sr-users January 2022

sr-users@lists.kamailio.org

51 participants
47 discussions

by Björn Klasen

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

3 weeks, 5 days

Is Min-SE check broken?

by harneet singh

Hi All, I am facing an issue in understanding how the min_se should be working in kamailio. As per the SST documentation, it seems like if the min_se is configured as 500, then any value of Session-Expires OR MIN-SE if lower than 500, can be responded to by a 422. However, I strangely see the reverse happening. To investigate further, I looked in to the ki_sst_check_min() code in the master, and these seems like a potential issue. Ref Code: Inside ki_sst_check_min(), there is an if condition like below: if (sst_min_se < MIN(minse, se.interval)) { However, shouldn't it be the other way around? ie if (sst_min_se > MIN(minse, se.interval)) { because we need to send 422 if the received value(in INVITE etc) is smaller than the sst configure min_se value? I also found a different documentation, at https://git.sgu.ru/oldssu/ex-opensips/blob/cb9df8d59dbb254a9d862569fd5d11f6… where the check is as below? if (sst_min_se > MIN(minse, se.interval)) { Can someone confirm if this is broken, or my understanding is incorrect? Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

10 months, 3 weeks

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

by Daniel Tryba

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos backend. So the bug is restricted to topos-redis.

11 months, 3 weeks

Possibility of mirroring the rpm/deb.kamailio.org repos efficiently via rsync?

by Malte Schmidt

Hello, is there an rsync endpoint available or is there a possiblity of setting this up? Creating a mirror via HTTP is a rather quick and dirty solution and while the deb-repo can be mirrored using debmirror, the rpm repo is hard to sync to a non-CentOS-based machine due to missing dependencies such as yum and reposync in latest Debian-based systems. Would be great to get some input in regards to this topic. Cheers

1 year, 10 months

Problem with auth_ephemeral and parse_uri(): bad port in uri

by Vlasis Chatzistavrou

Hello, I have a problem with Kamailio 5.4.6 and auth_ephemeral. I have the following in the Kamailio configuration loadmodule "auth_ephemeral" modparam( "auth_ephemeral", "sha_algorithm", 3 ) modparam( "auth_ephemeral", "username_format", 0 ) modparam( "auth_ephemeral", "secret", 1234 ) as per https://kamailio.org/docs/modules/4.1.x/modules/auth_ephemeral.html#auth_ep… and registrations fail. In the logs we see: Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:777]: check_parse_uris(): looking up From header Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:817]: check_parse_uris(): parsing From URI Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} <core> [core/parser/parse_uri.c:1296]: parse_uri(): bad port in uri (error at char 5 in state 2) parsed: <sip:3518929:16411>(17) /<sip:3518929:1641150726@192.168.2.99> (35) Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: WARNING: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:820]: check_parse_uris(): failed to parse From uri Apparently Kamailio is confused by the timestamp following the username separated by the : character. The REGISTER message is below: REGISTER sip:192.168.2.99 SIP/2.0 Via: SIP/2.0/WSS 192.0.2.202;branch=z9hG4bK5452321 Max-Forwards: 70 To: "3518929" <sip:3518929:1641148397@192.168.2.99> From: "3518929" <sip:3518929:1641148397@192.168.2.99>;tag=ht76o8b2b6 Call-ID: phkj9mi2n3s3ju7uu3qq2f CSeq: 274 REGISTER Contact: <sip:edh7mmti@192.0.2.202;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:ca5e9372-dfa1-459a-b6ba-4398d23bd896>";expires=300 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, outbound User-Agent: Raspberry Phone (SipJS - 0.11.6) Content-Length: 0 and Kamailio parses it as sip:<username>:<port> instead of sip:<username>:<timestamp>. Is this a bug that should be reported or is there any setting that I am missing?

2 years, 1 month

by Cyril Ramière

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); } } ##### END SNIP Best regards!

2 years, 2 months

Homer7 and Kamailio 5.3

by Angelo Sipper

Hi to all, We have a Kamailio 5.3 that sends all traffic to a homer7 using siptrace module with parameters force_send_sock and hep_mode_on,1 and hep_version,3 and trace_mode,1. Everything works fine getting all traffic to Homer7 but, in case Homer7 is not reachable, Kamailio then no longer accepts traffic or registrations and it is like it crashes. When we revive Homer7 and is it again reachable accepting traffic, Kamailio then works again fine without any restart or actions. Any ideas why this may be happening? Is there anything we need to add to the config to avoid this behavior as having Homer7 now is a single point of failure for Kamailio? Thank you! Kind Regards, Angelo

2 years, 2 months

two SIP proxies

by Uwe Sigleur

2 years, 2 months

tel2sip is giving unpredictable result

by Jyoti Bansal

hello, we are using kamailio 5.3 . while establishing a call the tel uri is getting converted into sip uri sometimes incorrectly . In the sip uri some extra characters are getting appended in the invite message. Can somebody please provide some guidance on this, attaching the pcap for reference(packet no. 8). Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 years, 2 months

kamailio log dialog correlation

by marek

hi, is it possible create log of SIP dialog like https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging ? it looks like its possible like this xlog("L_INFO", "D$dlg(h_id) something to log \n"); but i must copy $dlg(h_id) in every xlog tried this modparam("xlog", "prefix", "D$dlg(h_id)") but it not works with variables any other tips/ideas? goal is to correlate logs for one dialog (call) thanks Marek

2 years, 2 months

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sr-users January 2022