sr-users August 2021

sr-users@lists.kamailio.org

49 participants
56 discussions

[VoLTE] 401 unauthorized error
by 오택경 24 Aug '21

24 Aug '21

Hi. I am implementing the VoLTE setup with the dockerized project (https://github.com/herlesupreeth/docker_open5gs https://github.com/herlesupreeth/docker_open5gs). I have almost done to run the VoLTE service, but 401 unauthorized error in sip and auth-pending error in fhoss have occured. How can I fix this problem? I will share the discussion note in which I tried to solve some problems including the above one. : https://github.com/herlesupreeth/docker_open5gs/issues/55 https://github.com/herlesupreeth/docker_open5gs/issues/55 Very thanks, Taekkyung Oh.

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Race conditions upon quick reinvite
by David VILLAUME 24 Aug '21

24 Aug '21

Hello, We encounter sometimes reinvite from some provider really quickly after 200 OK ACK so sometime there a race conditions where the re-invite exits Kamailio before the ACK leading to some Freeswitch “500 Overlapping Requests”. [cid:image001.png@01D798FA.6F5B9D80] I tried to catch the 500 on failure-route unsuccessfully as transaction is already established. Do you have any idea on how I could deal with this behavour ? Running 5.3.6 with 8 childs modules used :(tm + dialog) No costly action performed on ack that can explain slowliness. Regards, David

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kazoo error
by 崔群 24 Aug '21

24 Aug '21

Dear sir: I have some problems。the kamailio version is 5.3.0. My linux version is centos 7. the kamailio log: ERROR: kazoo [kz_amqp.c:2238]: kz_amqp_send_ex(): Failed to publish -9 : a socket error occurred ERROR: kazoo [kz_amqp.c:2707]: kz_amqp_publisher_send(): error sending publish to zone : local , connection id : 1, uri : amqp://admin The connection between kamailio and rabbitmq was interrupted, which caused the queue of MQ to be blocked and the message could not be consumed。

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Some questions about drouting module
by Андрей Яри н 23 Aug '21

23 Aug '21

Hello! I test module on Centos 7, kamailio version 5.5 and found some strange things: 1) Definition of table dr_gateways at https://kamailio.org/docs/modules/5.5.x/modules/drouting.html#idm98 - in DB there is field 'attrs' and only notice about it is module param 3.8. attrs_avp but it doesnt describe table field. In OpenSIPS docs i found description * attributes (not used by DR engine, but only pushed to script level when routing to this GW) In kamailio this field is deprecated or its only docs misses param? 2) Previous moment was found when i try to resolve: WARNING: drouting [dr_load.c:529]: dr_load_routing_info(): route <1> does not exist "Definition of dr_rules table" says that field has type of integer, but realy in database its varchar and i think its old docs. But i use KEMI question is - how i can tell module that it must be called in KEMI, not in native? 3) Every time when i call 'kamcmd drouting.reload' i get message CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: bad pointer 0x7f3e4cea04c0 (out of memory block!) called from core: core/utils/tmrec.c: tmrec_free(393) - ignoring I rollback to 5.4 and still get this error. As i understand its old module and its can be internal error - how critical this error can be? 4) Also planned commands like drouting.dump gws|gw_lists|rules? -- С уважением, Ярин Андрей Анатольевич инженер, группы управления телефонией Группа компаний «Интерсвязь» e-mail: yarin-aa(a)intersvyaz.net

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Kamailio as basic dispatcher for load balancing
by Philipp Trenz 23 Aug '21

23 Aug '21

Hey there, I’m Philipp, an IT Systems Engineer from Potsdam, Germany and want to deepen my knowledge in SIP communication on scale. Currently, my setup is quite simple: A single stateless Asterisk instance, fully managed via ARI, which registers to multiple VoIP providers, only processes incoming calls to play audio files (no outgoing or conference calls). Now, with growing load on the single Asterisk instance, I would like to have one or two Kamailio instances, which will balance the load of incoming calls to multiple Asterisk instances. On which Asterisk instance they end up is irrelevant for me, as the calls can be processed on every instance. I looked into some Kamailio and Astricon conference speeches and checked out some tutorials. But still I’m not quite sure where to start. So here are some basic questions which would really help me out to get going: As I don’t want to have registrations of all my Asterisk instances at the VoIP providers, I should use the UAC module to let Kamailio do the registering to the VoIP providers and register my Asterisk instances to Kamailio, right? Will Kamailio automatically apply new outbound registrations when added to the database or do I have to trigger that manually? Should I use one or two instances of Kamailio, how hard is it to configure them fail-safe? For writing my first config file, should I start blank or from the standard config, what’s best practice? Thanks in advance, looking forward to your replies! Philipp

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How tell if rtp packets are being relayed through rtpengine in kamailio
by Faiz Fazaluddin 23 Aug '21

23 Aug '21

Hi, I want to see if rtp packets are being relayed through kamailio. I am attmepting to connect rtpengine with kamailio, but am not sure if it is working, how can I tell? I have the kamailio server on a different VM than the clients. In wireshark, no rtp packets show on the server VM and rtp packets only seem to be going directly through the client VM. Rtp engine mentions being unable to find hashtables when attempting to implement a delete_node command. Thank you very much for your help, Faiz

3 2

21 Aug '21

Hi there! I bumped into this post to perform forwarding of REGISTER requests and then saving a local cache on 2xx replies from main Registrar: https://lists.kamailio.org/pipermail/sr-users/2020-October/110779.html I think I understand all the steps described, but some features I need are missing: - How to change the Via headers to perform topology hiding? I understand TOPOS and TOPOH do not work on these types of messages. - Change contact header so that registrar responses traverse the Kamailio box. (Use textops I suppose?) - Besides, is this approach still the best in comparison to OpenSIPs mid-registrar module? Thanks in advance, -- *Thomás Alimena Del Grande* Engenharia - Aligera Tel. 51 3500-0121

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record_route() from websocket UA to tls UA
by Juha Heinanen 20 Aug '21

20 Aug '21

I noticed that when jsSIP UA that has registered over wss calls another SIP UA that has registered over tls, record_route() adds only one Route URI to outgoing INVITE (example below). This causes BYE to fail. This issue may be caused by the fact that both UAs register over the same Kamailio tls listening socket. Still two Route URIs should be added in the same way as is done when one UA registers over tcp and the other over udp. -- Juha --------------------------------------------------------------------- 09:25:43.639833 wss:87.95.73.155:19458 wss:192.27.134.1:5061 INVITE sip:foo@test.tutpro.com SIP/2.0 Via: SIP/2.0/WSS jovobf9n4svm.invalid;branch=z9hG4bK2639638 Max-Forwards: 69 To: <sip:foo@test.tutpro.com> From: " Test" <sip:test@test.tutpro.com>;tag=f7lpsuo7cb Call-ID: ovhd9fu1fl1a66nec011 CSeq: 1387 INVITE Contact: <sip:test@test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5> ... 09:25:43.649462 tls:192.27.134.1:5061 tls:87.95.73.155:19461 INVITE sip:foo-0x793ee87a90@10.158.141.103:38378;transport=tls SIP/2.0 Record-Route: <sip:192.27.134.1:5061;transport=ws;sn=ext_tls;lr;n2;dtlsf=avp;pm=0;ice> Via: SIP/2.0/TLS 192.27.134.1:5061;branch=z9hG4bK637c.7e15ef5b84ea054cfc8ea3d4e860521f.0 Via: SIP/2.0/WSS jovobf9n4svm.invalid;rport=19458;received=87.95.73.155;branch=z9hG4bK2639638 Max-Forwards: 68 To: <sip:foo@test.tutpro.com> From: " Test" <sip:test@test.tutpro.com>;tag=f7lpsuo7cb Call-ID: ovhd9fu1fl1a66nec011 CSeq: 1387 INVITE Contact: <sip:test@test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5>

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install latest sems-server on debian 10 buster
by Mojtaba 18 Aug '21

18 Aug '21

Hello there, Does anybody have experience installing the latest SEMS-Server (1.6.0) on debian 10 buster? I tried to install both 1.6.0. and 1.7-dev and some issues are occurred, but i installed the version 1.3.1 on debian 8 before. Any help would be appreciated. Thanks with best regards -- --Mojtaba Esfandiari.S

4 11

How to insert user=phone header INVITE TO FROM
by César Matheus 18 Aug '21

18 Aug '21

Hi list. mY provider asks me to insert user = phone in my header. because it requires to receive my INVITE with user = phone in TO and FROM However, I add the user = phone in my configuration like this: $ ru = "sip:" + $ rU + "@" + $ sel (cfg_get.pstn1.gw_ip) + ":" + $ sel (cfg_get.pstn1.gw_port) + "; user = phone"; but only the INVITE appears, not in the TO or FROM. INVITE sip: 09872323232(a)X.Y.X.K: 5060; user = phone SIP / 2.0 From: <sip: XXXXYYYYY(a)172.20.12.4: 5060>; tag = 15552444533876 To: <sip: 09872323232(a)172.20.12.2: 5060> You can help me solve it. Thanks. -- César Matheus

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sr-users August 2021