[SR-Users] XML-RPC reply doesn't contain any data
Hi guys! We have an issue with XML-RPC module. >From this doc: http://kamailio.org/docs/modules/4.3.x/modules/xmlrpc.html "Success replies always contain at least one return value. In our case the simplest success replies contain single boolean with value 1" We are trying to reload domain module after adding/removing new domain, so we send the following request: XX - web server; YY - kamailio T XX.XX.XX.XX:55028 -> YY.YY.YY.YY:8080 [AP] POST /RPC2 HTTP/1.1. Host: YY.YY.YY.YY:8080. Accept: */*. Accept-Encoding: deflate, gzip. User-Agent: cURL. Accept-Charset: UTF-8. Content-Length: 120. Content-Type: application/x-www-form-urlencoded. . domain.reload T YY.YY.YY.YY:8080 -> XX.XX.XX.XX:55028 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP XX.XX.XX.XX:55028. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 108. . As you can see in the response from the kamailio there is no any data, so our script is waiting until timeout... Please advice. Thanks, -- Regards, Arsen. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] webrtc tlsv1 alert access denied
Hi guys! I am trying to configure kamailio with WSS. We have trusted certificate installed SIP over TCP/TLS works fine. But when I try WSS I got error: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL routines:SSL3_READ_BYTES:tlsv1 alert access denied ERROR: [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading Before above error it was 'bad certificate', so I have imported CA in the firefox and now I get these errors.. I have tried sipml5 and tryit.jssip.net same issue with both clients, also it seems I have these errors only when I use firefox, when I use chrome it even doesn't show me an error.. Any ideas? Thanks! -- Regards, Arsen. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] webrtc tlsv1 alert access denied
Hi Daniel, nope debug=3 doens't give more info. I have the same certificate on the web server and on the kamailio (same crt/key on both) Thanks in advance On Thu, May 26, 2016 at 5:58 PM, Daniel-Constantin Mierla wrote: > Hello, > > if you run with debug=3, do you get more hints from the debug messages? > > I guess you require client certificate in your config. > > Cheers, > Daniel > > On 26/05/16 15:06, Arsen wrote: > > Hi guys! > > I am trying to configure kamailio with WSS. > We have trusted certificate installed SIP over TCP/TLS works fine. > > But when I try WSS I got error: > > ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL > routines:SSL3_READ_BYTES:tlsv1 alert access denied > > ERROR: [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: > error reading > > Before above error it was 'bad certificate', so I have imported CA in the > firefox and now I get these errors.. > > I have tried sipml5 and tryit.jssip.net same issue with both clients, > also it seems I have these errors only when I use firefox, when I use > chrome it even doesn't show me an error.. > > Any ideas? > > Thanks! > > -- > Regards, > Arsen. > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlahttp://www.asipto.com - > http://www.kamailio.orghttp://twitter.com/#!/miconda - > http://www.linkedin.com/in/miconda > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Regards, Arsen. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] RTPProxy
Hi guys, In addition to this interesting and useful thread, what is the best way to implement media session recovery, for example in Active/Passive HA scenario? I know that it is possible with rtpengine (redis db), is it possible with rtpproxy? Thanks, Arsen. On Wed, Oct 19, 2016 at 11:19 AM, Daniel-Constantin Mierla < mico...@gmail.com> wrote: > Hello Maxim, > given the discussion here, I would like to get some updates for myself > regarding 2.0 in terms of capacity and other stuff. > > I was using rtpproxy 1.x with kamailio doing load balancing across many > instances of rtpproxy. I was using 1000 streams as estimation for one > instance and I see it's what you mentioned as well. Is it the recommended > (or the good) value for 2.0? Most of deployments still use v1.2, given it's > presence in stable/old OS distros. > > It's any relevant architectural change in 2.0? Like more threads used by > the app or other I/O refactoring? Iirc, v1.x uses one for control commands? > > I wanted to report at some point, with v1.x, on some centos (iirc), when > there was no active call, rtpproxy was eating a lot of cpu. With a call (or > more) going on, the cpu went to normal. I think it was like waiting for I/O > was using the cpu. Switching to debian was a solution at that moment, so > might not be rtpproxy, but I am wondering if you or anyone else faced same > issue. Also, if I am not wrong, the person that reported to me said that > 2.0 didn't revealed the same behaviour. > > Cheers, > Daniel > > > On 19/10/16 09:46, Maxim Sobolev wrote: > > Alex, no problem. Nobody knows everything. :) > > -Max > > On Wed, Oct 19, 2016 at 12:35 AM, Alex Balashov > wrote: > >> Hi Maxim, >> >> Duly noted! I certainly did not intend to mislead anyone or to be >> disingenuous; I gave information that was, to the best of my knowledge, >> true. I appreciate your followup and clarification, which certainly is >> useful for my own knowledge as well! >> >> My sincere apologies... >> >> -- Alex >> >> >> On October 19, 2016 3:32:24 AM EDT, Maxim Sobolev >> wrote: >> >Alex, with all due respect, things you said about rtpproxy capacity is >> >somewhat outdated and misleading. We have some nodes in the field, that >> >handle 5,000-6,000 rtp sessions in peak. Those are running 6 rtpproxy >> >instances, 1,000 sessions each. 2-3 year old CPUs, 12 cores in total. >> > >> >We also have an open source solution called rtp_cluster, which allows >> >building larger scale deployments, for at least up to 50,000 >> >bidirectional >> >streams using multiple nodes running rtpproxy. Available here >> >https://github.com/sippy/rtp_cluster. You are also welcome to check our >> >talk last summer at the opensips devsummit in Austin where we gave it >> >some >> >limelight. >> > >> >So you are off by two orders of magnitude roughly with regards to the >> >capacity. :) >> > >> >And yes, we've been happily running large deployments at AWS for at >> >least 6 >> >years now. >> > >> >Rodrigo, speaking about your original question, I could not tell much >> >about >> >rtpengine due to a lack of practical experience with it. But from what >> >I >> >read on its website it seems to be logical continuation of the >> >mediaproxy >> >package packed with some cutting edge sexy features. >> > >> >In a nutshell rtpproxy and mediaproxy/rtpengine are just two >> >independently >> >developed pieces of software, doing somewhat similar function. What >> >would >> >work in your particular setting depends on your requirements and >> >constraints. >> > >> >Here at Sippy Labs we focus on stability, compatibility and portability >> >for >> >a predominantly regular audio traffic. >> > >> >We also have a test suite that check compatibility of the latest >> >production >> >and development versions of the rtpproxy against array of different SIP >> >engines, including Kamailio. https://travis-ci.org/sippy/voiptests >> > >> >So with rtpproxy you are not locked in into single SIP engine, you can >> >mix >> >and match to fit your particular goal. >> > >> >And yes, last but not least, all our code is BSD licensed, so you can >> >build >> >you proprietary box that uses it. >> > >> >Hope it helps. >> > >> >-Max >> > >> >On Oct 17, 2016 11:33 AM, "Alex Balashov"
Re: [SR-Users] cnxcc doesn't terminate calls
Hi Igor,
Make sure that you create a dialog for this call and $dlg_var(key) is
available (This pseudo-variable will be available only for subsequential
requests after doing loose_route().)
Also check kamailio logs maybe you try to terminate call for wrong customer
id.
On Wed, Oct 26, 2016 at 10:52 AM, Igor Potjevlesch <
igor.potjevle...@gmail.com> wrote:
> Hi,
>
>
>
> I'm testing cnxcc module in order to make a simple call duration
> limitation.
>
> My understanding is to use the function like this :
>
>
>
> $dlg_var(customer_id) = "customer-" + $fU;
>
> $var(max_time) = 5;
>
>
>
> if (!cnxcc_set_max_time("$dlg_var(customer_id)",
> "$var(max_time)"))
>
>
>
> Unfortunately, after 5 seconds, nothing happen. I don't even see an error
> from dialog module.
>
>
>
> Anyone has a better understanding of the way the module runs? Thank you!
>
>
>
> Regards,
>
>
>
> Igor.
>
> ___
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
Regards,
Arsen.
___
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Re: [SR-Users] Kamailio + Asterisk with multidomain
Hi Tomas, By default in asterisk all domains are accepted and sent to the default context or the context associated with the user/peer placing the call. (from sip.conf.sample) You can specify the context per domain: domain=[,] domain=customer.com,customer-context But you probably might find it easier by checking From/To header at the asterisk side and route calls appropriately within a single dialplan context. Regards, Arsen. On Tue, Feb 7, 2017 at 9:36 AM, Tomas Zanet wrote: > Hello, thanks to this guide > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x- > asterisk-11.3.0-astdb > I successfully installed Asterisk 11.6 and Kamailio 4.4.4 on the same > machine. Everything works fine. > > Right now I would like to extend this scenario adding MULTIDOMAIN support, > which is not enabled by default > > Do you see any problems / technical limitations to add multidomain support > in this scenario? > Before enhancing Kamailio cfg file and Asterisk configuration I would like > to know: is this possible? > As far as I know Kamailio works fine with multidomain support but I don’t > know if Asterisk or both processes can support it. > > > Thanks in advance > Regards, > T. > > > > Tomas Zanet > Software Design Department > tza...@came.com > CAME S.p.A. > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Regards, Arsen. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Cannot hear voice with symmetric NAT and STUN
Hi Daniel,
I am not sure that nat_uac_test can determine type of NAT device.
and why you need all these checks if you always use rtpproxy? (another q
from 2013 :)
The idea is to reduce using of rtpproxy for better scalability and voice
quality. If we can beat all types of NAT with a near-end NAT traversal,
except symmetric NAT why do we always use proxy option in case if NAT is
detected.
On Wed, Feb 8, 2017 at 12:01 PM, Daniel Tryba wrote:
> On Wed, Feb 08, 2017 at 01:12:05AM -0700, Arsen Semionov wrote:
> > good question from 2013 :)
> > Maybe someone has experience and can confirm this?
>
> The answer to the 2013 question is: if you can depend on this (I have
> never seen it) you can script kamailio to make use of it.
>
> > My main question: is it possible to determine when it's required to use
> RTP
> > proxy ? In other words can we know that the client is behind symmetric
> NAT
> > device or we just use rtpproxy as a silver bullet?
>
> The "default" config has rtpproxy on detected NAT as an example. But
> personally I go for the always proxy option, it never failed so far
> (except for misconfigured client side firewalls that wouldn't have
> permitted p2p rtp anyway but now can easily fixed by permitting the
> traffic from the rtp range of udp ports from the rtpengine/proxy
> servers).
>
> NAT detect example below (so take a look at nat_uac_test to see what
> kind of tests there are):
>
> route[NATDETECT] {
> #!ifdef WITH_NAT
> force_rport();
> if (nat_uac_test("19")) {
> if (is_method("REGISTER")) {
> fix_nated_register();
> } else {
> if(is_first_hop())
> set_contact_alias();
> }
> setflag(FLT_NATS);
> }
> #!endif
> return;
> }
>
>
> route[NATMANAGE] {
> #!ifdef WITH_NAT
> if (is_request()) {
> if(has_totag()) {
> if(check_route_param("nat=yes")) {
> setbflag(FLB_NATB);
> }
> }
> }
> if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
> return;
>
> rtpproxy_manage("co");
>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
--
Regards,
Arsen.
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Re: [SR-Users] sip6 BYE
hG4bK3d97649d;rport=5060. > Record-Route: . > Record-Route: . > Contact: 37d7b85846910961>. > To: ;tag=d6323b02. > From: "6093756295 <(609)%20375-6295>" ; > tag=as372f68e0. > Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. > CSeq: 102 INVITE. > Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, > MESSAGE. > Content-Type: application/sdp. > Supported: replaces. > User-Agent: X-Lite release 4.9.6 stamp 82167. > Content-Length: 215. > . > v=0. > o=- 620006012 3 IN IP4 107.23.47.88. > s=X-Lite release 4.9.6 stamp 82167. > c=IN IP4 107.23.47.88. > t=0 0. > m=audio 10072 RTP/AVP 0 101. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > a=sendrecv. > a=rtcp:10073. > > # > U 107.21.18.6:5060 -> 10.0.0.68:5060 > ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 > SIP/2.0. > Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. > ff7b9cc9b887d892d7fd83e6f046b4d4.0. > Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= > z9hG4bK4e4d5063;rport=5060. > Route: . > Max-Forwards: 68. > From: "6093756295 <(609)%20375-6295>" ; > tag=as372f68e0. > To: ;tag=d6323b02. > Contact: . > Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Content-Length: 0. > . > > # > U 10.0.0.68:5060 -> 108.41.170.187:53487 > ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 > SIP/2.0. > Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. > 3a3af8e184f57334c9d42b8ce7d77849.0. > Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. > ff7b9cc9b887d892d7fd83e6f046b4d4.0. > Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= > z9hG4bK4e4d5063;rport=5060. > Max-Forwards: 66. > From: "6093756295 <(609)%20375-6295>" ; > tag=as372f68e0. > To: ;tag=d6323b02. > Contact: @54.236.97.30: > 5060;alias=107.21.18.6~5060~1>. > Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Content-Length: 0. > . > > # > U 108.41.170.187:53487 -> 10.0.0.68:5060 > . > . > > # > U 108.41.170.187:53487 -> 10.0.0.68:5060 > BYE sip:6093756295 > <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 > SIP/2.0. > Via: SIP/2.0/UDP 172.22.1.2:53487;branch=z9hG4bK-524287-1--- > 1034af445c791a63;rport. > Max-Forwards: 70. > Route: . > Route: . > Contact: 37d7b85846910961>. > To: "6093756295 <(609)%20375-6295>" ; > tag=as372f68e0. > From: ;tag=d6323b02. > Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. > CSeq: 2 BYE. > User-Agent: X-Lite release 4.9.6 stamp 82167. > Content-Length: 0. > . > > # > U 10.0.0.68:5060 -> 107.21.18.6:5060 > BYE sip:6093756295 > <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 > SIP/2.0. > Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bKbdb2. > c896050feb9a2151548b9fe6c3f776fb.0. > Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- > 524287-1---1034af445c791a63;rport=53487. > Max-Forwards: 68. > Route: . > Contact: 37d7b85846910961>. > To: "6093756295 <(609)%20375-6295>" ; > tag=as372f68e0. > From: ;tag=d6323b02. > Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. > CSeq: 2 BYE. > User-Agent: X-Lite release 4.9.6 stamp 82167. > Content-Length: 0. > . > > # > U 107.21.18.6:5060 -> 10.0.0.68:5060 > SIP/2.0 404 Not here. > Via: SIP/2.0/UDP 107.23.47.88:5060;rport=5060;branch=z9hG4bKbdb2. > c896050feb9a2151548b9fe6c3f776fb.0. > Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- > 524287-1---1034af445c791a63;rport=53487. > To: "6093756295 <(609)%20375-6295>" ; > tag=as372f68e0. > From: ;tag=d6323b02. > Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. > CSeq: 2 BYE. > Server: kamailio (4.2.5 (x86_64/linux)). > Content-Length: 0. > . > > # > U 10.0.0.68:5060 -> 108.41.170.187:53487 > SIP/2.0 404 Not here. > Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- > 524287-1---1034af445c791a63;rport=53487. > To: "6093756295 <(609)%20375-6295>" ; > tag=as372f68e0. > From: ;tag=d6323b02. > Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. > CSeq: 2 BYE. > Server: kamailio (4.2.5 (x86_64/linux)). > Content-Length: 0. > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Regards, Arsen Semionov cell: +442035198881 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] RTPPorxy -> RTPEngine migration issue
Hi everyone, i’ve been having this “not so problem” going on. So I have rtpengine installed on a server and use the default rtpproxy module on kamailio and it works beautifully. Having read the rtpengine modules description, I see that it is a drop in replacement of rtpproxy. So I keep trying to update the module names and function names but it keeps giving me this message when I start up kamailio. ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid response Software versions: Kamailio: 4.3.4 RTPEngine: 4.1.1 Any ideas on what it could be? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] RTPPorxy -> RTPEngine migration issue
Hi Richard! Here’s the output at log level 7 [1452404737.610730] WARNING: Failed to properly parse UDP command line '11683_0 d7:command4:pinge' from 10.0.0.10:54602, using fallback RE [1452404737.619006] WARNING: Failed to properly parse UDP command line '11683_1 d7:command4:pinge' from 10.0.0.10:59567, using fallback RE [1452404737.621995] WARNING: Failed to properly parse UDP command line '11694_2 d7:command4:pinge' from 10.0.0.10:57256, using fallback RE [1452404737.623204] WARNING: Failed to properly parse UDP command line '11689_1 d7:command4:pinge' from 10.0.0.10:52176, using fallback RE [1452404737.629991] WARNING: Failed to properly parse UDP command line '11687_1 d7:command4:pinge' from 10.0.0.10:55951, using fallback RE [1452404737.630268] WARNING: Failed to properly parse UDP command line '11686_1 d7:command4:pinge' from 10.0.0.10:51267, using fallback RE [1452404737.630392] WARNING: Failed to properly parse UDP command line '11690_1 d7:command4:pinge' from 10.0.0.10:55452, using fallback RE [1452404737.634052] WARNING: Failed to properly parse UDP command line '11688_1 d7:command4:pinge' from 10.0.0.10:49329, using fallback RE [1452404737.635731] WARNING: Failed to properly parse UDP command line '11693_1 d7:command4:pinge' from 10.0.0.10:43723, using fallback RE [1452404737.636118] WARNING: Failed to properly parse UDP command line '11685_1 d7:command4:pinge' from 10.0.0.10:55434, using fallback RE [1452404737.636293] WARNING: Failed to properly parse UDP command line '11692_1 d7:command4:pinge' from 10.0.0.10:34893, using fallback RE Could it be that I am starting it up wrong? Thanks a million! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] RTPPorxy -> RTPEngine migration issue
Richard that was a champion’s answer thank you so much! Maybe it should be noted kamailio's rtpengine module documentation. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Cluster
Since this thread is open, I wanted to ask, is it possible to replicate dialog data over to multiple nodes as well? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Cluster
Sorry forgot to specify (in-memory) not using the DB, I feel like using the DB for such a task would be such a drag on it ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Presence - Subscription based NOTIFY to enable MWI
Hi everyone, I’m trying to send a NOTIFY event using “sipsak” to enable the MWI I read up a lot of documentation and didn’t really find the information needed to accomplish this. So I have: Phone 1 -> P1 Server 1 -> S1 (Kamailio 4.3.4) Server 2 -> S2 (sipsak) P1 registers to S1 and creates a new entry in the active watchers table with a "message-summary” event. At this point I am assuming that any request I would send will be out of dialog. Now the question is, How could I reply to that SUBSCRIPTION with a NOTIFY? Thanks in advance! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Presence - Subscription based NOTIFY to enable MWI
I see, what i’m trying to do is actually to limit querying the DB by updating the MWI only when there is activity. In other words to send a NOTIFY when the user receives a new message in his voicemailbox. I am not sure if it’s possible to send the NOTIFY for example 5-30 mins after the initial subscribe. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Presence - Subscription based NOTIFY to enable MWI
Alright I guess this is one of those duhh… moments, all this time I was forgetting to add the content length to the NOTIFY event. Remark: DO NOT FORGET the content length! Thanks for the help, very much appreciated! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Cannot hear voice with symmetric NAT and STUN
Hi guys! good question from 2013 :) Maybe someone has experience and can confirm this? My main question: is it possible to determine when it's required to use RTP proxy ? In other words can we know that the client is behind symmetric NAT device or we just use rtpproxy as a silver bullet? Thanks! Arsen. Khoa Pham wrote > Hi all, > > When using STUN, I can detect my NAT type. The SDP contain x-NAT field (0: > unknown, 1: full cone, ..., 6: symmetric) which tells Kamailio the NAT > type > of clients. Why doesn't Kamailio use that ? > > -- > Khoa Pham > HCMC University of Science > Faculty of Information Technology - Arsen Semionov Eurolan VoIP Solutions Tel: +442035198881 -- View this message in context: http://sip-router.1086192.n5.nabble.com/Cannot-hear-voice-with-symmetric-NAT-and-STUN-tp115922p155802.html Sent from the Users mailing list archive at Nabble.com. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
