Re: [OpenSIPS-Users] Packet analysis using wireshark

ent from Outlook for Android > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 _

Re: [OpenSIPS-Users] Continously ringing

> > Sent from Outlook for Android > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://eva

Re: [OpenSIPS-Users] socket defination in opensips 3.2

t; ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 _

Re: [OpenSIPS-Users] Reg media server

ine is required? > > > -- > Regards, > B.Prathibha > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant Evariste Systems LLC Web: ht

Re: [OpenSIPS-Users] Strange ACK between OpenSIPS and Kamailio

ny hint, please? > -- > --- > I'm SoCIaL, MayBe > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant E

Re: [OpenSIPS-Users] CDR not generated on 302 redirect

> On Mar 15, 2022, at 10:13 AM, Saint Michael wrote: > > Every call that goes through Opensips should generate a record. Is this just your opinion, or…? -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaris

Re: [OpenSIPS-Users] Delayed Bye?

This is precisely due to the proportion of short-duration calls. But I agree that it is a very stupid way to approach the problem, particularly when you consider where most of the hangups in short duration calls actually come from. Some of them do you come from the originator, in the case of

Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

/sipwise/rtpengine.git But still i dont have a clue on how to solve it, any advise? Thank you Mario On Fri, Jul 17, 2020 at 2:34 PM Alex Balashov <mailto:abalas...@evaristesys.com>> wrote: This happens when an SDP body that has already been passed to RTPEngine, and already ad

Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

This happens when an SDP body that has already been passed to RTPEngine, and already adulterated by RTPEngine, is passed to it yet again. Newer versions of RTPEngine have a loop protection feature to deal with it. It involves injecting an unregistered a=rtpengine attribute into the SDP, to say

Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

ffer from all the problems of user-space in turn, so the economics can be really different. Mileage of course greatly varies with the implementation details. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: h

Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

allocator), but nothing on the order of gigabytes upon gigabytes. Assuming 4 KB per call and 200,000 concurrent calls, that's ~800 MB, and that is a very generous assumption indeed. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920

Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

a "tuning" or "optimisation" that yields a "performance increase" is profoundly misleading. -- 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/ __

Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

that the before and after don't lie. On Fri, Jun 12, 2020 at 4:02 PM Alex Balashov mailto:abalas...@evaristesys.com>> wrote: But increasing the depth of the queue by 78x (if I'm not mistaken, 212992 is the default--at least, it is on

Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

oad don't cause non-trivial packet or connection queueing on the OS side. -- 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/ ___ User

Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

can't process them fast enough isn't a solution to the problem of not processing them fast enough. You're infinitely better off just processing them faster. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http

Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

x systems, which I guess also have "absolutely terrible sysctl defaults") is faker than a Ponzi scheme. In some other contexts, this would be called morally bankrupt and intellectually fraudulent. I guess here we call it "mad dialer CPS" or whatever. -- Alex On 6/12/2

Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries

alvin.elli...@voxox.com> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://

Re: [OpenSIPS-Users] install RTPEngine on Centos 8

pensips.org/cgi-bin/mailman/listinfo/users -- 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/ ___ Users mailing list Users@lists.opensips.org ht

Re: [OpenSIPS-Users] Question

        │        ├─pickup         │        ├─qmgr         │        └─trivial-rewrite ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800

Re: [OpenSIPS-Users] fr-timer expriy

It will trigger a failure_route. — Sent from mobile, with due apologies for brevity and errors. > On Mar 15, 2020, at 4:20 AM, johan wrote: > > How can I catch in the script that fr-timer has expired ? > > I need to be able to see this expiry as I would like to failover on this. > > > BR,

Re: [OpenSIPS-Users] Ide for module development

On Mon, Aug 12, 2019 at 10:52:09AM +0200, Giovanni Maruzzelli wrote: > https://xkcd.com/378/ <3 -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswit

Re: [OpenSIPS-Users] Recover TCP Connections after restart

Unfortunately, binding to the same port will not preserve TCP connection state. On May 1, 2018 10:05:39 PM EDT, "Podrigal, Aron" wrote: >Hi everyone I hope you are having a good time @opensipsSummit I wish I >would be able to attend. > >Can anyone suggest how to

Re: [OpenSIPS-Users] Username in DB URL

ips") > > > > which fails > > > > Any ideas on a workaround on this? > > > > BR / Olle > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/

Re: [OpenSIPS-Users] Optimal trunk capacity filling algorithm or capacity-aware LCR

The simplest and lowest-hanging fruit here is profit margin protection. Don't return routes from the LCR whose cost exceeds the sell rate. This will ensure that client 1 is never routed out Trunk 2, period - if Trunk 1 is exhausted, you don't complete the calls. Then, add some logic to ensure

Re: [OpenSIPS-Users] Intercepting a 302 response and dispatching an INVITE to a new destination set

>failure_route[initial_request] { ># How can we arrive here right upon the receipt of the 302, not in >onreply_route? >} > >> On Sep 5, 2017, at 4:54 PM, Alex Balashov <abalas...@evaristesys.com> >wrote: >> >> Yes, failure_route is the answer to all your objectives h

Re: [OpenSIPS-Users] Intercepting a 302 response and dispatching an INVITE to a new destination set

Yes, failure_route is the answer to all your objectives here. You can intercept the 302, extract what you want from it, create a new branch and fork the call elsewhere. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http

Re: [OpenSIPS-Users] do_action error in logs

gt; WARNING:core:do_action: error in expression at > /etc/opensips/opensips.cfg:602 > > does anyone have any idea what is causing this error or if this flag is > even being evaluated ? > ___ > Users mailing list > Users@lists.opensips.

Re: [OpenSIPS-Users] Call continuity

, presume that the IP and port endpoints on both ends stay the same. So, if you suddenly start sending media from another place and expecting to receive it there likewise, that will not be considered to be part of the same phone call. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC

Re: [OpenSIPS-Users] Accounting of 200 OK and BYE

ave, yeah. -- 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/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bi

Re: [OpenSIPS-Users] Accounting of 200 OK and BYE

My understanding is that this is a rather simple module without sophisticated state componentry, and that it logs things immediately as received, in the same iteration of message processing. -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus.

Re: [OpenSIPS-Users] Accounting of 200 OK and BYE

r on 200 OK of BYE? Are you referring to the ACC module, or some other method of accounting? :-) -- 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: [OpenSIPS-Users] 408 timeout

OpenSIPS didn't hang up anything. On April 28, 2017 10:30:08 PM EDT, volga...@networklab.ca wrote: >Hello Everyone, >Why opensips hang up session with 408 on >SIP;cause=480;text="NO_ANSWER". I expected b2bua will send to >voicemail. >What way possible fix it. > >Please see attached trace. >

Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix

at 11:39:14PM -0400, Satish Patel wrote: > after google found bunch of post where people suggesting use > fix_nated_sdp() is that right approach ? > > On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov > <abalas...@evaristesys.com> wrote: > > Yes, but RTP can come fr

Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix

isn't public then > media never work. > > c=IN IP4 192.168.1.8. > > It should be > > c=IN IP4 > > On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov > <abalas...@evaristesys.com> wrote: > > Satish, > > > > When you say "origin public address&q

Re: [OpenSIPS-Users] SIP SDP rfc1918 address fix

_ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- 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: [OpenSIPS-Users] rtpengine database support

How do you view Redis? On April 4, 2017 4:03:31 AM EDT, Ziv Gabel wrote: >Hi All, >Is there any intention to add database support to the rtpengine module >? >The same as rtpproxy. > >*Ziv Gabel *l Professional services l CommuniTake Technologies Ltd. > > > >*M*:

Re: [OpenSIPS-Users] port number in record-route

Record-Route headers contain URIs, and like any SIP URI, they can contain a port component. If that port component is omitted, 5060 is presumed. On March 29, 2017 6:27:30 PM EDT, Satish Patel wrote: >what is the use of port number in record-route? > >I am having major

Re: [OpenSIPS-Users] Monitoring Mediaproxy

If MediaProxy is a SIP endpoint, that would be news to me. -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus. ___ Users mailing list Users@lists.opensips.org

Re: [OpenSIPS-Users] Minimum SIP Headers

at the UA layer where failure to adhere to them does not adversely affect the proxy's ability to relay the message. -- 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: [OpenSIPS-Users] Remove to-tag from 1XX provisional responses

On 10/21/2016 06:36 PM, Newlin, Ben wrote: Not only that, but provisional responses (except 100 Trying) are required to have a To tag [1]. So you would likely run into issues with UAs if you start returning messages without them. That is an astute point. -- Alex Balashov | Principal

Re: [OpenSIPS-Users] Remove to-tag from 1XX provisional responses

result on subsequent messages. Do you guys see any problem on removing the to-tag of all 1XX messages? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov | Principal

Re: [OpenSIPS-Users] Help with big amount of data for routing

___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/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: [OpenSIPS-Users] reject 3xx code

It can certainly be done in the onreply_route. Conceptually, doing it there makes more sense. The failure_route is not triggered by any particular SIP reply per se, but rather a branch failure event on a transaction. That can be the result of a timeout (e.g. fr_timer) or something else that

Re: [OpenSIPS-Users] Preventing invalid packet parsing

nnecessary work to parse the whole message if the first line doesn't suggest it's SIP. I wouldn't worry about this "premature optimisation". :-) -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920

Re: [OpenSIPS-Users] Preventing invalid packet parsing

stinfo/users -- 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: [OpenSIPS-Users] contact after fix_nated_contact()

. -- 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/ ___ Users mailing

Re: [OpenSIPS-Users] contact after fix_nated_contact()

uot;); You can run msg_apply_changes() after calling fix_nated_contact(), assuming it doesn't have any effects harmful to your cause: http://kamailio.org/docs/modules/4.4.x/modules/textopsx.html#textopsx.f.msg_apply_changes -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447

Re: [OpenSIPS-Users] How to overcome SIP ALG on Wi-Fi routers

server would then send this data to OpenSIPS over UDP. This sounds substantially similar to a VPN, except without the benefit of encryption. -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free

Re: [OpenSIPS-Users] Memory fragments

On 05/02/2016 05:08 PM, Dragomir Haralambiev wrote: I have registration activity with Radius asin So, why do you expect fragmentation from time to time as the OpenSIPS memory manager allocates and frees SHM blocks? -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street

Re: [OpenSIPS-Users] Memory fragments

On 05/02/2016 04:44 PM, Alex Balashov wrote: On 05/02/2016 04:22 PM, Dragomir Haralambiev wrote: Opensips has not routed any calls. Has it done anything else, including passive registration activity or any periodic database-bound synchronisation tasks? Also, what about passively deflecting

Re: [OpenSIPS-Users] Memory fragments

On 05/02/2016 04:22 PM, Dragomir Haralambiev wrote: Opensips has not routed any calls. Has it done anything else, including passive registration activity or any periodic database-bound synchronisation tasks? -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE

Re: [OpenSIPS-Users] Failure cause code in case of transaction timeout

cumstances of the timeout in a failure_route. -- 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.csrp

Re: [OpenSIPS-Users] Can I use fix_nated_sdp with domain as parâmeter?

NOT be included in an application-level       referral that might leave the scope). ‎ -- 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

Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

On 03/31/2016 02:49 PM, Travis Manson-Drake wrote: I would be more than happy to send it to you privately if that's ok? Of course! -- 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: [OpenSIPS-Users] What is the module to fix SDP ?

hardware as your SIP proxy, correct? -- 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: [OpenSIPS-Users] Asterisk Unrecognized sip header

otline: 520.545.0333 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel:

Re: [OpenSIPS-Users] New Accounting - PROBLEM

On 03/25/2016 03:48 PM, Dragomir Haralambiev wrote: unknown command , missing loadmodule? That sounds like a typo in the config. -- 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

Re: [OpenSIPS-Users] How can the global variable "listen" to listen WLAN ?

On 03/21/2016 04:36 PM, Rodrigo Pimenta Carvalho wrote: According to the documentation "...It can be an IP address, hostname or network interface id". So, can I do the following configuration? listen=tcp:wlan0:5060 Why can't you just do exactly that? -- Alex Balashov |

Re: [OpenSIPS-Users] Reqriteuri question

Travis, rewriteuri() is a legacy core function that does not support PVs. Have you considered ...? # Option 1. $rU = $fU; $rd = $avp(variable); # Option 2. $ru = "sip:" + $fU + "@" + $avp(variable); -- Alex -- Alex Balashov | Principal | Evariste System

Re: [OpenSIPS-Users] Out of memory problem

What is the exact text of the error? ‎ -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ Sent from my

Re: [OpenSIPS-Users] Out of memory problem

Have you attempted to increase the shared memory allocation to OpenSIPS (-m CLI option)? ‎ -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http

Re: [OpenSIPS-Users] Connection IP in SDP is messed up after RTPPRoxy Offer

Ali, Is there any danger that you are calling rtpproxy_offer() twice, or using rtpproxy_offer() in combination with fix_nated_sdp()[1]? -- Alex [1] http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899 -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter

Re: [OpenSIPS-Users] Things to take care of when dealing with preload route invite.

Hello Aqs, Why not strip the Route header instead of denying the request? That is to say: if(is_present_hf("Route")) remove_hf("Route"); -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United Sta

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Process:: ID=9 PID=743941 Type=SIP receiver udp:x.x.x.x:5060 Process:: ID=10 PID=743942 Type=SIP receiver udp:x.x.x.x:5060 Process:: ID=11 PID=743943 Type=SIP receiver udp:x.x.x.x:5060 -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

On 02/11/2016 08:50 PM, surya wrote: I am facing some performance issues that's why I was thinking about those. Performance issues of what nature? -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

I'm pretty sure that only the SIP receiver threads create DB connections. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http

Re: [OpenSIPS-Users] B2B bye method handling

I'm not a proper programmer to make a real patch. Thanks, Ross. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite

Re: [OpenSIPS-Users] recommended settings for performance

--Alex Balashov | Principal | Evariste Systems LLC303 Perimeter Center North, Suite 300Atlanta, GA 30346United StatesTel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)Web: http://www.evaristesys.com/, http

Re: [OpenSIPS-Users] ACK not forwarding

Bogdan, That's interesting. I had no idea OpenSIPS had a module specifically devoted to fixing bad SIP that really shouldn't be fixed. :-) -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920

Re: [OpenSIPS-Users] ACK not forwarding

On 02/04/2016 11:08 AM, Alex Balashov wrote: Bogdan, That's interesting. I had no idea OpenSIPS had a module specifically devoted to fixing bad SIP that really shouldn't be fixed. :-) s/a module/dialog module functionality/ -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter

Re: [OpenSIPS-Users] ACK not forwarding

no reasonable way to work around it: the UA in question needs to be fixed. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http

[OpenSIPS-Users] B2B Call-ID GUID generation scheme

d time unless overridden by method-specific behavior. All SIP UAs must have a means to guarantee that the Call- ID header fields they produce will not be inadvertently generated by any other UA. 2) Is there some way to override this that I don't realise, e.g. using a minimalistic scenario

Re: [OpenSIPS-Users] issues with prepared statements in mysql

http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ ___ Users mailing list Users@lists.opensips.org http

Re: [OpenSIPS-Users] opensips as a transcoder

OpenSIPS is a SIP proxy. It has no imaginable relation to codecs or anything audio-bearer related. It does not deal with RTP at all. -- Sent from mobile device On Feb 8, 2010, at 2:11 AM, Live School in...@live-school.net wrote: Hi, is it possible to use openSIPS as transcoder only ? if

Re: [OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge

://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ ___ Users mailing list Users@lists.opensips.org http

Re: [OpenSIPS-Users] Iñaki Baz Castillo all wisecrac k but no knowledge

. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com

Re: [OpenSIPS-Users] A thorough waste of my time fuck you Inaki good Bye

on your/development group beautiful solution. Thanks to Inaki I will be looking into something else to enable me to do what I want. FUCK YOU INAKI. great waste of my time. Now go fuck yourself. So long. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1

Re: [OpenSIPS-Users] Music On Hold Opensips

@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ ___ Users mailing list Users

Re: [OpenSIPS-Users] Is RPID being cached?

or a feature? In either case, is there a way to get OpenSIPs to only use the current DB value? Thanks! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste

Re: [OpenSIPS-Users] INVITE as a variable?

missing something? It would seem pretty straight forward but no way to do this, am I just blind? Thanks for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov

Re: [OpenSIPS-Users] Need help to forward Access number

Rewrite Request URI ($ru) and t_relay(). -- Sent from mobile device On Jan 1, 2010, at 1:40 AM, Ahmed Munir ahmedmunir...@gmail.com wrote: Hi, I want to forward an Access Number from OpenSIPS to Asterisk machine. Kindly advise how can i do that? Which modules/functions are use to

Re: [OpenSIPS-Users] re-REGISTER with To tag gets 404

be added for REGISTERs? Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678

Re: [OpenSIPS-Users] Any word on the B2B bug?

/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin

Re: [OpenSIPS-Users] Any word on the B2B bug?

. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] loose_route: loop on ACK requests

); } ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel

Re: [OpenSIPS-Users] Transfer issue

@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users

Re: [OpenSIPS-Users] Transfer issue

SEMS is another option for a B2BUA, though all its prebuilt B2BUA apps are not pure B2BUA but incorporate some sort of initial media-based announcement premise. But you can use its Python or C++ API to create one. Another option is YATE? -- Sent from mobile device On Oct 23, 2009, at

Re: [OpenSIPS-Users] send bye in case of parallel forked call

No. -- Sent from mobile device On Oct 21, 2009, at 9:34 AM, Uwe Kastens ki...@kiste.org wrote: Hi, I have the following requirement: If a from tm generated cancel is answered with a 200 OK I want to send a BYE to the UAC. Is this possible? BR Uwe -- kiste lat: 54.322684,

Re: [OpenSIPS-Users] [OpenSIPS-News] How modify Contact header

THIS E-MAIL ___ News mailing list n...@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/news -- Alex Balashov - Principal Evariste

Re: [OpenSIPS-Users] Using RTPProxy to proxy all non-NAT'ed calls as well

Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list

Re: [OpenSIPS-Users] how do you determine first registration of a device

://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http

Re: [OpenSIPS-Users] how do you determine first registration of a device

there? maybe the callid or cseq or something? On Sat, Sep 26, 2009 at 1:43 PM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: As Inaki said, what you're trying to achieve is a little ridiculous. But, if you must do it, I recommend using a database capable

Re: [OpenSIPS-Users] how do you determine first registration of a device

Alex Balashov wrote: A new REGISTER request will certainly have a different Call-ID. However, I think the correct way to handle this is to just live with it. If the phone did not unregister, then it will be subject to the consequences of that, and it's okay. Kamailio 1.5.x's registrar

Re: [OpenSIPS-Users] Can i use openser to insert Service Route header in 2xx Register response.

mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users

Re: [OpenSIPS-Users] Can i use openser to insert Service Route header in 2xx Register response.

All changes are made in openser.cfg. priyank luthra wrote: Hi Can you tell me in detail, which file to make this change in.. i am a newbie to opensips I have openSER 1.3.0. On Thu, Sep 17, 2009 at 12:03 PM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com

Re: [OpenSIPS-Users] Can i use openser to insert Service Route header in 2xx Register response.

Iñaki Baz Castillo wrote: 2009/9/17 Alex Balashov abalas...@evaristesys.com: insert_hf() But that is just for request, not for responses... There is some function in tm module to add headers in responses (AFAIR). That's not true, AFAIK: http://www.opensips.org/html/docs/modules/1.5.x

Re: [OpenSIPS-Users] RTP traffic opensips

-- Personal webpage - www.jonbaraq.eu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov

Re: [OpenSIPS-Users] explizit handling auf replyto

://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http

Re: [OpenSIPS-Users] explizit handling auf replyto

Uwe Kastens wrote: Replies are automatically routed only if they are statefully routed. Statefull = t_relay() ? Yes. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

Re: [OpenSIPS-Users] explizit handling auf replyto

layer used to open the transaction from the initial request will not know that you did so, and vice versa, etc etc. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

  1   2   >