sr-users March 2014

sr-users@lists.kamailio.org

101 participants
165 discussions

by Samuel Ware

I having issue updating my allow list for the PERMISSIONS module. I added an address to the ADDRESS table. I have tried to do a service restart, kamctl address reload, and kamcmd permissions.addressReload. The kamctl address show displays the new address; however kamcmd permissions.addressDump does not neither does kamcmd permissions.subnetDump. The messages from this new address return a false to the !allow_source_address("1”) command in my routing logic. I am wondering if this is a bug or I am doing something wrong. I am on the most recent GIT version to the best of my knowledge. Sam

10 years, 7 months

Disjoint REGISTER and INVITE username/uri sets

by James Cloos

Given a proxy where the only registrations will be next-hop pbxes, where those names will not be INVITEable, what is the ideal way to tell kama the list of accepted usernames and authentication info for REGISTER? It is important to ensure that any INVITE to any of the REGISTERable usernames 404s. Thanks, -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 1024D/ED7DAEA6

10 years, 7 months

message 484

by Slava Bendersky

Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ? <--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" <sip:1300@networklab.loc>;tag=as0a530a8d To: <sip:192.168.100.145>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367(a)networklab.loc CSeq: 102 OPTIONS Contact: <sip:1300@192.168.100.145:5062>;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 Slava.

10 years, 7 months

Kamailio Freeswitch

by MrIhaveAnOpinionOnEverything

To whom it may concern: We are configuring a SIP platform with Kamailio and Freeswitch with this setup: UAC 1 ==> Kamailio ==> Freeswitch UAC 2 <== Kamailio <== Freeswitch We followed the instructions in http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc We are using Kamailio on kamailio 4.0.3 and Freeswitch FreeSWITCH Version 1.5.6b Everything is working fine with the default setting. But when we configured TLS what we noticed some errors. UAC configured with no TLS calling a UAC configured with TLS works ok. UAC configured with TLS calling a UAC configured with TLS gets disconnected 15-30 seconds after answering the call. UAC configured with TLS calling a UAC configured with no TLS gets disconnected 15-30 seconds after answering the call. I cant seem to find any error in the logs or in the ngrep. Can anyone assist me regarding this issue. Good day. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com

10 years, 7 months

Crash on REGISTER

by Igor Potjevlesch

Hello, I experienced an issue yesterday. I move a customer to Kamailio 4.0.4 (previously he was on an old SER instance without issue). He uses the same SIP account for connecting two IPBX (same platform, same firmware). When the second REGISTER came on Kamailio and, may be 500ms/1s after (I have sufficient time to see 200 OK (2 bindings) on Wireshark), Kamailio crash with coredump generated. I know that I must analyse the coredump but I hope that could be a known issue. In addition, when I try to run "gdb" on the core files, I got the following: Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m 256 -M 64'. Program terminated with signal 11, Segmentation fault. #0 0x00000030f2230f30 in escape_string_for_mysql () from /usr/lib64/mysql/libmysqlclient.so.16 Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.9-33.el6.x86_64 libcom_err-1.41.12-12.el6.x86_64 libconfuse-2.6-2.el6.rf.x86_64 libselinux-2.0.94-5.3.el6.x86_64 mysql-libs-5.1.61-4.el6.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64 openssl-1.0.0-20.el6_2.5.x86_64 zlib-1.2.3-27.el6.x86_64 I guess that I'm missing packages to troubleshoot. Many thanks. Regards, Igor.

10 years, 7 months

Dimension a kamailio server

by Rizwan Khan

HI Guys, can you refer me to some resources which help me dimension the hardware for a setup. Total users will be 20,000. What percentage should i assume for concurrent audio and video calls? What is the standard practice? How does this all map to the cpu, ram and storage etc. given that I will be storing the cdrs as well. In the futute this setup should be scalable so that it should support 200,000+ users within a span of 6-8 months. What would be the ideal architecture? Thanks, Rizwan Khan

10 years, 7 months

Intermittent Audio/video call

by Wingsravi R

Dear All, I am working on Kamailio server (V 4.1.2) with RTPproxy (1.2.1) integrated, in a standalone intranet infrastructure (no any connection with internet). I have a strange behavior with my set-up here is : --> Audio/Video calls are just go fine with only kamailio server (Without running RTPproxy instance). But with the same script and set-up sometimes i'm experiencing one-way audio/video !!. What could be the reason for this behavior ? Is it a NAT problem ? How can i solve this issue ? --> I was just suspecting it as NAT problem and installed RTPproxy server, with this all the audio/video calls are going fine. But i'm curious to know about this intermittent audio/video calls that too without any changes to Kamailio configuration and my set-up. Please below find the attachment for kamailio config file. Please anybody help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi

10 years, 7 months

Kamailio on Android

by jay binks

I realise this is a little left field, but has anyone compiled kamailio to run on Android ? ( and rtpproxy etc if you know ) would be really handy if there was some experience here. -- Sincerely Jay

10 years, 7 months

rls, rls_get_by_service_uri(): match not found

by Mihai Marin

Hello Sirs, I'm trying from some days to make RLS work but without success. I checked every character from kamailio.cfg, xml files from xcap but I can't find the problem and I'm out of ideas. My problem is the the subscribe is not done => 0 notifications with status. I think the key error is that: DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:279]: rls_get_service_list(): service uri sip:alice@93.187.138.214 not found in rl document for user sip:bob@93.187.138.214 Request: SUBSCRIBE sip:bob-list@93.187.138.214 SIP/2.0 Via: SIP/2.0/WS kvvj9i9uhvbp.invalid;branch=z9hG4bK4622972 Max-Forwards: 69 To: <sip:bob-list@93.187.138.214> From: "Bob Test" <sip:bob@93.187.138.214>;tag=e9144sdk2v Call-ID: 52k3b9gheoj6gk2vmjlrr0 CSeq: 81 SUBSCRIBE Expires: 300 Event: presence Contact: <sip:v8jmlmeu@kvvj9i9uhvbp.invalid;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:152abb8b-085d-4ca5-b658-4ecb38d1b79f>";expires=60000 Supported: eventlist Accept: multipart/related, application/pidf+xml, application/rlmi+xml, application/rpid+xml Allow-Events: refer, presence, presence.winfo, xcap-diff, conference Require: recipient-list-subscribe Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER User-Agent: JsSIP 0.3.7 Content-Length: 0 Receive: NOTIFY sip:v8jmlmeu@188.215.94.132:25702;transport=ws SIP/2.0 Via: SIP/2.0/WS 93.187.138.214:5040;branch=z9hG4bKf43b.cc179644000000000000000000000000.0 To: <sip:bob@93.187.138.214>;tag=e9144sdk2v From: < sip:bob-list@93.187.138.214>;tag=67d32ce551425daaac3772fa02abe99f-daa0 CSeq: 1 NOTIFY Call-ID: 52k3b9gheoj6gk2vmjlrr0 Content-Length: 394 User-Agent: kamailio (4.2.0-dev3 (x86_64/linux)) Max-Forwards: 70 Event: presence Contact: <sip:rls@93.187.138.214> Subscription-State: active;expires=300 Require: eventlist Content-Type: multipart/related;type="application/rlmi+xml";start="<1396291389.sip:bob-list@93.187.138.214.834374984>";boundary="TnOoxHgwMlltoGjpMlgd1kNc" --TnOoxHgwMlltoGjpMlgd1kNc Content-Transfer-Encoding: binary Content-ID: <1396291389.sip:bob-list@93.187.138.214.834374984> Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list uri=" sip:bob-list@93.187.138.214" xmlns="urn:ietf:params:xml:ns:rlmi" version="1" fullState="true"> <resource uri="sip:alice@93.187.138.214"/> </list> --TnOoxHgwMlltoGjpMlgd1kNc-- I have 2 users: bob & alice. Alice xcap files: <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Default"> <display-name>All Contacts</display-name> <entry xmlns="urn:ietf:params:xml:ns:resource-lists" uri=" sip:bob@93.187.138.214"> <display-name>bob</display-name> </entry> </list> </resource-lists> <rls-services xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns="urn:ietf:params:xml:ns:rls-services"> <service uri="sip:alice-list@93.187.138.214"> <resource-list> http://93.187.138.214:5060/xcap-root/resource-lists/users/sip:alice@93.187.… </resource-list> <packages> <package>presence</package> </packages> </service> </rls-services> Bob xcap files: <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Default"> <display-name>All Contacts</display-name> <entry xmlns="urn:ietf:params:xml:ns:resource-lists" uri=" sip:alice@93.187.138.214"> <display-name>alice</display-name> </entry> </list> </resource-lists> <rls-services xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns="urn:ietf:params:xml:ns:rls-services"> <service uri="sip:bob@93.187.138.214"> <resource-list> http://93.187.138.214:5060/xcap-root/resource-lists/users/sip:bob@93.187.13… </resource-list> <packages> <package>presence</package> </packages> </service> </rls-services> And, global file: <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> </namespaces> </xcap-caps> KAMAILIO log is here: http://pastebin.com/4XGSCPkc Kamailio route presence from config is here: http://pastebin.com/LYSr5yPz What I'm doing wrong? Are xml configured properly? Thank you. Best regards, Mihai M

10 years, 7 months

Kamailio ICSCF strange forking

by DURECU, OLIVIER (OLIVIER)

Hi all, I have deployed an IMS based on Kamailio 4.1.2 and I don't manage establishing a call. Users registration is OK but when I initiate the call from A(a)ims.net to B(a)ims.net, I notice a strange I-CSCF behavior The INVITE message goes OK from P-CSCF to S-CSCF then to I-CSCF. Till this point everything is fine... Then the I-CSCF performs its LIR diameter request and gets an Ok response from the HSS with a S-CSCF URI The I-CSCF forwards the INVITE at the received S-CSCF. BUT at the same time the I-CSCF generates a DNS request to resolve ims.net SIP SRV (it gets itself as answer...) and forwards a second INVITE (exact copy from the one sent to the S-CSCF) to itself So I have 2 copies of the same INVITE dialog travelling in the network and messing everything. Any idea of the cause of this behavior . Thanks for your support... Olivier

10 years, 7 months

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sr-users March 2014