- sr-users - kamailio.org

SBC rejects SIP INVITE from Kamailio+topos
by José Lopes 12 Mar '18

12 Mar '18

Hello, I am using Kamailio working as edge proxy with topos module active. Kamailio is proxing requests from Freeswitch's to the SBC Metaswitch. The Freeswitch sends SIP REGISTER requests and it is ok (Contact: < sip:1003@172.29.210.164:5070>). When Freeswitch sends SIP INVITE quest, it receives a reply with Forbiden (Contact: <sip:btpsh-5aa0219a-2a32-2@172.29.210.164:5070>). You can check the sip trace at https://pastebin.com/raw/JZUNgYx5 I analysed this issue on SBC Metaswitch and it requires that the INVITE and REGISTER have the same Contact Header. I know that this is not required from SIP RFC, but that it is SBC default security policy. There is any way to configure Kamailio using topos to have the same Contact Header on INVITE and REGISTER? Best regards, José Lopes

2 1

Dialog Terminator
by Leonardo Assunção 12 Mar '18

12 Mar '18

Greetings, Does Kamailio has a method or variable that indicates who cancelled,rejected or terminated a call and, therefore ended the dialog? Currently i'm looking for the sources of BYE's, CANCEL and replies greater than 300 to first INVITE, but i would safer if such a method existed. Thanks in advance

2 1

Presence not notifying?
by Mark Boyce 11 Mar '18

11 Mar '18

Hi All Got a presence problem with what should be a very simple config. Would much appreciate someone having a look at the following and seeing if I’ve missed something obvious :-) Thanks Mark I’ve gone right back to basics and using the example at http://kb.asipto.com/kamailio:presence:k43-blf <http://kb.asipto.com/kamailio:presence:k43-blf> Changes to that example are; - Kamailio 5.1.2 - RTPEngine - NAT Enabled - JSON RPC replacing mi_rpc - HA1 Passwords Setup is; Three phone 20,21,22 all on site behind a NAT (10.1.3.x local IP) at 1.2.3.4 External IP. Phone 21 has BLF Subscriptions on 20 and 22 Domain is set as “mykamailio.com” Kamailio is an Ubuntu server “In the cloud” on 5.6.7.8 (Assuming I have got my domain & ip rewrites correct on the cut’n’paste) What’s Happening; All three phones register 21 Issues SUBSCRIBE:Dialog for 20 and 22. Which get 202 Accepted back after a 407 Proxy Auth SUBSCRIBE sip:22@mykamailio.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.84:5065;rport;branch=z9hG4bK3894730624 From: "21@SBC-2" <sip:21@mykamailio.com>;tag=1678729960 To: <sip:22@mykamailio.com> Call-ID: 1544456221(a)10.1.3.84 CSeq: 1 SUBSCRIBE Contact: <sip:21@10.1.3.84:5065> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: Yealink SIP-T26P 6.73.0.50 Expires: 1800 Event: dialog Content-Length: 0 I can see these arrive in active_watchers table; *************************** 2. row *************************** id: 2 presentity_uri: sip:20@mykamailio.com watcher_username: 21 watcher_domain: mykamailio.com to_user: 20 to_domain: mykamailio.com event: dialog event_id: to_tag: a6c744f901d779619c580c9ef48ab5e9-2651 from_tag: 1809677495 callid: 2674059037(a)10.1.3.84 local_cseq: 0 remote_cseq: 2 contact: sip:21@10.1.3.84:5065;alias=1.2.3.4~5065~1 record_route: expires: 1520711354 status: 1 reason: version: 0 socket_info: udp:5.6.7.8:5060 local_contact: sip:5.6.7.8:5060 from_user: 21 from_domain: mykamailio.com updated: -1 updated_winfo: -1 flags: 0 user_agent: Yealink SIP-T26P 6.73.0.50 If I make a call from 20 to 22 it works as expected but no notify is sent. For completeness the changes I’ve made to the kamailio.cfg file are; # diff kamailio.cfg_presence_example kamailio.cfg 6a7 > #!define WITH_NAT 119c120 < #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" --- > #!define DBURL "mysql://kamailiorw:password@localhost/test_kamailio" 169a171 > alias="mykamailio.com" 221c223 < mpath="/usr/local/lib/kamailio/modules/" --- > mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" 228c230 < loadmodule "mi_fifo.so" --- > loadmodule "jsonrpcs.so" 245d246 < loadmodule "mi_rpc.so" 279c280 < loadmodule "rtpproxy.so" --- > loadmodule "rtpengine.so" 304,307d304 < # ----- mi_fifo params ----- < modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") < < 375,376c372,374 < modparam("auth_db", "calculate_ha1", yes) < modparam("auth_db", "password_column", "password") --- > modparam("auth_db", "calculate_ha1", no) > modparam("auth_db", "password_column", "ha1") > modparam("auth_db", "password_column_2", "ha1b") 414c412 < modparam("presence", "server_address", "sip:192.168.178.54:5060" ) --- > modparam("presence", "server_address”, "sip:5.6.7.8:5060") 454c452 < modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") --- > modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") 835c833 < rtpproxy_manage("co"); --- > rtpengine_manage("co”);

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Re: [SR-Users] From URI == Myself
by Fred Posner 09 Mar '18

09 Mar '18

If you're not listening or identifying that IP and specific port as an alias, then yes, this is expected. One reason why would be to run another program on that port. Fred Posner direct: +1 (224) 334-FRED (3733) > On Mar 9, 2018, at 6:52 AM, Nicolas Breuer <Nicolas.Breuer(a)belcenter.biz> wrote: > > --_000_VI1PR0402MB359898CF4C2CDFCD4EBA5DA189DE0VI1PR0402MB3598_ > Content-Type: text/plain; charset="us-ascii" > Content-Transfer-Encoding: quoted-printable > > Hello the List, > > I saw a strange behavior on the last Kamailio v4. > > The Kamailio is binding on port 5060. > > If (from_uri=3D=3Dmyself){ > ...... > } > > > * When the From uri in the INVITE is <xxxx@myalias:5070> > * When the myalias is an alias > No issue the IF routine is executed. > * When the myalias is the ip address of the server > The IF Routine is = > NOT executed because there is a mismatch in the port (5060 >< 5070) > > > Is it the expected behavior ? > > Thanks > > > > > > > --_000_VI1PR0402MB359898CF4C2CDFCD4EBA5DA189DE0VI1PR0402MB3598_ > Content-Type: text/html; charset="us-ascii" > Con

1 0

From URI == Myself
by Nicolas Breuer 09 Mar '18

09 Mar '18

Hello the List, I saw a strange behavior on the last Kamailio v4. The Kamailio is binding on port 5060. If (from_uri==myself){ ...... } * When the From uri in the INVITE is <xxxx@myalias:5070> * When the myalias is an alias > No issue the IF routine is executed. * When the myalias is the ip address of the server > The IF Routine is NOT executed because there is a mismatch in the port (5060 >< 5070) Is it the expected behavior ? Thanks

1 0

ways to track all messages of a dialog
by Fabian Borot 09 Mar '18

09 Mar '18

I need to track all messages (SIP Requests, SIP Responses) for some SIP dialogs. Let's say I need to log something specific on each state for calls initiated from a certain IP (initial request, temporary replies, rejection with a negative response, call was accepted with 200, termination with BYE, termination with CANCEL etc. ). I am thinking on using flags but I am not sure after reading the tutorial on flags (http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-flag-operations) if that is going to work properly. The doc says: "They provide a very easy and fast way of keeping states during processing a request or during a transaction". I am afraid that the ACK if the call is accepted and the BYE won't be tracked properly because they are not part of the initial transaction. Also the doc does not mention if the flag is preserved among branches spawned during the processing of the call. The idea is for example: request_route { if (is_method("INVITE")){ if ($si == "192.168.168.100"){ #condition that would set the flag, INVITE came from "my_IP" #flag/mark dialog setflag(1); #log call from "my_IP" was received } route("INCOMING); } If(loose_route() ){ if (is_method("ACK")){ if(isflagset(1)){ #log ACK related to "my_IP" was received } } if (is_method("CANCEL")){ if(isflagset(1)){ #log CANCEL related to "my_IP" was received } } if (is_method("BYE")){ if(isflagset(1)){ #log BYE related to "my_IP" was received } } } } Then : route["INCOMING"] { # do some processing # do some processing t_onreply("INCOMING"); t_on_failure("INCOMING"); t_relay(); } onreply_route[INCOMING] { #do some processing if(isflagset(1)){ #log a reply related to "my_IP" was received } } failure_route[INCOMING] { #do some processing if(isflagset(1)){ #log a negative reply related to "my_IP" was received } } Will the flags work on this case ? My 2nd approach is to use a htable and use the call_id for my calls as the key. The on each route section (reply, failure_reply, ACK processing etc.) find the $ci on the htable and if it is there do the appropriate action. But I need to delete the key:value once the call ends or leave it autoexpire with a expire value bigger than the max_call_duration_time that we have in our system.

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sql_pvquery free result
by Ben Hood 09 Mar '18

09 Mar '18

Hi, Does sql_pvquery() clean up after itself? The other query APIs return a result handle that you clean up with sql_result_free(), but sql_pvquery() doesn’t appear to return any handle, so is it self-collecting? Cheers, Ben

2 2

Why does all the examples on configuring Kamailio install PostgreSQL
by Wilkins, Steve 09 Mar '18

09 Mar '18

Hello All, I have noticed that almost all, if not all examples of installing Kamailio install the PostgreSQL libraries. If one is using MySQL, is there a need for this? Is there something in Kamailio that requires these PostgreSQL libraries to be installed? Thank you

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Resend Authenticate REGISTER to two or more Asterisk PBX
by Social Boh 08 Mar '18

08 Mar '18

Hello, I'd like know which is the best way to resend a Authenticate REGISTER to two or more Asterisk. Repeating rewritehostport and forward commands for the number of Asterisk Servers, there is a problem with the Via Header; Kamailio add one with his IP for each forward. I try to remove the first Via before send the REGISTER to the second Asterisk without success. Any hint or tip? Thank you Regards -- --- I'm SoCIaL, MayBe

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how to check if pseudo-variable contains a String
by Laurent Schweizer 08 Mar '18

08 Mar '18

Dear all, I have a question regarding pseudo-variable, how can I check if a pseudo-variable contains a specific string ? Like to check if $ct contains "transport=tcp". Thanks BR Laurent

1 0

Info: Meeting during FOSSASIA 2018
by Daniel-Constantin Mierla 08 Mar '18

08 Mar '18

Hello, soon I am going to travel to Singapore for FOSSASIA 2018 conference (March 22-25 - https://2018.fossasia.org/), giving a talk about Kamailio. Given that it is rather rare I travel to south-east Asia, I wonder if there are other people from Kamailio community attending FOSASSIA or happen to be in the city during the event. Maybe we can plan something like a small group meetup to greet, have a drink or so and chat about the project and real time communications. Write me if you can join such a meetup. It would be great to meet some kamailians around there! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - April 16-18, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Kamailio Memory Settings
by Duarte Rocha 08 Mar '18

08 Mar '18

Hey. I have some questions about memory settings. I can see the default values with "kamailio -I" and change the current values with "kamailio -M 12 -m 128". How do i see the active settings and how do i change the default value? I want to setup a proxy capable to handle 150CPS, 6k active calls, with Dispatcher, Database, Number Normalizations and other operations. What are good settings for CPU, RAM Memory, PKG_SIZE and SHM_SIZE? Thanks for the help, Best Regards, Duarte Rocha

2 1

Failed auth
by Aidar Kamalov 08 Mar '18

08 Mar '18

Hello! I have kamailio 5.0.5 and some cisco spa 50x phones. Sometime phone can't register to kamailio. REGISTER packets with right nonce,reponse, etc gets to kamailio but it response 401 Unauthorized with code -3 (user not found). After some attempts(or time, phone send register every 1200 seconds) or after reboot phone it register succesfully Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110432]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110432]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -5) fd=voip.xxxx.org, adu=<null>, aa=<null>, ar=<null>, au=<null>, ad=<null>, aU=<null>, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110433]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110433]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -3) fd=voip.xxxx.org, adu=sip:voip.xxxx.org, aa=MD5, ar=voip.xxxx.org, au=33855, ad=, aU=33855, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110431]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110431]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -3) fd=voip.xxxx.org, adu=sip:voip.xxxx.org, aa=MD5, ar=voip.xxxx.org, au=33855, ad=, aU=33855, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] -- Aydar A. Kamalov

2 1

Issue starting with kamctl
by Ruben Lezama 08 Mar '18

08 Mar '18

Hi, I am having issues to start Kamailio using kamctl. root@kamailio2:/etc/systemd/system# kamctl restart INFO: Stopping Kamailio : ERROR: No PID file found (/var/run/kamailio.pid)! Kamailio probably not running INFO: check with 'ps axw | /bin/egrep kamailio' Using #Kamailio starts works. I can register peers and make calls between them. But I cannot update configuration. root@kamailio2:/var/log# kamailio Listening on udp: 127.0.0.1:5060 udp: 192.168.1.100:5060 tcp: 127.0.0.1:5060 tcp: 192.168.1.100:5060 Aliases: tcp: kamailio2:5060 tcp: localhost:5060 udp: kamailio2:5060 udp: localhost:5060 I would appreciate any help provided. Attached the syslog output Ruben

2 1

KeepAlive Module
by Social Boh 08 Mar '18

08 Mar '18

Hello, I'm using keepalive module with drouting module... the event route works fine: NOTICE: <script>: Gateway up: OPTIONS sip:1.2.3.4 NOTICE: <script>: Gateway up: OPTIONS sip:5.6.7.8 but kamcmd command not: *kamcmd keepalive.list* foobar 42 foobar 42 { text: foobar number: 42 } { list: { nn: 17 nn: 22 } } { uri: sip:1.2.3.4 owner: drouting last checked: Wed Feb 21 11:20:12 2018 *last up: Wed Dec 31 19:00:00 1969* *last down: Wed Feb 21 11:20:13 2018* } { uri: sip:5.6.7.8 owner: drouting last checked: Wed Feb 21 11:20:12 2018 *last up: Wed Dec 31 19:00:00 1969* *last down: Wed Feb 21 11:20:12 2018* Thank you Regards... -- --- I'm SoCIaL, MayBe

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Strongswan on same Server in EC2 as kamailio 4.4
by Jonathan Hunter 08 Mar '18

08 Mar '18

Hi Guys, I have an issue with sending a BYE message back down a strongswan VPN tunnel on a server in the Amazon EC2 environment. To get the configuration working correctly I have the Public IP address bound to a loopback virtual interface, and I also have kamailio listening to it. Calls establish without issue, but when the called party send a BYE, I cant seem to force the SIP message down the correct interface. Due to client confidentiality I need to be careful what information I send but does anyone have a similar configuration set up, and working, I am just looking for reasons as to why force_socket and mhomed=1 doesnt appear to force the BYE message via the Public Interface but the private IP address on the server as its a NAT'd environment. So essentially call establishes; CLIENT---------(VIA IPSEC)-INVITE----->(UDP)KAMAILIO PUBLIC IP-------->(UDP)KAMAILIO PRIVATE IP(TCP)------------------------->CLIENT B | | | | |<---------(VIA IPSEC)-100 TRYING----|<----------100 TRYING---------------|<----------100 TRYING----------------------------| | | | | |<---------(VIA IPSEC)-180 RINGING---|<----------180 RINGING--------------|<----------180 RINGING---------------------------| | | | | |<---------(VIA IPSEC)-200OK --------|<----------200OK -------------------|<----------200OK --------------------------------| | | | | |----------(VIA IPSEC)-ACk---------> |-----------ACk--------------------->|-----------ACk---------------------------------->| How can I ensure the BYE is forced out via the PUBLIC IP? So I do this; CLIENT---------(VIA IPSEC)-INVITE----->(UDP)KAMAILIO PUBLIC IP-------->(UDP)KAMAILIO PRIVATE IP(TCP)------------------------->CLIENT B |<---------BYE (VIA IPSEC)----------------------|<----------BYE ----------------------------|<----------BYE -------------------| As it appears to be going from Private IP and not being pushed down the IPSEC tunnel CLIENT---------(VIA IPSEC)-INVITE----->KAMAILIO PUBLIC IP-------->KAMAILIO PRIVATE IP------------------------->CLIENT B |<---------BYE ----------------------<----------BYE ----------------------------|<----------BYE -------------------| on the BYE I have tried; force_send_socket(KAMAILIO PUBLIC IP); However doesnt appear to work, is this an issue for ip routing on the server to push the BYE correctly down the IPSEC tunnel? or is it configuration related? Any comments welcome if people have had this issue before. Many thanks Jon

2 1

Question about t_branch_timeout behavior.
by José Seabra 08 Mar '18

08 Mar '18

Hi there, I have some situations where Kamailio receives replies (100 and then 180) from the other end and after 2 seconds (timer defined on function t_set_fr(0,2000)) after receive the last reply(180) the INVITE transaction enters on the failure route with variable $T_reply_code=408. What can be wrong here to kamailio generates a 408 internally after receive 1xx replies? Best Regards José Seabra

2 1

SIP header when 911 called
by Abdul Basit 08 Mar '18

08 Mar '18

Hi Team, Does kamailio has compatibility with E911, i.e. through location in SIP header when 911 called? Is there any reference document? -- regards, abdul basit

2 1

Choosing a Testing and Validation Tool
by Duarte Rocha 08 Mar '18

08 Mar '18

Hey I need to create several testing and validation processes for the deploy of an SBC and i would like to know your opinion on the best tools to use. I need to create an automated group of testing scenarios and validation of the packets sanity and SIP compliance. I also would like to validate the packets according to some parameters of mine (ex. All numbers in E164, etc). I'm looking at Star Trinity's SIP Tester right now but i'm not sure it can fulfill all my needs. I've also looked into SIPp and considered creating a Kamailio to do the validation, but i'm not sure it's worth the work if i can find a tool prepared to do this. Cheers :)

2 1

Disable non-existent user warning
by Stefan Rüetschli 08 Mar '18

08 Mar '18

Hello sr-users! I'm using Kamailio 5.1.2. If I execute the command "serctl acl grant 1234 mygrant" I get a warning which I have to accept with "Y": Non-existent user '1234'. Still proceeding? [Y|N] Is it possible to deactivate this warning? I know why this warning is there (it's not a user, it's an alias), so I think I know what I'm doing ;-) Regards Stefan

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Kamailio not send NOTIFY for the state of the subscription when using Event: dialog (BLF)
by Borislav Gergov 08 Mar '18

08 Mar '18

Hello, The initial NOTIFY that shows the server accepts our subscription is not sent: SUBSCRIBE sip:002@10.7.2.22;transport=TCP SIP/2.0. Via: SIP/2.0/TCP 10.2.4.38:63718 ;branch=z9hG4bK-524287-1---837a95d50bef81d4;rport. Max-Forwards: 70. Contact: <sip:444@10.2.4.38:63718;transport=TCP>. To: <sip:002@10.7.2.22;transport=TCP>. From: <sip:444@10.7.2.22;transport=TCP>;tag=810bbd39. Call-ID: 8fMhJ4GnMrkSu8qxez70aw... CSeq: 1 SUBSCRIBE. Expires: 120. Accept: application/dialog-info+xml. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE. User-Agent: Z. Event: dialog. Allow-Events: presence, kpml, talk. Content-Length: 0. . SIP/2.0 202 OK. Via: SIP/2.0/TCP 10.2.4.38:63718 ;branch=z9hG4bK-524287-1---837a95d50bef81d4;rport=63718;received=10.2.4.38. To: <sip:002@10.7.2.22 ;transport=TCP>;tag=339c21513a585618341888f199c2a0f6-c10d. From: <sip:444@10.7.2.22;transport=TCP>;tag=810bbd39. Call-ID: 8fMhJ4GnMrkSu8qxez70aw... CSeq: 1 SUBSCRIBE. Expires: 120. Contact: <sip:10.7.2.22:5060;transport=tcp>. Server: kamailio (5.1.1 (x86_64/linux)). Content-Length: 0. . SIP/2.0 200 OK. Via: SIP/2.0/TCP 10.2.4.38:63718 ;branch=z9hG4bK-524287-1---5b9c08901116e843;rport=63718;received=10.2.4.38. To: <sip:444@10.7.2.22 ;transport=TCP>;tag=75017b5d51cabe3de4b519c896167ee7.88a9. From: <sip:444@10.7.2.22;transport=TCP>;tag=d831f844. Call-ID: p-vce2LWcasOPThxdFZP6A... CSeq: 3 REGISTER. Contact: <sip:444@10.2.4.38:63718 ;transport=TCP;rinstance=045d4a61cf8385dc>;expires=120. Server: kamailio (5.1.1 (x86_64/linux)). Content-Length: 0. . NOTIFY sip:444@10.2.4.38:63718;transport=TCP SIP/2.0. Via: SIP/2.0/TCP 10.7.2.22;branch=z9hG4bK94d5.046c6940000000000000000000000000.0. To: <sip:444@10.7.2.22>;tag=810bbd39. From: <sip:002@10.7.2.22>;tag=339c21513a585618341888f199c2a0f6-c10d. CSeq: 2 NOTIFY. Call-ID: 8fMhJ4GnMrkSu8qxez70aw... Content-Length: 0. User-Agent: kamailio (5.1.1 (x86_64/linux)). Max-Forwards: 70. Event: dialog. Contact: <sip:10.7.2.22:5060;transport=tcp>. Subscription-State: terminated;reason=timeout. . SIP/2.0 200 OK. Via: SIP/2.0/TCP 10.7.2.22;branch=z9hG4bK94d5.046c6940000000000000000000000000.0. Contact: <sip:444@10.2.4.38:63718;transport=TCP>. To: <sip:444@10.7.2.22>;tag=810bbd39. From: <sip:002@10.7.2.22>;tag=339c21513a585618341888f199c2a0f6-c10d. Call-ID: 8fMhJ4GnMrkSu8qxez70aw... CSeq: 2 NOTIFY. User-Agent: Z. Content-Length: 0.

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consume_credentials question
by Aymeric Moizard 07 Mar '18

07 Mar '18

Hi list! I'm trying to remove Proxy-Authorization dedicated to local subscriber without -and not the other ones dedicated to other server- I realized that "consume_credentials" doesn't work because it requires that a call to "www_authorize, proxy_authorize, www_authenticate or proxy_authenticate" has been made before. I have a use-case where this isn't called. I don't want to use such code: if (is_present_hf("Proxy-Authorization")) { remove_hf("Proxy-Authorization"); } Because this may remove a "Proxy-Authorization" for another service. I have seen this: has_credentials but it doesn't help to know which header is the one containing the header... Any good way to find the header and index? And then remove it with remove _hf_value2? Regards Aymeric -- Antisip - http://www.antisip.com

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SIP RFC behaviour for 180/200 to_tag mismatch
by Mititelu Stefan 07 Mar '18

07 Mar '18

Hi people, I am trying to search for the RFC behavior in case of 180/200 to tag mismatch of an INVITE. From my readings so far, I'm not 100% convinced of the correct behavior. However, there may be a scenario, where this might be plausible: 1. parallel forked two calls 2. 180 ringing comes for first one, but not for second one(various reasons) 3. 200 ok comes for second one Any opinions about this? Do you have some references that will make this 100% clear? Thank you, Stefan

2 2

sqlops module error
by Fabian Borot 06 Mar '18

06 Mar '18

I am using the sqlops module with odbc to INSERT,UPDATE only and as the documentation says I can omit result parameter in "sql_query". However I always see these ERRORS when processing the call: This is an example of a query: sql_query("c04", "insert into SIPInfo302CDR (SIPCallID,SessionID) values('$ci',$var(SessionID))"); And these are the error lines in the log: ERROR: db_unixodbc [res.c:60]: db_unixodbc_get_columns(): no columns returned from the query ERROR: db_unixodbc [res.c:261]: db_unixodbc_convert_result(): getting column names failed ERROR: db_unixodbc [dbase.c:216]: db_unixodbc_store_result(): failed to convert result ERROR: <core> [db_query.c:188]: db_do_raw_query(): error while storing result ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [insert into SIPInfo302CDR (SIPCa] I have also tried using the parameter like shown below but the ERROR events happen anyways. sql_query("c04", "insert into SIPInfo302CDR (SIPCallID,SessionID) values('$ci',$var(SessionID))", "ra"); sql_result_free("ra"); Even though the errors line show up and say that " cannot do the query .." the query does work and the records are inserted/updated accordingly. I tried using the "sql_query_async" function that does not need the result parameter but I can't use it with unixODBC, I see these lines in the log when trying to use it: "the db driver module doesn't support async query" and the record is not inserted/updated. I also added the "async_workers" core parameter as well just in case. I also tried declaring them in the module sqlops initialization section even though in my case there is no need: "modparam("sqlops", "sqlres", "ra")" but the errors show up anyways. When I test with SEELCT query the result parameter can be used, I can count rows etc. But I am only doing INSERT/UPDATE. Am I am missing something here or this is something that can be safely used like that and I just need to ignore them. I am using " Kamailio (OpenSER) SIP Server v5.0" Thank you Fabian

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Re: [SR-Users] cnxcc module - make auth for redis connection
by Donat Zenichev 06 Mar '18

06 Mar '18

Hi Daniel, first of all thanks for your attention. Perhaps I missed something. I read documentation about cnxcc and ndb_redis again, but I haven't found information that can help me with my goal. ndb_redis is separate module that implements connections to redis databases (and also is able to perform authentication), has it's own functions like redis_cmd() to interact directly with redis. But cnxcc stores real-time amounts in redis hash keys using it's own parameter - modparam("cnxcc", "redis", "addr=127.0.0.1;port=6379;db=1") And I can't see possibility to use ndb_redis to perform authentication for cnxcc module, correct me if I'm wrong. Do you offer me to read ndb_redis code sources to use them to edit cnxcc code sources? Thanks in advance for your advice. 2018-03-01 9:28 GMT+02:00 Donat Zenichev <donat.zenichev(a)gmail.com>: > Hi Daniel, first of all thanks for your attention. > > Perhaps I missed something. I read documentation about cnxcc and ndb_redis > again, but I haven't found information that can help me with my goal. > ndb_redis is separate module that implements connections to redis > databases (and also is able to perform authentication), has it's own > functions like redis_cmd() to interact directly with redis. > > But cnxcc stores real-time amounts in redis hash keys using it's own > parameter - modparam("cnxcc", "redis", "addr=127.0.0.1;port=6379;db=1") > And I can't see possibility to use ndb_redis to perform authentication for > cnxcc module, correct me if I'm wrong. > > Do you offer me to read ndb_redis code sources to use them to edit cnxcc > code sources? > Thanks in advance for your advice. > > > > 2018-02-28 14:33 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Hello, >> >> maybe authentication support was not implemented in cnxcc. Eventually it >> can be extracted from the ndb_redis, that module has it implemented already. >> >> * https://www.kamailio.org/docs/modules/stable/modules/ndb_red >> is.html#ndb_redis.p.server >> >> Cheers, >> Daniel >> >> On 27.02.18 14:39, Donat Zenichev wrote: >> >> Nevertheless, I want to find the needed solution. >> In my opinion - If module is able to connect to redis databases, it'd >> better to perform it with authentication. >> >> Because, there are cases when db located on localhost is improper >> solution or other users mustn't see hkeys, amounts inside hkeys it etc. >> So I think no one will be mind, to have whole possibility - real-time >> billing (cnxcc) working with redis securely. >> >> So please don't be apathetic to my request. >> >> 2018-02-26 9:44 GMT+02:00 Donat Zenichev <donat.zenichev(a)gmail.com>: >> >>> Hi community. >>> This message is specially dedicated to developers of the kamailio >>> project. >>> >>> I need to find solution how to make authentication for connections of >>> cnxcc with redis. >>> For that moment, cnxcc module is able to connect to redis only without >>> password (Carlos Ruiz Díaz will add this possibility later?). >>> So that, this makes database vulnerable. >>> >>> Well, furthermore I don't want to use local redis database, cuz I think >>> machine that contains kamailio daemon, must perform just this kind of >>> actions - SIP proxying (kamailio server is not to be a database server or >>> anything else). >>> >>> For that moment we use separate redis db for cases like that, but this >>> is not best practice I think. >>> I already tried to find solution (for e.g. connections via another >>> module like db_redis: >>> https://github.com/kamailio/kamailio/pull/1432) but I got advice - >>> achieving my goal by sr-users mailing list. >>> >>> Please pay attention for my request. I think this will be very helpfull >>> for others. >>> >>> >>> >>> -- >>> -- >>> BR, Donat Zenichev >>> Wnet VoIP team >>> Tel Ukraine: +380(44) 5-900-800 >>> Tel USA: +164(67) 8-174-17 (ext 1320) >>> https://w-net.us/ <http://wnet.ua> >>> >> >> >> >> -- >> -- >> BR, Donat Zenichev >> Wnet VoIP team >> Tel Ukraine: +380(44) 5-900-800 >> Tel USA: +164(67) 8-174-17 (ext 1320) >> https://w-net.us/ <http://wnet.ua> >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com >> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com >> >> > > > -- > -- > BR, Donat Zenichev > Wnet VoIP team > Tel Ukraine: +380(44) 5-900-800 > Tel USA: +164(67) 8-174-17 (ext 1320) > https://w-net.us/ <http://wnet.ua> > -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 (ext 1320) https://w-net.us/ <http://wnet.ua>

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