sr-users February 2019

sr-users@lists.kamailio.org

78 participants
99 discussions

Regex issue with subst_body
by Kristijan Vrban 21 Feb '19

21 Feb '19

Hi, in a sdp i need to rewrite a a=sendrecv to a=sendonly but onyl for the for the video stream. I made a regex like this: (?<=m=video)(?s)(.*)(a=sendrecv) Full version here: https://regex101.com/r/rovecJ/1 But if have problems to translate this into kamailio config via subst_body function. I always end with ERROR: fix_actions(): fixing failed (code=-3) Somebody sees the mistake? Kristijan

3 2

Default Modules while installing kamailio
by Prabhat Kumar 21 Feb '19

21 Feb '19

Hi, I am installing kamailio from source on my ubuntu 16.04 server. *How would i know the default modules installing automatically with kamailio?* *Also How can i install all modules(if i want to) with kamailio 5.1?* Thanks, Prabhat Kumar

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kamailio-sipcapture-daemon-config package not found for CentOS7
by Patrick Wakano 20 Feb '19

20 Feb '19

Hello list, I was trying to yum install Kamailio 5.2.1 on a CentOS 7 from the http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… repo, but the kamailio-sipcapture-daemon-config can't be installed because it depends on the kamailio-sipcapture package, which can't be found... Is anyone aware of that? Thank you, Kind regards, Patrick Wakano # yum install kamailio-sipcapture-daemon-config Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.optus.net * epel: mirror.optus.net * extras: mirror.intergrid.com.au * updates: mirror.intergrid.com.au Resolving Dependencies --> Running transaction check ---> Package kamailio-sipcapture-daemon-config.x86_64 0:5.2.1-1.el7.centos will be installed --> Processing Dependency: kamailio-sipcapture = 5.2.1 for package: kamailio-sipcapture-daemon-config-5.2.1-1.el7.centos.x86_64 --> Finished Dependency Resolution Error: Package: kamailio-sipcapture-daemon-config-5.2.1-1.el7.centos.x86_64 (home_kamailio_v5.2.x-rpms) Requires: kamailio-sipcapture = 5.2.1 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest # yum deplist kamailio-sipcapture-daemon-config Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.optus.net * epel: mirror.optus.net * extras: mirror.intergrid.com.au * updates: mirror.intergrid.com.au package: kamailio-sipcapture-daemon-config.x86_64 5.2.1-1.el7.centos dependency: /bin/sh provider: bash.x86_64 4.2.46-31.el7 dependency: kamailio-sipcapture = 5.2.1 Unsatisfied dependency

2 2

Topoh disabling
by Nicolas Breuer 20 Feb '19

20 Feb '19

Hello, Using this event route (see doc) event_route[topoh:msg-outgoing] { if($sndto(ip)=="10.1.1.10") { drop; } } Cause Feb 20 21:02:22 proxy2 /usr/sbin/kamailio[16417]: ERROR: <core> [core/parser/parse_via.c:2051]: parse_via(): bad char <´> on state 100 Feb 20 21:02:22 proxy2 /usr/sbin/kamailio[16417]: ERROR: <core> [core/parser/parse_via.c:2704]: parse_via(): parsing via on: <´#037#027#015#012X> Feb 20 21:02:22 proxy2 /usr/sbin/kamailio[16417]: ERROR: <core> [core/parser/parse_via.c:2710]: parse_via(): via parse error Feb 20 21:02:22 proxy2 /usr/sbin/kamailio[16417]: ERROR: topoh [th_msg.c:454]: th_unmask_via(): cannot find cookie in via2 Feb 20 21:02:22 proxy2 /usr/sbin/kamailio[16416]: ERROR: <core> [core/parser/parse_via.c:2051]: parse_via(): bad char <´> on state 100 Feb 20 21:02:22 proxy2 /usr/sbin/kamailio[16416]: ERROR: <core> [core/parser/parse_via.c:2704]: parse_via(): parsing via on: <´#037#027#015#012X> Feb 20 21:02:22 proxy2 /usr/sbin/kamailio[16416]: ERROR: <core> [core/parser/parse_via.c:2710]: parse_via(): via parse error Feb 20 21:02:22 proxy2 /usr/sbin/kamailio[16416]: ERROR: topoh [th_msg.c:454]: th_unmask_via(): cannot find cookie in via2 Kamailio 5.1.7

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Kamailio Asterisk Retransmitting problem
by Techinnovation 20 Feb '19

20 Feb '19

Hello, Could you please help me correctly to configure Kamailio with load balancing between two asterisk servers and topology hidding with topoh module. In attach sip trace from kamailio and my config. Currently my problem is that after call UP on Asterisk side, Asterisk send 10 times Retransmitting to <-------------> Retransmitting #5 (no NAT) to 37.148.171.162:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKf534.157d7d1321019895cc6ba367dd3ae94c.0;received=37.148.171.162 Via: SIP/2.0/UDP 37.148.171.162;branch=z9hG4bKsr-mDYdVEts1x8-yJc91MhEVMtOVMhC6pni1uai1uSAQiytLvyKkMqHB5JsSg98PMYayEyHzESE1gdA6E9C From: "testaccount" <sip:37125511039@213.21.197.50>;tag=as7abbf1f1 To: <sip:100@37.148.171.162:5060>;tag=as143173ac Call-ID: !!:1MeMfMhpfGtgfo6MQu6HfMNH1v-515SC1omDQocT1Efc1MhEVMtOVMhC6pni1uai1uSA CSeq: 102 INVITE Server: Asterisk PBX GIT-master-c83a44c Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:100@37.148.171.164:5060> Content-Type: application/sdp Require: timer Content-Length: 267 v=0 o=root 1502512559 1502512559 IN IP4 37.148.171.164 s=Asterisk PBX GIT-master-c83a44c c=IN IP4 37.148.171.164 t=0 0 m=audio 19658 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv And no ACK answer from Kamailio. After 32 sec call drops. I have call flow: Provider 213.21.197.50 -> Kamailio 37.148.171.162 -> Asterisk 37.148.171.163. kamailio -v version: kamailio 5.2.0 (x86_64/linux) 535e13 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 535e13 compiled on 01:27:51 Dec 16 2018 with gcc 4.8.5 On Asterisk side I have config [general] language=en maxexpirey=3600 defaultexpirey=3600 bindport=5060 subscribecontext=lab allowsubscribe=yes limitonpeers=yes notifyringing=yes notifyhold=yes disallow=all alwaysauthreject=yes allowguest=no allow=alaw bindaddr=37.148.171.163 t38pt_udptl=yes,redundancy faxdetect=yes directmedia=no rtptimeout=60 rtpholdtimeout=300 [kamailio] type=friend host=37.148.171.162 port=5060 nat=no qualify=no canreinvite=yes insecure=invite context=from-pstn ;directmedia=yes Many thanks ! BR, Alex -- Best regards, Techinnovation mailto:techinnovation@inbox.lv

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Installing and setup of RTP engine on kamailio
by Prabhat Kumar 20 Feb '19

20 Feb '19

Hi, How can i install and setup RTP-Engine on ubuntu server? Regards, Prabhat Kumar

3 3

Gracefully dropping all branches
by Andreas Granig 20 Feb '19

20 Feb '19

Hi all, I’ve a scenario where after lookup() I get multiple branches, and I want to drop some or all of them based on some filter criteria in a branch route before sending requests out. Problem is that if all branches are dropped, kamailio replies with some implicitly generated 500 server error since there are no more branches, but I want to send a 404 or 480 instead. Is there a way to determine if I’m processing the last active branch (so all others have been dropped already), so I can send a graceful reply like 404 or 480 instead of dropping this last branch as well if I decide I have to filter that one too? I’m thinking of a hash table table or something to track and decrease the number of branches for a call, and based on a flag (whether one of the previous branches has been sent out or all have been dropped so far) either drop the last branch or reply with an error. However I’m wondering if kamailio comes with a mechanism to determine the branch state out of the box? Thanks, Andreas

2 2

How to setup Kamailio to dispatch SIP client requests to WebRTC enabled Asterisk
by Adesh Pandey 20 Feb '19

20 Feb '19

Hi Guys, I have recently setup a Kamailio server which can accept SIP or WebSocket connections but voice is not coming in the case of SIP client. I have no idea to change avp in the incoming request, please advise how to proceed.

3 3

Topos and subst_hf
by Igor Olhovskiy 19 Feb '19

19 Feb '19

Hi! I'm using subst_hf to add some extra info to contact subst_hf("Contact", "/^(.*)sip:(.*)@(.*)$/\1sip:encei$var(extra_info)-\2@\3/i", "f"); With combination of topos it's giving me strange results like 2(26) WARNING: sanity [sanity.c:282]: check_ruri_scheme(): failed to parse request uri [sip:encei302e3734-btpsh-5c6c00ac-2e-f@172.28.0.74><sip:encei302e3734-btpsh-5c6c00ac-2e-f@172.28.0.74] Seems, it's doubling R-URI on in-dialog requests. If I'm disabling topos - all is ok. What is interesting - this can't be reproduced with remove_hf/append_hf mechanisms.

1 1

Problem with requests routing when topoh is enabled
by Leonid Fainshtein 19 Feb '19

19 Feb '19

Hi, I have Kamailio 5.1 and I'm using the topology hiding module (topoh) that is configured as the following: modparam("topoh", "mask_key", "...") modparam("topoh", "uparam_name", "....") modparam("topoh", "uri_prefix_checks", 0) Unfortunately Fanvil SIP phone modifies the remote party Contact's URI before it puts it as R-URI in the re-INVITE request. For example, The remote party Contact header: Contact: <sip:127.0.0.8;tr=sr-O2Sdn60d3qrt3otz3o7L.6pF.6mUr6lF3qpd> The re-INVITE request line: INVITE sip:127.0.0.8;tr=sr-O2Sdn60d3qrt3otz3o7L.6pF.6mUr6lF3qpd;user=phone SIP/2.0 As you can see, ";user=phone" was added. As result of that Kamailio tries to send the request to 127.0.0.8. If I disable sending 'user=phone' in Fanvil configuration then everything works well. I assumed that if uri_prefix_checks=0 then the topoh module should take rid of such URI modifications. Am I wrong? Is there any way to solve the problem except to modify the endpoint configuration? Best regards, Leonid Fainshtein

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sr-users February 2019