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
Send sr-users mailing list submissions to
sr-users@lists.kamailio.org
To subscribe or unsubscribe via the World Wide Web, visit
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You can reach the person managing the list at
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When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..."
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_-_destination_uri
(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.kamailioworld.com&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-ac4b2b5016612d36d9bf3bf438b6b436b1f1ae8e
------------------------------
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
------------------------------
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 mentioning 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#idm1049416972
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
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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%2f%2fqxork.com&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-a38bc2851524c5b967bfb1f129c9240c6587a15a
Direct/SMS: +1 (336) 439-3733
Need Fred? Call Fred. 336-HEY-FRED
Matrix: @fred:matrix.lod.com
------------------------------
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.append_to_reply
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
> _______________________________________________
> 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: https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2f%2fwww.evaristesys.com%2f&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-ab557c94242e133aecb84497292656f0d65f2e0e, https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2f%2fwww.csrpswitch.com%2f&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-19a051a2187eac59a96885fd6635f5c3e5b9561d
------------------------------
Message: 13
Date: Fri, 17 Jan 2020 00:58:49 -0500
From: Dario Vargas <dariovargas.gcs@gmail.com>
To: sr-users@lists.kamailio.org
Subject: [SR-Users] UCaaS Platform
Message-ID:
<CAHqy4U_jecJALyvoZM6p6oGB5VyC_kZN=Sj3_vqjEhkO-tmORw@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
Take the LEAD in Unified Communications as a Service! We can help you get
started in the next Telco Disruptive Business.
White Label Hosted Cloud Platform as a Service (CPaaS) will enable you to
offer all of the below products under your own brand -
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Hosted Unified Communications (UCaaS)
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Caller Profile Solutions (Display your clients' corporate identity on cell
phones) to improve phone answer ratio.
Video Surveillance (VSaaS integrated with AWS)
Web Conferencing
Virtual FAX
Business SMS
Software developed in the USA to have all the routing features required to
connect with US and Canadian carriers for inbound and outbound (LRN, LNP,
TF, SMS, Push, Video etc).
Use our existing contracts and pricing for telco services or BYOC (bring
your own carrier model).
Alternately we can provide you with introductions to the carriers of choice
below to get started -
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Centurylink.com / Level 3 - USA DiDs and outbound and inbound - minimum
volumes required
Verizon.com - Outbound and Inbound enterprise business calls - minimum
volumes required
Dario Vargas
info@dnl-switch.com
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------------------------------
Message: 14
Date: Fri, 17 Jan 2020 07:46:13 +0000
From: Henning Westerholt <hw@skalatan.de>
To: "Kamailio (SER) - Users Mailing List"
<sr-users@lists.kamailio.org>, Dario Vargas
<dariovargas.gcs@gmail.com>
Subject: Re: [SR-Users] [commercial announcement]
Message-ID: <2dcadf18-240a-1eb4-4a46-89d1d84c6007@skalatan.de>
Content-Type: text/plain; charset="utf-8"
Hello Dario,
this type of content is not appropriate for the sr-users lists, please post it here.
Please use our dedicated business list for this kind of commercial announcements.
https://lists.kamailio.org/cgi-bin/mailman/listinfo/business
Regards,
Henning
Am 17.01.20 um 06:58 schrieb Dario Vargas:
commercial announcement
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Henning Westerholt - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%2f%2fskalatan.de%2fblog%2f&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-e3709a77a2e08cc7e9654b6f4a5353ca23af4429
Kamailio services - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%2f%2fgilawa.com&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-a187531eefe27344ba655db8d03ef71a41ad9b49
Kamailio Merchandising - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%2f%2fskalatan.de%2fmerchandising%2f&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-26d5e5e56523691d4156b2f08200b1019eb4c8f9
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