientB

Invite from the clientA:

*tls:212.252.34.165:32980 https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt p%3a%2f%2f212.252.34.165%3a32980&umid=39581E1E-9C53-E505-96A6-E057F22B D18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-d4b503e38b32fd4dee8 59434f30f8c5f3e832496*

*tls:172.21.193.100:5061 http://172.21.193.100:5061*

*INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 http://sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0*

*Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport*

*From: <sip:905368601286@testsip.tvoip.turkcell.com.tr sip%3A905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx*

*To: sip:905364167881@testsip.tvoip.turkcell.com.tr sip%3A905364167881@testsip.tvoip.turkcell.com.tr*

*CSeq: 20 INVITE*

*…*

Needed Ip (Client B’s IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message.

Invite to the Kamailio2 :

*udp:172.21.193.100:5060 http://172.21.193.100:5060*

*udp:172.21.193.98:5060 http://172.21.193.98:5060*

*INVITE sip:905364167881@**10.252.180.51**:36945;transport=tls SIP/2.0*

*Record-Route: sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes*

*Record-Route: sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f 5.f572;nat=yes*

*Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0 593548557.1;i=f*

*Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;r port=32980*

*From: <sip:905368601286@testsip.tvoip.turkcell.com.tr sip%3A905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx*

*To: sip:905364167881@testsip.tvoip.turkcell.com.tr sip%3A905364167881@testsip.tvoip.turkcell.com.tr*

*CSeq: 21 INVITE*

*…*

When the first invite message receives, the dialog variables or sel(msg…) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message’s variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts.

Is there any way to get it without diving C code’s abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue.

Thanks for all...

S. Cumhur Karahan

https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt p%3a%2f%2fturkcell.li%2fncfYh&umid=39581E1E-9C53-E505-96A6-E057F22BD18 D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-bbcefdc4443d64fef54812 6bd3c5f932d722a0a4

Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir.

TURKCELL bu mesajin icerdigi bilgilerin doğruluğu veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir

Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir.

This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted.

TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL.

This e-mail has been scanned for all known computer viruses. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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Message: 4 Date: Thu, 16 Jan 2020 17:29:53 +0000 From: Nuno Miguel Reis nmreis@student.dei.uc.pt To: "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: CAGTw3V7Y8pHD+a8psPL7Tah=HTgrmvVRPhPBKB9VCuoujX_Nnw@mail.gmail.com Content-Type: text/plain; charset="utf-8"

Hi again.

Thanks for all the help and suggestions. I realized the issue happens if using kernel forwarding only. If I change rtpengine to start at userspace without the kernel module enabled everything works fine as expected. Do you have any hints on why this could be happening with the kernel module?

I'm running rtpengine like this:

usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 --listen-ng= 127.0.0.1:2223 --tos=184 --sip-source

kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 --listen-ng=127.0.0.1:2223 --tos=184 --no-fallback --sip-source

when using kernel module, I'm also setting this rules on chain INPUT under iptables:

-A INPUT -p udp -m udp --dport 5000:16383 -j RTPENGINE --id 0 -A INPUT -p udp -m udp --dport 16384:32768 -j RTPENGINE --id 0 -A INPUT -p udp -m udp --dport 49152:65535 -j RTPENGINE --id 0

Looking forward to hear from you guys.

Regards,

-- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

On Thu, Jan 16, 2020 at 1:23 PM Richard Fuchs rfuchs@sipwise.com wrote:

On 15/01/2020 13.39, Nuno Miguel Reis wrote:

Hi guys.

I'm replacing a environment where I was using kamailio + freeswitch by another where I'm adding rtpengine to the mix. One of the issues I'm having now is when I have a SIP Client behind NAT:

When I send the INVITE from the SIP Client, the SDP is using the private LAN IP address + the advertised media port, let's admit it's 10.10.10.10:5000 when the call establishes with a public server running kamailio + rtpengine, the actual RTP arrives from the home router public IP on a natted port, let's admit it's 100.100.100.100:65100, event though, RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000. I'm using the '--sip-source' with RTPENGINE to make it use the received ip address instead of the private IP coming in the SDP but I'm not finding anything to make RTPENGINE adapt an start sending the RTP FLOW to the port where he starts receiving the RTP flow from, discarding the SDP media port information.

Rtpengine does this automatically, unless the `asymmetric` flag is used. You should see a log message about a `confirmed peer address` as soon as the first RTP is received on a port.

Cheers _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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Message: 5 Date: Thu, 16 Jan 2020 20:45:04 +0100 From: Federico Santulli kam.list@nhm.it To: miconda@gmail.com, "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org Cc: "business@lists.kamailio.org" business@lists.kamailio.org Subject: Re: [SR-Users] Kamailio/RTC dinner at Fosdem 2020 Message-ID: 54CDD921-48D2-47F0-935E-0B181F037774@nhm.it Content-Type: text/plain; charset=us-ascii

Hi Daniel,

We would like to join the dinner, 3 people from US.

Kind regards.

Federico Santulli

NHM - S.R.L. Via Raffaello Sanzio, 88 81031 Aversa (CE) Italy

Il giorno 9 gen 2020, alle ore 09:52, Daniel-Constantin Mierla miconda@gmail.com ha scritto:

Hello,

Fosdem 2020 is approaching, there will be a talk from Henning about Kamailio, many other friends and related projects are presenting in the RTC Devroom (Giacomo Vacca and Federico Cabiddu, Wazo, Asterisk, Homer, Janus, Jitsi, CGRateS, Linphone, ...):

• https://fosdem.org/2020/schedule/track/real_time_communications/

I plan to be at the event and wondering if there are enough interested participants for having the traditional Kamailio/RTC dinner. If yes, Torrey will help to book a place to accommodate us (first choice could be the same restaurant like the last year, if available).

Reply if you want to join the dinner and say how many other people are joining you.

Cheers, Daniel

-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=www. kamailioworld.com&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d 006abdb2241308511f72807fe0bc6594b4a-ac4b2b5016612d36d9bf3bf438b6b436b1 f1ae8e

---

Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

------------------------------

Message: 6 Date: Thu, 16 Jan 2020 22:56:03 +0100 From: Daniel-Constantin Mierla miconda@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org, Fred Posner fred@palner.com Subject: Re: [SR-Users] http_client interface Message-ID: 4ab3422e-c3d6-7d85-fb3e-53b09bfa6c1a@gmail.com Content-Type: text/plain; charset=utf-8

The libcurl seems to offer an option for that as well:

- https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html

So it can be a small patch to the module and support it in the future.

If you look for a solution on current stable, maybe one of the embedded scripting languages can help, you can execute inline a small script in Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the http client from lua to do more complex http api queries (e.g., with specific headers, ...).

Cheers, Daniel

On 13.01.20 17:00, Fred Posner wrote:

I'd never had a reason to before, but on a deployment I need http messages to go out a specific interface. Curl has an --interface option, but I don't see how to do the same with http_client. Unless I'm missing something...

Anyone ever do something like this?

-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=www.kamail...

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Message: 7 Date: Thu, 16 Jan 2020 09:49:40 -0700 (MST) From: adeykunov adeykunov@intermedia.net To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Kamailio stop to process incoming SIP traffic via TCP. Message-ID: 1579193380553-0.post@n5.nabble.com Content-Type: text/plain; charset=us-ascii

Daniel,

Got the same issue on 5.3.1 with openssl1.1, debian9. After 3 working days of tests (about ~30-50 wss clients), suddenly we've got a lot of connections stucked in CLOSE_WAIT state. Kamailio called sig_alarm_abort() when we try to reboot.

Thanks, Andrey

-- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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Message: 8 Date: Thu, 16 Jan 2020 22:02:40 +0000 (UTC) From: Jimmy Rodley jimmyrodley@yahoo.com To: "sr-users@lists.kamailio.org" sr-users@lists.kamailio.org Subject: [SR-Users] Question about xhttp module kamailio 5.2.x Message-ID: 443353431.8467378.1579212160245@mail.yahoo.com Content-Type: text/plain; charset="utf-8"

Hi, I am on kamailio 5.2.4 and want to use the xhttp module for certain management operations. The request involves some DB queries . The DB is remote and via ODBC and might have some delay.The documentation on the xhttp module states about latency http://kamailio.org/docs/modules/devel/modules/xhttp.html%C2%A0mentioning that the request is handled in an asynchronous manner.  However, the JSONRPC-S module documentation says that this module implements the support for asynchronous RPC commands only for HTTP and HTTPS transports.

https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm1049416...

I am confused about the process under whose context the xttp request . Can there be a dedicated process to handle certain management requests which do some custom instructions like certain DB operations or initiate certain http requests ? Thanks,Rupesh

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Message: 9 Date: Thu, 16 Jan 2020 15:54:54 -0500 From: Jim Leahy rickfromsunnyvale@gmail.com To: sr-users@lists.kamailio.org Subject: [SR-Users] Trouble with 302 Redirect Message-ID: CAH3xi0-AEEj_bwLYd4OzSH96Cz8=qyrdsNWTjWnf5Sm1efrvhg@mail.gmail.com Content-Type: text/plain; charset="utf-8"

Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config:

route { # drop some things right away route(DROP);

# perform sanity check route(SANITYCHECK);

# handle registrations if(is_method("REGISTER")) { route(REGISTRATION); exit; }

# handle invites if(is_method("INVITE")) { route("INVITE"); exit; }

route(RELAY); }

route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route();

# if we can find this user in the location DB... if(lookup("location")) { # relay the invite to the new dURI t_on_failure("FOURDIGITFAIL"); t_on_reply("FOURDIGITREPLY"); route(RELAY); # go back to the main route return; }

onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } }

I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily");

But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1

Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily");

But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/66fc8ba2/attachment-0001.html

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Message: 10 Date: Thu, 16 Jan 2020 17:10:06 -0500 From: Richard Fuchs rfuchs@sipwise.com To: sr-users@lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: 0b704a3c-afc4-3ca0-8d8b-38e2d70bcae9@sipwise.com Content-Type: text/plain; charset="utf-8"; Format="flowed"

On 16/01/2020 12.29, Nuno Miguel Reis wrote:

Hi again.

Thanks for all the help and suggestions. I realized the issue happens if using kernel forwarding only. If I change rtpengine to start at userspace without the kernel module enabled everything works fine as expected. Do you have any hints on why this could be happening with the kernel module?

I'm running rtpengine like this:

usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 --listen-ng=127.0.0.1:2223 http://127.0.0.1:2223 --tos=184 --sip-source

kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 --listen-ng=127.0.0.1:2223 http://127.0.0.1:2223 --tos=184 --no-fallback --sip-source

The kernel module receives its instructions from the userspace daemon, so there's no reason there should be a difference. Can you post 1) logs and 2) the flags you use for your offers/answers?

Cheers

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Message: 11 Date: Thu, 16 Jan 2020 17:11:23 -0500 From: Fred Posner fred@palner.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org Subject: Re: [SR-Users] http_client interface Message-ID: 9808e784-3d56-d100-4c46-b136cc09ef08@palner.com Content-Type: text/plain; charset=utf-8

On 1/16/20 4:56 PM, Daniel-Constantin Mierla wrote:

The libcurl seems to offer an option for that as well:

- https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html

So it can be a small patch to the module and support it in the future.

If you look for a solution on current stable, maybe one of the embedded scripting languages can help, you can execute inline a small script in Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the http client from lua to do more complex http api queries (e.g., with specific headers, ...).

Cheers, Daniel

Thank you Daniel,

I implemented a work around (little proxy in go) but would love if the option was added for the future.

Fred Posner fred@qxork.com https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%2... Direct/SMS: +1 (336) 439-3733

Need Fred? Call Fred. 336-HEY-FRED Matrix: @fred:matrix.lod.com

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Message: 12 Date: Thu, 16 Jan 2020 17:31:43 -0500 From: Alex Balashov abalashov@evaristesys.com To: sr-users@lists.kamailio.org Subject: Re: [SR-Users] Trouble with 302 Redirect Message-ID: 20200116223143.GA857@typhoon Content-Type: text/plain; charset=us-ascii

Try doing this in a failure_route[]; onreply_route[] isn't really meant to transform replies, just to passively observe or drop them.

Also, you'll want to use append_to_reply():

https://kamailio.org/docs/modules/5.3.x/modules/textops.html#textops.f.appen...

Note that the conditions that give rise to the invocation of a failure_route are broader and somewhat more abstract than any given SIP reply -- e.g. transaction reply timeouts. So, you would want to structure your failure_route like this:

failure_route[REDIRECT_HANDLE] { if(t_is_canceled()) exit;

if(t_branch_timeout()) { # Handle the timeout case... exit; }

# Use $T_rpl to access reply code ($rs) and reply reason if needed # ($rs), per: # # https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#t_rpl_pv

if($T_rpl($rs) == 486) { append_to_reply("Contact: sip:...\r\n"); t_reply("302", "Moved Temporarily"); exit; }

# Some other case. }

-- Alex

On Thu, Jan 16, 2020 at 03:54:54PM -0500, Jim Leahy wrote:

Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config:

route { # drop some things right away route(DROP);

    # perform sanity check
    route(SANITYCHECK);

    # handle registrations
    if(is_method("REGISTER")) {
            route(REGISTRATION);
            exit;
    }

    # handle invites
    if(is_method("INVITE")) {
            route("INVITE");
            exit;
    }

    route(RELAY);

}

route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route();

    # if we can find this user in the location DB...
    if(lookup("location")) {
            # relay the invite to the new dURI
            t_on_failure("FOURDIGITFAIL");
            t_on_reply("FOURDIGITREPLY");
            route(RELAY);
            # go back to the main route
            return;
    }

onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } }

I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily");

But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1

Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily");

But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks

---

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