Re: [SR-Users] Authentication on Kamailio with MySQL

:-) Kamailio's route script is brutally unforgiving in this regard. Welcome to the community! -- Alex > On Apr 10, 2017, at 5:38 PM, Nicolas Pace wrote: > >> On Mon, 2017-04-10 at 17:15 -0400, SamyGo wrote: >> Maybe a restart on kamailio service wasn't done. Or maybe you're >>

Re: [SR-Users] Kamailio 5.0 can't get version of the RTP proxy

__ > 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 -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www

[SR-Users] Exclusive: Quitting the fast life of open-source SIP

(Filed by UC Celeb Talk - 1 Apr 2017.) BERLIN (Tegel), GERMANY--In the now-infamous breakdown episode on the VIP tarmac at Berlin's Tegel Airport, Alex Balashov shocked the IP telecoms celebrity world last month. Shielding his eyes from the punishingly bright daylight of an overcast day

Re: [SR-Users] Kamailio Registration load Issue

On Tue, Mar 28, 2017 at 12:52:32AM +0500, Jade SZ wrote: > PS: Also thanks a lot for the great blog post on SIP server tuning. You're very welcome. Glad it proved useful to someone! -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)

Re: [SR-Users] Kamailio Registration load Issue

You might be running into semi-invisible/not widely publicised (but definitely very real) PPS limits on given instance sizes. On March 27, 2017 3:09:56 PM EDT, Jade SZ wrote: >Hi Guys, > >I am running a simple REGISTER load test on: > >1) Kamailio sever with 2 cores -

Re: [SR-Users] getting source IP from Contact header

you'd use: $(ct{nameaddr.uri}{uri.host}) -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) an

Re: [SR-Users] Configuration to fetch SIP response body.

base. > > Thanks, > Vivek > ___ > 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 -- Alex Balashov | Principal

Re: [SR-Users] SIP contact header question

the received reply. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mai

Re: [SR-Users] SIP contact header question

append_to_reply("Record-Route: \r\n"); But I'd like to advise against it once more. :) -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users

Re: [SR-Users] Several Asterisk on this same IP

ivr >creator. After that I would create option for record calls for client >and >this is why I look for solution :) > >2017-03-14 7:47 GMT+01:00 Alex Balashov <abalas...@evaristesys.com>: > >> Yes, though of course you would have to correlate the calls (mos

Re: [SR-Users] Several Asterisk on this same IP

lication server, and rtpproxy as media relay and recorder ? > >2017-03-14 7:44 GMT+01:00 Alex Balashov <abalas...@evaristesys.com>: > >> It can record, as can a number of other media relays. >> >> On March 14, 2017 2:43:15 AM EDT, przeqpiciel <przeqpic...@gmail.com> >

Re: [SR-Users] Several Asterisk on this same IP

en / Best regards >>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef >>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning >>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk >>> >>> On 03/13/2017 11:05 PM, Alex Balashov wrote: >>> >>&g

Re: [SR-Users] Several Asterisk on this same IP

tp, there is > >> possibilities to have 2 concurent calls each through another asterisk > >> instance with this same rtp port. Am i right? >> >> So mqybe this magic device could see source IP address and route rtp >> to correct adterisk? >> >> 13.03.2017 7

Re: [SR-Users] Several Asterisk on this same IP

Otherwise, I'm not sure how you're avoiding it, since Asterisk isn't aware of which ports from within the range are in use. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrps

Re: [SR-Users] Several Asterisk on this same IP

available to every instance. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-us

Re: [SR-Users] Several Asterisk on this same IP

Can you not assign different Asterisk instances different ranges of RTP ports to allocate from? On March 12, 2017 3:47:22 PM EDT, przeqpiciel wrote: >I would like to ask you how you deal with several asterisks and >kamailio on >that same IP address, I have installation

Re: [SR-Users] Dispatcher Module - Hava limit?

formation? The dispatcher has no such limits. Also, how did you determine that the dispatcher module is the essence of the "loss" that you are seeing? It is most likely happening in a more general context: http://blog.csrpswitch.com/tuning-kamailio-for-high-throughput-and-performance/ -- Alex

Re: [SR-Users] SIP contact header question

ver. That's on the endpoints, of course, but it can make things worse. > I'm trying to find a way to get a pesky setup to start behaving, of course, > without requiring them to make any changes. That's always a tough gig. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800

Re: [SR-Users] SIP contact header question

quests and their replies contain Route headers that are constructed from that Record-Route set. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _

Re: [SR-Users] SIP contact header question

the remote target URI. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr

Re: [SR-Users] Transaction-persistent data and uac_req_send()

Daniel, Would you accept a patch to modify the size of that buffer to something larger than 128? Like 256? On Mon, Feb 20, 2017 at 07:25:02PM -0500, Alex Balashov wrote: > On Mon, Feb 20, 2017 at 10:11:32PM +0100, Daniel-Constantin Mierla wrote: > > > There is also a way to set sor

Re: [SR-Users] Transaction-persistent data and uac_req_send()

is also in 4.4, but it was missing in the wiki docs, I just added for > devel. I think that's exactly what I need, thank you! -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://

Re: [SR-Users] Transaction-persistent data and uac_req_send()

it should be okay as long as I don't run into any arbitrary limits on the size of the username. On Mon, Feb 20, 2017 at 03:39:06PM -0500, Alex Balashov wrote: > On Mon, Feb 20, 2017 at 09:19:45PM +0100, Olle E. Johansson wrote: > > > If you configure yourself as an outbound proxy y

Re: [SR-Users] Transaction-persistent data and uac_req_send()

rictly endogenously generated request. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and Kamailio (O

Re: [SR-Users] Transaction-persistent data and uac_req_send()

On Mon, Feb 20, 2017 at 09:19:45PM +0100, Olle E. Johansson wrote: > If you configure yourself as an outbound proxy you can start a > transaction as you would normally do. So you're saying I can wrap this entire exchange in t_newtran()? -- Alex Balashov | Principal | Evariste Systems LL

Re: [SR-Users] Transaction-persistent data and uac_req_send()

that I can pass to the redirect server that it is going to send back in the 3xx response. On Mon, Feb 20, 2017 at 02:40:50PM -0500, Alex Balashov wrote: > Hello, > > I am using uac_req_send() to send INVITEs to a redirect server. However, > upon receiving the redirect, I need access t

[SR-Users] Transaction-persistent data and uac_req_send()

? If not, what are my options, other than building my own state using htable or what have you? I would really rather not maintain—and be responsible for garbage-collecting—externally constructed state like that if possible. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510

Re: [SR-Users] record_route() function

t > to both. > > Thanks. > > > -- > Andy Chen > ___ > 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 -- Alex Bal

Re: [SR-Users] nathelper module fix_nated_register(); question

ks. > > JR > -- > JR Richardson > Engineering for the Masses > Chasing the Azeotrope > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.s

[SR-Users] pua.publish using RPC

Hello, Is pua.publish strictly an MI function, or is it possible nowadays to call it via the RPC channel, and specifically, using jsonrpc_exec()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http

Re: [SR-Users] execute config route by ctp command?

0", "OK", "text/plain", ""); } else xhttp_reply("404", "Not Found", "text/html", ""); } -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (di

Re: [SR-Users] Kamailio not processing SIP TCP

ctive Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp0 0 homer02.me.com:sip*:* LISTEN > > > I don't think this is a homer issue because l

Re: [SR-Users] String split function

,$var(i),DELIMITER})) { $var(token) = $(var(str){s.select,$var(i),DELIMITER}); $var(i) = $var(i) + 1; } -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://

Re: [SR-Users] rtpproxy/rtpengine. Help to understand.

Oh, I see. Yes, that could be. -- Alex > On Dec 22, 2016, at 12:49 PM, Daniel Tryba <d.tr...@pocos.nl> wrote: > >> On Thu, Dec 22, 2016 at 12:23:52PM -0500, Alex Balashov wrote: >> That just sounds like the rtpproxy is not being engaged, i.e. that the >> rtpproxy_

Re: [SR-Users] rtpproxy/rtpengine. Help to understand.

That just sounds like the rtpproxy is not being engaged, i.e. that the rtpproxy_manage() call is failing. When that happens, the SDP from .2 will be passed through unaltered. The Kamailio log should give you some idea of why the rtpproxy invocation has failed. RTPProxy is certainly not

Re: [SR-Users] Limiting calls based on CLI and dst number

The dialog module (dialog, not dialog_ng) would be a cleaner and more natural solution, since it handles most possible eventualities of dialog state transition for you more cleanly than if you keep state yourself via htable. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1

Re: [SR-Users] question on call cancel

- sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ __

Re: [SR-Users] DTMF Using INFO

On Sat, Dec 10, 2016 at 09:07:09PM +0200, Nahum Nir wrote: > I tried Googling but didn'tfind anything. > Is it possible to support DTMF using INFO? Sure, absolutely. :-) -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web

Re: [SR-Users] kamailio bridge internal and external network

d be done > automatically. That's a good point. I forgot that mhomed= is off by default. :-) -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.cs

Re: [SR-Users] kamailio bridge internal and external network

s/rr.html#idp20509612 -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and Kamailio (Open

Re: [SR-Users] RTP Pyload

ilio. As such, it is off-topic for the list. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and

Re: [SR-Users] Limiting Registration Contacts

Wouldn't it be more fruitful to delete and reach old bindings in such a scenario? Or are these bindings not only for web clients? On December 1, 2016 5:54:06 PM EST, Colin Morelli wrote: >Hey all, > >I know Kamailio's registrar module has a max_contacts parameter that

Re: [SR-Users] Real error?

On Thu, Dec 01, 2016 at 04:27:21AM +, Olvera, Victor wrote: > Is this a real error? > > 9(1899) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error > on finding local host address > Socket operation on non-socket Is that, like, a philosophical question?

Re: [SR-Users] Kamailio Processes Stop Responding

ese queries get "stuck". If they become even a little slow, and you don't have sufficient child processes to wait on them, at 200 CPS that can have consequences. Perhaps profile the queries and/or log slow queries? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel

Re: [SR-Users] Kamailio Processes Stop Responding

g-kamailio-for-high-throughput-and-performance/ -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express R

Re: [SR-Users] custom version database/table

uld not initiate a connect to the database -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and Kamai

Re: [SR-Users] Offload SSL from backends with Kamailio

Path is indeed the exact solution for this type of problem. -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list

Re: [SR-Users] tm:local-request does not intercept locally generated CANCEL

-router.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER

[SR-Users] Message flags preservation for replies in onsend_route

(BLAH)) do_things(); } -- But without the onreply_route, it doesn't work. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _

Re: [SR-Users] async_workers vs. async workers modparam in case of async_route()

tcp main process So it is safe to remove the async_workers declaration entirely or set it to 0, and I can still benefit from 8 async processes for async_route()? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http

[SR-Users] log_prefix ignored in resumed routes

ot;L_INFO", "xyz") happens to occur in a route[] block that is resumed after async_route()—I have not tested other async functions—there is no prefix: Nov 22 00:29:52 allegro-4.evaristesys.com /usr/local/sbin/kamailio[27678]: INFO: xyz -- Alex -- Alex Balashov | Principal |

Re: [SR-Users] flags documentation

order? I would happily contribute, I just don't know how to update the core cookbook. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrp

[SR-Users] flags documentation

it still an encouraged practice? Or should one use #!defined constants instead, as in the stock config? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrp

[SR-Users] async_workers vs. async workers modparam in case of async_route()

appens if I have fewer core async_workers than modparam workers? What if I have more? Should I aim to keep them in sync? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/

Re: [SR-Users] INVITE timeout timer

Hi, fr_timer and fr_inv_timer do different things: - fr_timer goes off when there's no response _at all_ to the INVITE, not even 100 Trying. - If you received a 100 Trying or other provisional (1xx) repy, fr_inv_timer will apply instead. -- Alex -- Alex Balashov | Principal | Evariste

Re: [SR-Users] ACC module and BYE attacks

prevent it is another; that's my preferred approach. The ACC module is stateless. Think of it as a relatively unintelligent logger. As with all loggers, it's your choice what to log, and to devise means by which to prevent certain things from being logged. -- Alex Balashov | Principal | Evariste

Re: [SR-Users] ACC module and BYE attacks

s? Don't do accounting for BYEs if they don't correspond to a tracked dialog, or add a composote unique constraint on Call-ID + some other column in your database to prevent the insertion of additional events. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) /

Re: [SR-Users] digest authentication based on two password columns

You can always issue a second challenge that requires different credentials than the first. On October 28, 2016 7:41:16 AM EDT, Jayesh Nambiar wrote: >Hello, >I was wondering if it is possible to do a failover in digest >authentication. For eg: I get two possible valid

Re: [SR-Users] How do I see the full SDP?

Hi, Registration messages do not have SDP payloads. Typically, only INVITE request messages and [some of] their corresponding replies do. But you can access the encapsulated body via $rb. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250

Re: [SR-Users] Dropping non-provisional replies in onreply_route

>point. > >Cheers, >Daniel > > >On 21/10/16 11:45, Alex Balashov wrote: >> Yeah, I'm trying to avoid something complex like keeping state in >htable. >> >> I did try it - the docs are correct. drop() on a >= 2xx reply does >nothing in a named (

Re: [SR-Users] Dropping non-provisional replies in onreply_route

alternative solution is using a hash table with expiration of the >items matching the max timeout for transactions. > >Cheers, >Daniel > > >On 21/10/16 11:24, Alex Balashov wrote: >> The core documentation says that in a named onreply_route[], only >> provisional

Re: [SR-Users] Basic TM suspend question

is going to be called at the 3000 mark anyway? If so, does that mean the transaction is implicitly woken up (t_continue()'d) at that moment? 2. Does the failure_route run in a normal SIP worker, or in an async worker, if using async_task_route() here? -- Alex Balashov | Principal | Evariste

[SR-Users] Dropping non-provisional replies in onreply_route

whether to drop it, I need access to either AVPs or, ideally, dialog variables. Since the global onreply_route is executed by the core, I presume I won't have access to anything that persists through TM there. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510

[SR-Users] Basic TM suspend question

Hi, If a transaction is currently t_suspended() and has not been t_continued(), will t_check_trans() still find that transaction for retransmission-dampening purposes, e.g. if a retransmission of the same request as the original one is received? Thanks, -- Alex Balashov | Principal

Re: [SR-Users] RTPProxy

And yes, I was remiss in failing to mention that an effective solution to scaling out rtpproxy is to bind multiple instances with different core affinities. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http

Re: [SR-Users] RTPProxy

pment 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 o

Re: [SR-Users] BYE issue

Handling NAT, perhaps, but not correctly. The RURI doesn't change. But not does it have to determine where the request is actually sent on the network and transport layers. -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus.

Re: [SR-Users] BYE issue

. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users

Re: [SR-Users] RTPProxy

. It also has lots of other features which can be useful in, for example, running an RTP relay in 1:1 NAT environments such as AWS, or in enabling WebRTC. However, it is a bit more complicated to set up than vanilla rtpproxy. Not much more, though. -- Alex -- Alex Balashov | Principal | Evariste

Re: [SR-Users] 503 Service unavailable to 500

Hello, Any reply code/status translation is going to invoke the absorption of the old reply and the generation of a new one, within the proxy. Proprietary SIP headers would not be conserved in such a scenario. To add headers to a reply, you can use append_to_reply(), which works in much the

Re: [SR-Users] Order of record-route

in in-dialog requests (e.g. end-to-end ACKs, BYEs, reinvites). Thus, it is entirely appropriate that your client is trying to send the ACK to 192.168.0.200 on the network/transport layer. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250

Re: [SR-Users] Kamailio number of concurrent calls

Are you truly talking about 1000 call setups per second, or are you talking about 1000 concurrent calls total? If the latter, SIPp can handle that, with media. If the former, you're going to need a lot of endpoints. -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent

Re: [SR-Users] Kamailio number of concurrent calls

support more than a few thousand concurrent calls with RTP, you would be correct, but the same holds true of any other endpoint out there. You would need a large, complicated, horizontal farm of endpoints to establish, for example, tens of thousands of calls with two-way RTP. -- Alex Balashov

Re: [SR-Users] How to check in Kamailio if a current sip message 200 OK is a response to the BYE.

That should work, but be sure that this is executed in the context of an onreply_route and not a request_route. -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus. ___ SIP Express Router (SER) and Kamailio

Re: [SR-Users] SIREMIS MANUAL

On 09/26/2016 08:34 PM, Carlyle Edwards wrote: Hi, is there a manual for SIREMIS? http://kb.asipto.com/siremis:index -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com

Re: [SR-Users] how to config gateway in kamailio

There's no simple answer to that. You have to learn to configure Kamailio in general. On September 26, 2016 3:50:02 AM EDT, Gaurav Bmotra wrote: >Hi >i want to config gateway in kamailio >i want to know how can i config IP Authenticate gateway and user/ pass

Re: [SR-Users] Textops replace question

Igor, It might be simpler to do this: remove_hf("Contact") append_hf("Contact: <$fu>\r\n"); -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1

Re: [SR-Users] Selective rport behaviour

in the Via rather than the received source port (as would be mandated by the presence of client-injected rport)? As for trusting client headers, I strongly agree. Unfortunately, sometimes business requirements collide with our philosophies. -- Alex Balashov | Principal | Evariste Systems LLC Tel

[SR-Users] Selective rport behaviour

to the source port of the request even if force_rport() is not called, because that's the RFC 3581-compatible thing to do. Right? 2) Does reply_to_via=1 override the behaviour hypothesised in #1? 3) Does reply_to_via=1 override force_rport()? Thanks! -- Alex -- Alex Balashov | Principal | Evarist

Re: [SR-Users] Commercial SBC or Kamailio

You can also evaluate my take on whether Kamailio is interchangeable with "SBC": https://likewise.am/2013/03/10/kamailio-as-an-sbc-session-border-controller/ -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)

Re: [SR-Users] Commercial SBC or Kamailio

Hello, Best advice I can give you is to listen to this podcast, where just this very issue was discussed, among others: https://www.vuc.me/2016/vuc-609-retrospective-highlights-of-15-years-of-ser-kamailio/ -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google

Re: [SR-Users] Mitigating DDOS attacks from carrier based on ani or dnis by limitating cps or blocking when detecting

On 09/08/2016 01:00 AM, anfecora wrote: THanks Alex, do you think i can do something like where i check $FU reaching the rate limit then blocket somehow for a few minutes. Absolutely! I think you have the right idea. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800

Re: [SR-Users] Mitigating DDOS attacks from carrier based on ani or dnis by limitating cps or blocking when detecting

-- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users

Re: [SR-Users] Looking for network version of ds_reload

Not a problem. Glad it worked out! -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org

Re: [SR-Users] Centos, kamailio 4.4 and jansson module load issue.

this issue? Thanks Jon ___ 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 -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1

Re: [SR-Users] Rate limiting

to create dynamic pipes. Excellent. That was indeed an important shift. :-) -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com

Re: [SR-Users] Looking for network version of ds_reload

Found", "text/html", ""); } Of course, in the real world, you'll want to consider authentication, IP ACLs, and other things that go into some sort of security thought process. But that's the basic idea. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC T

Re: [SR-Users] To/From tags in uac_req()

4.4.2. Thanks, Julia ___ 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 -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtre

Re: [SR-Users] Trouble with nat_uac_test()

days for all cases that were handled with fix_nated_contact() in the past for just this reason. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web:

Re: [SR-Users] Kamailio SIP 500

ropping scscf list on register failure"); I_scscf_drop(); } else { xlog("L_DBG", "This is a 401 - keep scscf list to do optimisation"); } break; } } -- Alex Balashov

Re: [SR-Users] Is ACK required in a re-invite?

for 2xx responses is a problem that occurs, both for initial and in-dialog invites (reinvites). However, it indicates that something is misconfigured or improperly implemented somewhere, and is a serious issue that needs to be addressed. -- Alex Balashov | Principal | Evariste Systems LLC 1447

Re: [SR-Users] Is ACK required in a re-invite?

the same as for the initial INVITE (Section 13.2.1). -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evarist

Re: [SR-Users] access header variables in failure route

of the negative reply on the winning branch. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com

Re: [SR-Users] Problem receiving large payloads from JSONRPC-S

Huh. You're right, it's not! :-) What value should I set it to for a 0.5 MB - 1 MB payload? On August 10, 2016 3:03:32 AM EDT, Daniel-Constantin Mierla <mico...@gmail.com> wrote: > > >On 10/08/16 09:01, Alex Balashov wrote: >> I have indeed - it was in my original p

Re: [SR-Users] Problem receiving large payloads from JSONRPC-S

l > > >On 09/08/16 18:46, Alex Balashov wrote: >> FWIW, the total response size according to Content-Length is close to >> 1 MB: >> >> -- >> Header says content-length is: 737406 >> Chunked: 5660 >> Chunked: 14348 >> Chunked: 17244 >>

Re: [SR-Users] Problem receiving large payloads from JSONRPC-S

vent fired when the server (Kamailio) aborts the connection and closes the socket. -- Alex On 08/09/2016 12:27 PM, Alex Balashov wrote: Hi, I'm requesting 'dlg.list' from JSONRPC-S via XHTTP on a rather busy server, i.e. event_route[xhttp:request] { xlog("L_INFO", 'action=ER

[SR-Users] Problem receiving large payloads from JSONRPC-S

"total",[root@proxy kamailio]# -- I can't seem to figure out what parameters regulate this. I'm using the following TCP settings: tcp_async=yes tcp_connect_timeout=5 tcp_crlf_ping=no tcp_keepalive=yes tcp_keepidle=30 tcp_linger2=10 tcp_max_connections=4096 tcp_send_timeout=3 tcp_ac

Re: [SR-Users] When Not Found

On 08/08/2016 07:58 AM, Nahum Nir wrote: When sending an invite and the calle not found or responding with an ACK I want my script do some costume work. When can I do it (using regular script)? Sorry, what? :-) -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE

Re: [SR-Users] app_python access to message body content

, the app_python API is extremely limited. -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com

Re: [SR-Users] app_python access to message body content

. On the Python side: import json ... class WhateverModule: ... def handler_func(self, msg, args): obj = json.loads(args) -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920

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