Re: [OpenSIPS-Users] NAT handling on a chain of proxies
On Fri, Oct 30, 2009 at 7:54 PM, mayamatakeshi wrote:
> Hello,
> is there an easy way to check if my opensips instance is the last proxy
> before the UAS?
> I need this to avoid calling nat_uac_test/fix_nated_contact in my
> onreply_route in case I'm receiving a response relayed by another proxy.
> I was thinking I could test if i'm in the top-most Record-Route in the
> reply, but I was unable to figure out how to do it.
>
Reading the page about transformations, I think something like this would
work:
if ( $(hdr(Record-Route)[0]{uri.host}) == myself ) {
}
However, I think to implement a generic way to handle this would become very
convoluted: I was thinking that I should do this considering the possibility
of dealing with redirection and downstream proxies that would do not
record_route or would do it conditionally. So I would have to check this,
set some parameter in Record-Route, take care in case of branching and
checking the Route header etc, etc.
But I think this is a lot of trouble and will not actually happen in our
network.
So I will write my cfg just testing:
if ( !from_gw() ) {
fix_nated_contact();
}
because I just need to fix the contact of the registered users.
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Re: [OpenSIPS-Users] Channel Limit
Hi, > I am trying to limit number of channels on a called DID going to opensips. > I read the following doc: > http://www.opensips.org/Resources/DocsTutConcurrentCalls > how do I use it for inbound calls coming from PSTN to opensips and limit > simultaneous calls on it? How would you like to configure your channel limit? Static or dynamic? You need to identify the source of the number (User Part of RURI?), mabye make a lookup against a db and then test, if the channel limit is reached. Its not so complicated. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] problem with uac registration
hi! i'm developing a sip client and i'm new to using opensips (before i relied on public sip servers) i'm having some trouble with registering to opensips. i do not think it's a problem with opensips in itself, but i think you could help me solve the issue (you've been warned, this may be slightly OT ;) ) when i try to connect to the opensips server using the telephone app[2], everything goes smooth. using my app, *sometimes it works, sometimes it doesn't*. i haven't been able to determine the reason. here are the wireshark capture of telephone's request: i.8...@mdx.bregister sip:biascica.pipps.net SIP/2.0 Via: SIP/2.0/UDP 93.148.135.32:14810;rport;branch=z9hG4bKPj6oC4Dpa6zNgapScuG75X8tBMdYk.qotP Max-Forwards: 70 From: "asdomar" ;tag=Pa0-GnWltHWcRSFW9cJAZqP3ut2eBss2 To: "asdomar" Call-ID: lR8wJZYTbGIyEsVUijltUxafN4.Ce94g CSeq: 41254 REGISTER User-Agent: Telephone 0.14.3 Contact: "asdomar" Expires: 300 Content-Length: 0 while this is the capture of my app's request: REGISTER sip:biascica.pipps.net SIP/2.0 Via: SIP/2.0/UDP 93.148.135.32:5060;rport;branch=z9hG4bK54695 Max-Forwards: 70 To: From: ;tag=z9hG4bK03393413 Call-ID: 030448912...@93.148.135.32 CSeq: 1 REGISTER Contact: Expires: 3600 User-Agent: mjsip stack 1.6 Content-Length: 0 (as you can see, i'm using mjsip[2] as my sip stack) note that both apps are running on my computer (one is actually running on the android emulator), and are stun-aware (both are using stun01.sipphone.com as their stun server). is there anything that would justify the opensips server not replying to the latter request? another, connected problem is that calls from my app to telephone go out ok, whereas incoming calls, from telephone to my app, seem to get lost. i can provide opensips logs, but i think they're quite irrelevant, since when my requests get ignored, there's no mention of them in the logs. thanks asymmetric [1] http://code.google.com/p/telephone/ [2] http://www.mjsip.org ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Channel Limit
Dynamic, number of allowed channels Per DID will be stored in mysql db So how can find out how many channels a particular DID is using? On Oct 31, 2009, at 9:45 AM, Uwe Kastens wrote: > Hi, > >> I am trying to limit number of channels on a called DID going to >> opensips. >> I read the following doc: >> http://www.opensips.org/Resources/DocsTutConcurrentCalls >> how do I use it for inbound calls coming from PSTN to opensips and >> limit >> simultaneous calls on it? > > How would you like to configure your channel limit? Static or dynamic? > > You need to identify the source of the number (User Part of RURI?), > mabye make a lookup against a db and then test, if the channel limit > is > reached. > > Its not so complicated. > > BR > > Uwe > > > -- > > kiste lat: 54.322684, lon: 10.13586 > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Parallel Forking messes up Voicemail two-way audio
I am wondering if anyone has run into this issue and how it might get fixed. I am testing a Hunt Group call where the user in the location table and a number out on the PSTN both get called at the same time since they both have the same Q value. The parallel forking works just fine but due to the PSTN taking a little longer to respond to the invite the call to the location table user will always cancel before the call to the PSTN number. Because of this I see that the call to the PSTN number is still going on when the Voicemail server picks up. I think because the call to the PSTN user was still in process it messes up the Two-Way audio. You can't hear the audio coming from the Voicemail server. I know for a fact that my mediaproxy functions are set up correctly because on occasion it will work correctly. Any idea how to fix this? I tried the sleep() function in the failure route, but that didn't seem to help. -- View this message in context: http://n2.nabble.com/Parallel-Forking-messes-up-Voicemail-two-way-audio-tp3925391p3925391.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
