sr-users January 2010

sr-users@lists.kamailio.org

73 participants
127 discussions

[Kamailio-Users] mk_proxy: could not resolve hostname

by kévin l'heveder

Hi, I'm using kamailio V3.0 In the kamailio.cfg when I use the "send" function with an IP inside (as send("127.0.01:51001"), I see the following error : mk_proxy: could not resolve hostname: "127.0.01:51001" At first, kamailio was sending request to the DNS server which define in resolv.conf to resolve "127.0.01:51001" The DNS could not resolve this name then the error occured So I removed this defined DNS and now no request are sending but still have the same error. I tried to set dns and use_dns_search_list parameters to no, but that's change nothing Anybody have any ideas? Regards, Koon

14 years, 3 months

[Serusers] ser scalability

by Sabatella, Michael

I just wanted to run this by the group. I did a complete overhaul on one of my ser environments and upgraded the physical server and all software around my sip proxies. I am also maintaining a complete front end for provisioning and billing that I also moved(PHP/mysql) I am running this on a Virtual machine (vmware esxi). If I need more proxies I can clone the VM and build in new service/capacity pretty fast. I am running 0.9.6 on Redhat ES 5. I have 4 cores dedicated to this server running Xeon 2.93GHz processers and 4 GIG memory. I am running the 32 bit OS. I hardly see the processor tick up to 1%. The machine I am testing has about 300 registered endpoints in the proxy db. My questions are: 1. What is the scalability of ser processing calls? 2. Are there any limits on the number of users that can be registered at one time. Is the registration memory based and am I severely limited with 4GB of memory? 3. Is there anyone that has 1000s of registration to the proxy and if so I would like to hear how you are scaling it. What type of server are you using? 4. What are some good metrics to look at on the sip proxy to gauge its performance? 5. Is there a way to capture calls per minute? Is anyone doing this? I have call accounting set up. Thanks, Michael ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

14 years, 3 months

[Kamailio-Users] 64kbit, quad/dual core support

by mbsip

Hi all, I am wondering if kamailio do support 64-bit OS systems and dual/quad core cpu platforms? The question is raised because i am about to prepare hardware specification for kamailio server. Thanks in advance, Maciej.

14 years, 3 months

Re: [Kamailio-Users] about kamailio with sctp

by Daniel-Constantin Mierla

Hi Bruce, On 1/14/10 12:54 PM, Bruce McAlister wrote: > Hi Daniel, > > >> you may ask this one on sr-dev(a)lists.sip-router.org since the devel of >> SCTP use to be more active there. >> >> > Thanks, I will ask there too. > > >> When I implemented in in old kamailio, it was no option for me than >> using kamailio-to-kamailio SCTP channel. >> >> > When you did the testing between the two kamailio servers, how did you > enforce SCTP as the transport? > > Does t_relay offer a way to specify UDP/TCP/SCTP as the transport to relay > the messages between servers? > > if you set dst uri with transport parameter then t_relay() will use it, e.g.: $du = "sip:10.0.0.1;transport=sctp"; t_relay(); Or you can use t_relay_to_sctp() or t_relay_to() with appropriate parameters: http://kamailio.org/docs/modules/3.0.x/modules/tm.html Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

14 years, 3 months

Re: [Kamailio-Users] about kamailio with sctp

by Daniel-Constantin Mierla

Hello, On 1/14/10 2:25 AM, Bruce McAlister wrote: > Hi Klaus, > > >> I do not know. For kamailio 1.5 I would just grep for SCTP in the >> sources to get the answer. >> For kamailio 3.0 (based on sip-router with a new (I guess much better) >> SCTP implementation there is probably another method needed. A cite from >> another email from Andrei: >> >> >>> Yes, make cfg SCTP=1 and then make all will compile with SCTP support. >>> I'm currently pondering whether or not I should make the SCTP support >>> automatic (if the needed *.h files are installed compile it >>> > automatically). > > I am compiling Kamailio 3.0.0 on Solaris 10 which does have SCTP support > inherent in the OS. I did a 'grep -i sctp Makefile*' and saw that there were > checks for SCTP in the makefiles. > > I ended up building Kamailio 3.0.0 with the following command line: > > make prefix=/opt/kamailio \ > SCTP=1 \ > CC_EXTRA_OPTS=-I/usr/gnu/include \ > group_include="standard postgres presence" \ > include_modules="snmpstats perl tls" \ > all > > The build went through successfully and I can see that sctp_server.o has > been linked into the kamailio binary when the build is running. > > Does anyone know of any tool that I can use to test Kamailio using the SCTP > protocol? > you may ask this one on sr-dev(a)lists.sip-router.org since the devel of SCTP use to be more active there. When I implemented in in old kamailio, it was no option for me than using kamailio-to-kamailio SCTP channel. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

14 years, 3 months

[Kamailio-Users] running Qjsimple with kamailio 1.5 + tls

by mustafa samara

Hello i am trying Qjsimple 0.6.3 with Kamailio 1.5 with tls support, i make the user register successfully but when i am trying to invite another user i got the following error: " transport_srtp Failed generating random key: unspecified failure pjsua_call.c Error initializing media channel: Unknown error 259800 [status=259800] Dialog destroyed QjSimple: Error calling buddy: Unknown error 259800 [status=259800] " the sip setting as the following: protocol: tcp srtp : mandatory srtp requirements: tls also i got the same error if the sip setting as the following: protocol: tls srtp : mandatory srtp requirements: tls or: protocol: tls srtp : mandatory srtp requirements: sips any body help me please? -- Eng.Mustafa Al-Samara

14 years, 3 months

Re: [Kamailio-Users] about kamailio with sctp

by Klaus Darilion

Bruce McAlister schrieb: > Hi All, > > I am wanting to play with Kamailio 3.0.0 and SCTP as well. I just want to > verify that I am building it in to Kamailio properly. When I build Kamailio > do I need to have USE_SCTP=1 as an option on the make command line to enable > SCTP support? > > I can see there are updates in the ChangeLog for SCTP but I cannot find how > to enable support for it in the INSTALL/README-MODULES documentation. > > Can someone point me in the right direction please. I do not know. For kamailio 1.5 I would just grep for SCTP in the sources to get the answer. For kamailio 3.0 (based on sip-router with a new (I guess much better) SCTP implementation there is probably another method needed. A cite from another email from Andrei: > Yes, make cfg SCTP=1 and then make all will compile with SCTP support. > I'm currently pondering whether or not I should make the SCTP support > automatic (if the needed *.h files are installed compile it automatically). regards klaus > > Thanks > Bruce > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin > Mierla > Sent: 11 January 2010 09:57 > To: mustafa samara > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] about kamailio with sctp > > Hello, > > On 1/11/10 1:37 AM, mustafa samara wrote: >> Hello >> i do not know how to try sctp, imake kmailio 1.5 with sctp=1 >> could you help me? > if you want to play with sctp then I recommend to use 3.0.0 (to be > released today). It has far more improvements and new features than > older versions to SCTP. You can take it from git now: > http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git > > Some highlights about sctp and 3.0.0: > http://by-miconda.blogspot.com/2010/01/best-of-new-in-kamailio-300-15-sctp.h > tml > > Cheers, > Daniel >

14 years, 3 months

[Kamailio-Users] Rtpproxy not recording media stream

by Vikram Ragukumar

Hello, I have setup Kamailio(v1.5) and Rtpproxy(v1.2.1) on a Centos5.4 linux box. Kamailio's config file has been edited to include Nat and mysql support. With Kamailio running, i started rtpproxy using rtpproxy -l ip_addrs -s udp:127.0.0.1:7722 -a -r /var/log -F However rtpproxy does not seem to be recording any session. I read that the -a switch can be used to record all sessions. Am i missing a switch/doing something wrong while invoking rtpproxy ? Regards, Vikram.

14 years, 3 months

[Kamailio-Users] Logical operators

by Antonio Reale

Hi, I'm trying this: if(is_avp_set("$avp(s:avp)") && !isflagset(1) || is_avp_set("$avp(s:avp)") && !isflagset(2)) { ...; } The first condition before OR operator is never matched. Could you please suggest the correct syntax? Thank you very much. Regards. -- Antonio

14 years, 3 months

[Kamailio-Users] SIP proxy is expecting "Route:" header to be filled information from the last "Contact:" header ?

by Julien Chavanton

Hi all, first thank you for maintaining this list and free/Open source SIP proxy software. I am working on one interco problem with a remote SIP proxy running on Kamailio. Cisco(U1) --> Opensips(P1) --> Internet --> Kamailio(P2) --> Asterisk(U2) The remote proxy(P2) is expecting us to fill the "Route:" header taking information from the "Contact:" header, according to my research the "Route:" header as to be filled using the information found in the last "Record-route:" header. Note that the "Contact:" header IP address is the one of U2, as it is the destination UA. I think our partner using Kalamailio only interconnected UA not Proxy, since they never faced this problem. Of course I do not suspect Kalamailio to be doing something wrong, I just want to validate what we should expect, and investigate further from there. I will have to contact the technicien that as configured the Kalamailio server to validate why face this behavior. -------------------------------------------------------------------------- " 16.12.1.3 Rewriting Record-Route Header Field Values " http://www.ietf.org/rfc/rfc3261.txt -------------------------------------------------------------------------- We found a temporary solution to make it worko, where we hardcoded the replacement of the IP address in the "Route:" header with the one of the U2 subst('/^Route: <sip:P2;(.*)$/Route: <sip:U2;\1/');

14 years, 3 months

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sr-users January 2010