sr-users October 2012

sr-users@lists.kamailio.org

87 participants
164 discussions

by Juha Heinanen

from the mailing list archives i have learned that dns_try_ipv6=yes in config file is supposed to prevent dns query on ipv6 address. if i have it "no", i get to syslog: Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [ut.h:314]: ERROR: uri2dst: failed to resolve "[2002:c062:670a::10]" :unresolvable A or AAAA request (-7) Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches and sip proxy naturally does not send anything out. if i have it "yes", request is sent out properly, but still sip proxy makes dns queries on the ipv6 address: Oct 24 14:08:35 siika /usr/sbin/sip-proxy[18196]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail why is that? how to completely prevent dns queries on ipv6 addresses? -- juha

11 years, 9 months

rtpproxy delete fails

by Juha Heinanen

i made rtpproxy test in setup where two sip phones have registered the same AoR test(a)test.fi. one is behind nat and the other is not. when i call this AoR, my sip proxy executes rtpproxy_manage("FROW3"); in branch route of the branch that is behind nat. syslog shows: Oct 19 15:49:59 siika /usr/sbin/sip-proxy[18596]: INFO: Calling rtpproxy_manage on INVITE <sip:test@192.168.0.31:5074;transport=tcp> Oct 19 15:49:59 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:48714: 18596_16 US ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm 192.98.103.10 8000 wnzdf;1 Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Creating new call Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50104/50105 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50106/50107 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Returning to SIP proxy: 18596_16 50104 192.98.103.10 4 then the uas hehind nat replies with 480 and sip proxy executes in onreply route rtpproxy_manage("FROW3"); that fails and i get to syslog: Oct 19 15:50:15 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:56183: 18594_12 D ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm wnzdf shsqn Oct 19 15:50:15 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Tags didn't match for delete message, ignoring looks like this mismatch is due to the missing ";1" in from tag param that was there when rtpproxy_manage was executed on invite. if my theory is correct, why is ";1" missing? if not correct, what did i do wrong? -- juha

11 years, 9 months

udp transport in rtpproxy sockets

by Juha Heinanen

rtpproxy module readme gives impression that udp transport is the only possibility: 4.1. rtpproxy_sock (string) Definition of socket(s) used to connect to (a set) RTPProxy. It may specify a UNIX socket or an IPv4/IPv6 UDP socket. if that is the case, what is the justification for it, i.e, why is tcp (and better yet tls) transport not supported? -- juha

11 years, 9 months

rtpproxy module flags 1 and 2 question

by Juha Heinanen

i'm new to rtpproxy (have been using mediaproxy for years) and now try to understand if it would make sense to switch over. rtpproxy readme has this on flags value 1: + 1 - append first Via branch to Call-ID when sending command to rtpproxy. ... This is especially useful if you have serially forked call scenarios where rtpproxy gets an "update" command for a new branch, and then a "delete" command for the previous branch, which would otherwise delete the full call, breaking the subsequent "lookup" for the new branch. how it would be possible in SERIAL forked scenario to get anything for a new branch before previous branch has first failed and gone? -- juha

11 years, 9 months

Manually add Via Header

by Vassilis Radis

I have a situation where a far end SIP provider doesn't behave properly when sending 487 replies. The scenario is this: I have a registered user calling into my kamailio which ,using lcr module, routes the call to a SIP provider. When the caller Cancels the call, my kamailio forwards the cancel message to the provider (along with a 200 Cancelling message to the caller). The problem is with the 487 sent by the provider to my kamailio: It does not contain all the two Via headers. It only contains the via header that names my kamailio. So when kamailio gets the 487, it removes its Via header and sends the 487 back to the caller, which now doesnt find any via header. Note that the far end provider, correctly sends all the Via Headers in other replies (183 etc). So now I am trying to "intercept" the 487 from this specific provider in the onreply route, and patch it with the header that the provider should have included. How can I do it? The problem is twofold: 1.I need to detect which header I should manually add (maybe store the Via Header set from the initial invites or the 183 replies, and get it from there? .I dont know how to do that) 2.I need to add it. (How can I manually add a Via header? .This seems easier Can anyone help me, maybe with another solution?

11 years, 9 months

ip address fields

by Juha Heinanen

lcr_gw table has: ip_addr varchar(47) trusted table has: src_ip varchar(50) address table has: ip_addr varchar(48) why is it that they are not all the same? also, why the common length is not 8 * 4 + 7 = 39 (http://en.wikipedia.org/wiki/IPv6#Address_format)? -- juha

11 years, 9 months

Kamailio & OMA Presence Compliance

by Thomas Hoffman

My colleague and I have been trying to test Kamailio configured as a Presence Server using OMA standards/messages. It appears that Kamailio does not support the OMA standards as it pertains to Presence. This is primarily in the area of the XCAP messages. Can someone please confirm if this is the case? If this is the case, it is on the roadmap to support it? If not, we'd like to know that too and we'll continue trying to get it to work. Thank you, Tom ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

11 years, 9 months

RFC5626 SIP Outbound Support of Flow-Timer Header

by Jonathan Hunter

Hi, Testing with RFC5626, can you confirm that the Flow-Timer header field is supported and configurable with Kamailio 3.3 ? Thanks Jon

11 years, 9 months

sercmd compiles and installs as kamcmd for kamailio

by Daniel-Constantin Mierla

Hello, short note to let everyone know that sercmd tool compiles and installs as kamcmd for flavour kamailio as of today in the master git branch. There was a conflict when trying to install both flavours from packages, because of overlapping sercmd file. I added a basic manpage for the tool, a recommendation for each binary app in many linux distros. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

11 years, 9 months

Registering Phones

by Re Alvarez

Hi I am setting Kamailio as SBC in my organization using Freeswitch as PBX. The integration between the two systems is working perfectly and now I am trying to add some CISCO 7970 phones to the mix. Here's a summary of the setup 1. Kamailio has 2 NICs (1 public, 1 private), RTPproxy is running 2. Freeswitch is on private network. 3. All the CISCO phones are also on the private network. We will never have a SIP phone outside our network, My question, in this case. Should I register my phones to Kamailio or Freeswitch. I am new to Kamailio and still getting down with the SIP concepts. I would prefer to register the phones with Freeswitch and only want Kamailio to get involved when calls are leaving/entering the network. If two users, registered to the same freeswitch server call each other, Kamailio should not have to do anything (ideally, not sure if its possible with SBC setup). Thanks Re

11 years, 9 months

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sr-users October 2012