Re: [OpenSIPS-Users] Call fetch_dlg_value inside a timer route

[foo] >{ get_profile_size(); } > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 5/18/23 3:13 PM, Daniel Zanutti wrote: > > Hi Alberto > > In fact, i need to u

Re: [OpenSIPS-Users] Access "Identity" header after calling stir_shaken_auth() on 3.2.x

com > https://www.siphub.com > > On 5/15/23 12:08 AM, Daniel Zanutti wrote: > > Hi Brett > > Just to respond, no it doesn't. This field is only visible when we are > authenticating headers, not generating. > > At the end, I copied the module from the 3.3.x version, to

Re: [OpenSIPS-Users] Registration passthrough

; > Sent with Proton Mail <https://proton.me/> secure email. > > --- Original Message --- > On Thursday, June 1st, 2023 at 9:31 PM, Daniel Zanutti < > daniel.zanu...@gmail.com> wrote: > > Check if you are manipulating contact with some function like > fix_contact() o

Re: [OpenSIPS-Users] Registration passthrough

> uac > i will try with stateless as i just want to forward it via opensips and > asterisk to not know opensips > > > Sent with Proton Mail <https://proton.me/> secure email. > > --- Original Message --- > On Thursday, June 1st, 2023 at 1:59 PM, Daniel Zanutti < > d

Re: [OpenSIPS-Users] Registration passthrough

Hi By standard, opensips does not change the Contact and your asterisk box should receive the original Contact, sent by UAC. Are you sure the contact is being changed by Opensips? I saw asterisk ignoring the contact and putting source IP and origin some times. Long time I don't work with

Re: [OpenSIPS-Users] Call fetch_dlg_value inside a timer route

ou can use dialog variables, you have to load > the dialog context by using func_load_dialog_ctx. Maybe it's the same with > timer routes. > > https://opensips.org/docs/modules/3.2.x/dialog#func_load_dialog_ctx > > On Wed, 17 May 2023, 21:07 Daniel Zanutti, > wro

[OpenSIPS-Users] Call fetch_dlg_value inside a timer route

Hi folks Why is it not possible to call *fetch_dlg_value *inside a timer route? Is there any other alternative to it? I wanted to generate some statistics every X seconds. Thanks ___ Users mailing list Users@lists.opensips.org

Re: [OpenSIPS-Users] X-header extraction

Just get the variable $hdr(X-notice) On Wed, May 17, 2023 at 2:25 PM nutxase via Users wrote: > Hi All, > > What is the best way to extract a custom X header from a sip message and > log it as a variable > example i receive X-notice:200 and i want to create a variable that with > 200 > > is

Re: [OpenSIPS-Users] Access "Identity" header after calling stir_shaken_auth() on 3.2.x

llo Daniel, > See if the $identity peudovariable offered by that module suits your need: > > https://opensips.org/docs/modules/3.2.x/stir_shaken.html#pv_identity > > -Brett > > > On Fri, Apr 28, 2023 at 9:03 AM Daniel Zanutti > wrote: > >> Hi >> >> Ho

[OpenSIPS-Users] Access "Identity" header after calling stir_shaken_auth() on 3.2.x

Hi How can I access the generated Identity header, after calling function stir_shaken_auth(), on opensips 3.2.x? On 3.3.x there is a new "out" parameter, is there a way on 3.2.x? It's just to store on DB. Thanks ___ Users mailing list

Re: [OpenSIPS-Users] ACC module for rejected calls

Hi Alberto You are correct, this is the line you need. I think you need a created transaction. Since you are responding in stateless, you may be missing the cdr. Try changing this and let me know if solves: sl_send_reply(488, "Not Acceptable Here"); -> t_reply(488, "Not Acceptable Here");

Re: [OpenSIPS-Users] SIP Call ID duplicated on 3.1.14

Yes, since 1.x it's there. On Sat, Mar 25, 2023 at 11:38 AM Saint Michael wrote: > I use 3.1, is that applicable? > > > On Sat, Mar 25, 2023 at 9:49 AM Daniel Zanutti > wrote: > >> Hi Federico >> >> Yes it does, need to create the transaction inside your

Re: [OpenSIPS-Users] SIP Call ID duplicated on 3.1.14

Hi Federico Yes it does, need to create the transaction inside your script: https://opensips.org/html/docs/modules/3.2.x/tm.html#func_t_newtran This will avoid opensips handling the duplicated invite as a new call. On Fri, Mar 24, 2023 at 11:40 PM Saint Michael wrote: > I have on a typical

Re: [OpenSIPS-Users] Check status of routes loading - drouting

Exactly! Thanks On Fri, Mar 24, 2023 at 11:25 AM Callum Guy wrote: > Hi Daniel, > > I believe you're looking for this feature as included since 3.3 > > https://www.opensips.org/Documentation/Interface-StatusReport-3-3 > > Enjoy, > > Callum > > On Fri, 24

[OpenSIPS-Users] Check status of routes loading - drouting

Hi Is there a way to check the status of initial loading of routes, on the drouting module? If routes are being loaded after a cold start, I want to do some alternate routing. Thanks ___ Users mailing list Users@lists.opensips.org

Re: [OpenSIPS-Users] Carrier_ID not writing on DRouting when using partitions

-3-2#GEN-DB-DR-PARTITIONS > > > > > > Ben Newlin > > > > *From: *Users on behalf of Daniel > Zanutti > *Date: *Tuesday, February 21, 2023 at 9:47 PM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Carrier_ID not writing on DRouting wh

[OpenSIPS-Users] Carrier_ID not writing on DRouting when using partitions

Hey I'm having a weird issue, possibly a BUG, using opensips 3.2.8. The carrier_id_avp is not being written, when I enabled partitions or drouting module. Everything else works, just this value is not written to the AVP setted. Routing works fine using carriers, just the AVP is not written.

Re: [OpenSIPS-Users] Version 3.1 it stops writing cdrs

Virtual Router Redundancy Protocol (VRRP) https://www.techopedia.com/definition/13483/virtual-router-redundancy-protocol-vrrp On Wed, Feb 15, 2023 at 3:21 PM Saint Michael wrote: > > what is VRRP ? > > On Wed, Feb 15, 2023 at 1:16 PM Kingsley Tart wrote: >> >> FWIW, I set up OpenSIPS here in

Re: [OpenSIPS-Users] kamilio's htable equivalent?

Hey David Did you take a look at core functions of cache? -> https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc4 On Wed, Dec 21, 2022 at 9:14 AM David Villasmil < david.villasmil.w...@gmail.com> wrote: > Hello folks, > > I'm trying to find in opensips an equivalent to

Re: [OpenSIPS-Users] - Not sending ACK back!

istrar", "max_contacts", 10) > > modparam("registrar", "received_avp", "$avp(rcv)") > > modparam("registrar", "retry_after", 30) > > > Regarding option (4) - I have both options. IP to IP and User/Pass >

Re: [OpenSIPS-Users] - Not sending ACK back!

Hi Nitesh As you already know, opensips is a low level software. You have to understand several aspects of SIP, network, RTP, DNS that when you use Asterisk, most you don't need to understand deep. Trying to help you, your script is way simple for you achievements. You need: 1) Check NAT on all

Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

No, works same way. Just look at docs of 3.1 On Mon, Sep 26, 2022 at 11:58 AM Saint Michael wrote: > I use opensips 3.1, does it matter? > > > On Mon, Sep 26, 2022 at 10:20 AM Daniel Zanutti > wrote: > >> can you write your own functions with opensips? >> Yes

Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

> Dear Daniel >> Can you point me to an example? >> Right now Opensios will get a clogged memory. >> Many thanks. >> >> >> On Sun, Sep 25, 2022, 11:45 AM Daniel Zanutti >> wrote: >> >>> You have to use dialog variable storing.

Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

You have to use dialog variable storing. Take a look at dialog module. Em dom., 25 de set. de 2022 10:42, Saint Michael escreveu: > I noticed that the variable > $avp(lineid) > set in the section of the code handling the original INVITE, is null when > I need to close the call. > Is there a way

Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

udp:127.0.0.1:7891 -F -L 10240 -m 15000 -M 2 -T 20 -d > WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889 > ExecStop=/usr/bin/pkill -F /var/run/rtpproxy2.pid > > > StandardOutput=syslog > StandardError=syslog > SyslogIdentifier=rtpproxy2 > SyslogFacility=local5 > > TimeoutS

Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

t; *Verzonden:* Wednesday, September 14, 2022 9:56:41 PM > *Aan:* OpenSIPS users mailling list > *Onderwerp:* Re: [OpenSIPS-Users] The update from yesterday makes all > calls fail after 20 seconds, how do I go back? > > how do I do this: > " put some log on local_route" >

Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

So your Opensips is hanging up the call. Do you see any log on it? Try put some log on local_route if you don't see anything. On Wed, Sep 14, 2022 at 4:40 PM Saint Michael wrote: > This is a trace showing a BYE from Opensips, but none of the sides did > actually hangup. > > > On Wed, Sep 14,

Re: [OpenSIPS-Users] Question about error 500 only via WIFI

> Any hint will be very helpful ! > > Thanks alot. > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL: 979 > > -- > *De:* Users em nome de Daniel Zanutti < > daniel.zanu...@gmail.com> &g

Re: [OpenSIPS-Users] Question about error 500 only via WIFI

Olá Rodrigo, tudo bem? Saudações de São Paulo! Opensips doesn't differentiate the network, it will look just to the sip packet. I recommend you sniff through your packets and check what's different. Probably there's somenthing on opensips log you didn't get yet, recommend you take a look there

Re: [OpenSIPS-Users] Topology Hiding

https://www.opensips.org/Documentation/Tutorials-Topology-Hiding On Tue, May 10, 2022 at 2:15 PM Saint Michael wrote: > Dear friends > I am using opensips 3.1.9, with rtp proxy, and without topology hiding it > would not talk to any carrier who has a Sonus box. I need to add topology > hiding

Re: [OpenSIPS-Users] incompatibility leads to massive CDR loss

I think this is your problem: branch=z9hG4bK-524287-1---b8aced18b4075aa3 *=49972* You have char "=" inside a string, which is a reserved character and not allowed on a string: https://datatracker.ietf.org/doc/html/rfc3261#section-25.1 Should be something on client of your customer, since you

Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission

, May 4, 2022 at 2:19 PM Yannick LE COENT wrote: > Hi Daniel, > > I do not think the ACK is sent by my script. It is sent by the TM module > since it is a negative response. > Am I wrong ? > > Thanks, > Yannick > > Le 04/05/2022 à 18:48, Daniel Zanutti a écrit : > &g

Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission

||--->| >|407 || >| X<-|| >| (no retrans.) || > > When the 407 is lost between OpenSIPS and Alice, it is not retransmitted > by OpenSIPS. > > I would like to force retransmission.

Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission

Generate in Stateful -> www_challenge or proxy_challenge? https://opensips.org/html/docs/modules/3.2.x/auth.html Is this what you are looking for? On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT wrote: > Hello all, > > Could you tell if there is a way to enable 407 in stateful mode ? > >

Re: [OpenSIPS-Users] strange INVITE transmission

nsips/nat-contact-and-via-fixing-in-sip-part-3/ > article but I have the same problem - no response for REGISTERs. > > Is there any way to know why opensips ignores or does not respond for > REGISTERs? > Please find my new opensips.cfg that Diniel's advice is applied. > > > Thank

Re: [OpenSIPS-Users] strange INVITE transmission

Hi Kiwon You need to handle NAT scenarios. Try putting this code on line 254, right after "t_check_trans()": if (nat_uac_test("7")) { #nathelper if(is_method("REGISTER")) fix_nated_register(); else fix_nated_contact(); xlog("L_NOTICE", "Fix contact - M=$rm RURI=$ru F=$fu T=$tu

Re: [OpenSIPS-Users] SIP Reg with radius

Take a look here: https://www.opensips.org/Documentation/Tutorials-Radius On Sat, Feb 12, 2022 at 1:16 PM Vishal Pai wrote: > Hello Team > > I am new to Opensips. Can we have the sip registration to lookup for auth > in Radius if yes then we can forward the sip invite to PBX with a unique >

Re: [OpenSIPS-Users] BYE from UAC bypasses OpenSIPS

John I highly recommend using the topology hiding module instead of inserting routes and forwarding the SIP message. Several IP devices have problems when you have a lot of routes. Even the SIP message size can be a problem if your call flows through several proxies. When you use topology

Re: [OpenSIPS-Users] learning the realm from authentication challenges

Don't forget to deal with CSEQ increment on the authenticated INVITE. Also we had problems when any in-dialog message is received, we have to deal with CSEQ on all of them. =( On Fri, Sep 25, 2020 at 12:30 PM johan wrote: > Jeff, be warned that the datafill for registrar is not obvious. > On

Re: [OpenSIPS-Users] Maybe it's a bug

Hi folks We implemented millisecond billing in our platform, so no need to round on the Opensips layer, the rounding is done in our business billing layer. This way customers can have a different rounding than VoIP providers. It's not a way to penalize customers, but some providers just work

Re: [OpenSIPS-Users] Opensips + rtpproxy issue

by enough max open do files? I do no linit or set >anything >- I traced with tshark and i can see issue with A and B leg > > > Thank you for help! > Br > Miha > > Miha > On 5 May 2020, 16:07 +0200, Daniel Zanutti , > wrote: > > No special configur

Re: [OpenSIPS-Users] Opensips + rtpproxy issue

big distortion it is impossibly to > comunicate with each other. > > We have two cors deticated to it. Do you have any special > thing set on it? > > tnx > miha > > On Tue, 5 May 2020 10:27:22 -0300 > Daniel Zanutti wrote: > > Hi Miha > > > > Could you

Re: [OpenSIPS-Users] Opensips + rtpproxy issue

Hi Miha Could you explaining how does it break? We use it in virtual machines and our safe limit is around 500 simultaneous calls, on dedicated single core VPS. Does CPU usage reach 100%? On Tue, May 5, 2020 at 10:11 AM Miha via Users wrote: > Hello > > we have virtualized opensips and

Re: [OpenSIPS-Users] How to check the transport of the RTP sessions?

He didn't said SDP, he said RTP Sessions. Opensips cannot inspect rtp sessions. On Fri, Jan 31, 2020 at 11:09 AM David Villasmil < david.villasmil.w...@gmail.com> wrote: > You can also use the textops’ search function. > > > On Fri, 31 Jan 2020 at 13:43, Daniel Zanutti

Re: [OpenSIPS-Users] How to check the transport of the RTP sessions?

Hi Are you using just Opensips or some RTP proxy solution? If you are using just Opensips, the RTP traffic will be Peer-to-peer and you have to monitore origin ou destination. If you are using some RTP proxy solution, just check on this machine. Regards On Fri, Jan 31, 2020 at 7:33 AM Abdoul

Re: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers?

Hi Diptesh We tried to implement a native prepaid system on Opensips but didn't found a way to do this natively, so we developed a custom prepaid mechanism to our solution. Our company (http://dazsoft.com) is focused on complete systems but we can negotiate this specific part if you want. Let

Re: [OpenSIPS-Users] SIP Proxy and Paid Support

Hi Rick I have a lot of experience on Opensips, maybe I can take a look at your project. Let me know if interested. Thanks On Mon, Sep 24, 2018 at 1:06 AM Alexander Jankowsky wrote: > > > Hello Rick, > > > > There are some books around with the fundamentals so you can experiment > and learn

Re: [OpenSIPS-Users] Doubts on call-center scenario

nder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 09/20/2018 09:32 PM, Daniel Zanutti wrote: > > Hi Bogdan > > I'm triggering the script via MI. The idea is to send some parame

Re: [OpenSIPS-Users] Doubts on call-center scenario

PS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 09/12/2018 12:32 AM, Daniel Zanutti wrote: > > Hi everyone, > > I'm using opensips to originate a call to 2 destinations

[OpenSIPS-Users] Doubts on call-center scenario

Hi everyone, I'm using opensips to originate a call to 2 destinations then bridge then, using B2B scenario. How to send some custom parameters to help accounting? I need to identify that this specific call, is related to some customer. Didn't find in docs a proper way to do it, so my idea is to

Re: [OpenSIPS-Users] Doubt about call center module

e > call queuing. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 08/31/2018 04:06 PM, Daniel Zanutti wrote: > >

Re: [OpenSIPS-Users] Doubt about call center module

er > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 08/30/2018 08:34 PM, Daniel Zanutti wrote: > > Hi Bogdan > > Yes, It's the same scenario and same message. The call flow is: > > Asterisk Dials(

Re: [OpenSIPS-Users] Doubt about call center module

e SIP URI is not valid. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 08/29/2018 10:26 PM, Daniel Zan

Re: [OpenSIPS-Users] Doubt about call center module

90]: Falha entrando na fila - erronum: -1 On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti wrote: > Trying to configure the call center modules, but found a problem when > there is no agents available. > > If there is 1 agent available, call is sent to him with no problem: > > Au

[OpenSIPS-Users] Doubt about call center module

Trying to configure the call center modules, but found a problem when there is no agents available. If there is 1 agent available, call is sent to him with no problem: Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk - Tentando entrar na fila fila-1 Aug 27 18:11:00 plat5

Re: [OpenSIPS-Users] Use RTPPROXY to bridge ipv4/ipv6

note "/" in front of IPv6 addr): > > /bin/rtpproxy -F -l "200.200.200.200" -6 "/2607:3f00:2 > <http://200.200.200.200/2607:3f00:2>" > > -Max > > On Thu, Aug 2, 2018 at 1:50 PM Daniel Zanutti > wrote: > >> Hi >>

[OpenSIPS-Users] Use RTPPROXY to bridge ipv4/ipv6

Hi I'm trying to configure RTPPROXY to bridge ipv4 and ipv6 networks, but didn't find the proper way. Supposing IPs "200.200.200.200" and "2607:3f00:2 " both on ETH0 interface. Tried: /bin/rtpproxy -F -l 200.200.200.200/2607:3f00:2 Got this error: Restarting rtpproxy: rtpproxy:

Re: [OpenSIPS-Users] Using Sipp stress tool with Opensips

I think the problem is related to configuring SIPP properly. If I'm not wrong, SIPP standard scenario for UAC/UAS is configured to work with a gateway (B2B), but Opensips is a proxy. You have to use Routes to properly handle the incoming call and respond it. Take a lookt at "rrs" param of recv

Re: [OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK

ACK the received 200 OK (even if > CANCEL was sent) and if it really wants to terminate the call, it has to > fire a BYE. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www

Re: [OpenSIPS-Users] Wrong number of children

Hi You should have 5 per interface + some internal control threads. I'm not sure exactly. Regards On Fri, Feb 23, 2018 at 9:25 AM, xaled wrote: > Hi > > > > I have configured 5 children in opensips.cfg, but 16 get logged and 18 > initiated what could be the cause of it? > > I

Re: [OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK

/www.opensips.org/events/Summit-2018Amsterdam > > On 02/20/2018 06:43 PM, Daniel Zanutti wrote: > > Hey > > I had a problem when receiving simultaneous CANCEL from customer and 200 > OK from gateway. > > Seems that the first CANCEL was rejected, but the second CANCEL was

[OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK

Hey I had a problem when receiving simultaneous CANCEL from customer and 200 OK from gateway. Seems that the first CANCEL was rejected, but the second CANCEL was accepted. This second CANCEL did NOT go to the gateway, just Opensips received and replied with 200 OK. This is the log of the first

[OpenSIPS-Users] Solution to storing a lot of siptrace

Hi I have around 2000 simultaneous calls, 50 CPS and would like to store sip trace for all of them. Storing on MySQL is not working. If you have some indexes on the table, after 1M register it starts to slow down the whole server. If no indexes, it's not searchable. Do you guys have a good

Re: [OpenSIPS-Users] Weird freeze/crash

ks like a crash. > What version of OpenSIPS are you using? If you could extract a coredump, > it would be really helpful. > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > On 09/19/2017 04:07 PM, Daniel Zanutti wrote

[OpenSIPS-Users] Weird freeze/crash

Hi Do you guys have an idea of what happened? Sep 10 09:35:13 /sbin/opensips[13579]: NOTICE:core:io_wait_loop_epoll: EPOLLIN(read) event: epoll_wait() set event EPOLLHUP - connection closed by the remote peer! Sep 10 09:35:13 /sbin/opensips[13579]: CRITICAL:core:receive_fd: EOF on 42 Sep 10

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

http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > On 07/20/2017 10:49 PM, Daniel Zanutti wrote: > > Hi Alex > > I'm having a billing problem from receiving BYE to 200 OK is taking more &

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

Hi Alex I'm having a billing problem from receiving BYE to 200 OK is taking more than 500ms. If BYE is accounted when it's received, great! Are you absolutely sure it works this way? Thanks On Thu, Jul 20, 2017 at 4:26 PM, Alex Balashov wrote: > My understanding is

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

Yes, ACC module. On Thu, Jul 20, 2017 at 3:45 PM, Alex Balashov <abalas...@evaristesys.com> wrote: > On Thu, Jul 20, 2017 at 03:40:29PM -0300, Daniel Zanutti wrote: > > > In what exactly moment the 200OK and BYE messages are accounted and > written > > to the data

[OpenSIPS-Users] Accounting of 200 OK and BYE

In what exactly moment the 200OK and BYE messages are accounted and written to the database? At the moment Opensips receive the 200 OK or after receive ACK of 200 OK? Also on BYE, when receive BYE or on 200 OK of BYE? Thanks ___ Users mailing list

[OpenSIPS-Users] Looking for Mediaproxy developer

I'm looking for help to customize some things on mediaproxy software. Need to: 1) Fix some bugs 2) Implement new features Please contact me for details. Thanks ___ Users mailing list Users@lists.opensips.org

[OpenSIPS-Users] RTPENGINE (sipwise) working with opensips?

Question: Is RTPENGINE (sipwise) working with opensips? I saw only Kamailio on rtpengine page at github, but there is a module for opensips 2.1 ( http://www.opensips.org/html/docs/modules/2.1.x/rtpengine). Thanks ___ Users mailing list

Re: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

money. You can even use free certificates from LetsEncrypt.org > > > On Jun 7, 2017, at 11:14 AM, Daniel Zanutti <daniel.zanu...@gmail.com> > wrote: > > Hi Alex > > I have tried with self-generated certificate and it is working fine. > > The problem is that this

Re: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

sorted out and working properly with your own certificates. > > > > Do you really just only need a certificate to get things running > > for now or does it have to be from a recognised authority. > > > > Alex > > > > *From:* Users [mailto:users-boun...@lists.open

[OpenSIPS-Users] Which SSL Certificate for SIP+TLS

I need to install an TLS certificate for secure SIP communication. Could you guys please point a valid certificate so I can buy it? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips Late reply.

Hi Qasim How did you limit CPS? Because i have a similar scenario (150cps) but i set children to 20 or 24, never 200. You don't need 1 children per request. On Wed, Apr 5, 2017 at 9:44 AM, qasimak...@gmail.com wrote: > Hi, > > I have this scenario where i originate calls

Re: [OpenSIPS-Users] Mediaproxy speed calculations

nds. > You can disable them by setting the sampling interval to 0. The warning > doesn't mean they are skipped, it only means the relay took too long to > compute them and was unresponsive for other requests during that time. > > > > > Thanks > > > > > > On Tue

Re: [OpenSIPS-Users] Mediaproxy speed calculations

Tue, Mar 28, 2017 at 2:27 PM, Dan Pascu <d...@ag-projects.com> wrote: > > On 24 Mar 2017, at 19:51, Daniel Zanutti wrote: > > > Hi > > > > Looks like i'm diving deep on mediaproxy. > > > > Some of our relays are not calculating the speed on the networ

Re: [OpenSIPS-Users] ACC Db Duration

Hi Did you check "cdr_flag" on Acc module? Otherwise it will generate 2 registers, 1 for INVITE and 1 for BYE. Regards On Tue, Mar 28, 2017 at 10:29 AM, qasimak...@gmail.com wrote: > Hi, > > Sorry for the spam last email i miss-clicked on send amidst writing the >

Re: [OpenSIPS-Users] ACC db duration

Hi Did you check "cdr_flag" on Acc module? Otherwise it will generate 2 registers, 1 for INVITE and 1 for BYE. Regards On Tue, Mar 28, 2017 at 10:09 AM, qasimak...@gmail.com wrote: > Hi, > > I have enabled acc module in my opensips installation with db, My CDR's > are

[OpenSIPS-Users] Mediaproxy speed calculations

Hi Looks like i'm diving deep on mediaproxy. Some of our relays are not calculating the speed on the network. If I restart a couple times it starts calculating fine. I found this log: media-relay[4100]: warning: Aggregate speed calculation time exceeded 10ms: 11644 us for 222 sessions Is there

[OpenSIPS-Users] Which internal module did the hangup?

I have 2 modules that may hangup the call: - Dialog - duration timeout or sip ping with sip options - Mediaproxy - RTP timeout On local_route, is there any way to know which module did the hangup? Thanks ___ Users mailing list

Re: [OpenSIPS-Users] Monitoring Mediaproxy

; } > } > if($num_relays==$argv[1]) > { > echo "OK IPs de Relays RTP: ".$str_salida."\n"; > exit(0); > } > else > { > echo "ERROR faltan Relays. IPs de Relays RTP: ".$str_salida."\n"; > exit(

Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

, you guys are great. On Fri, Mar 17, 2017 at 3:08 PM, Dan Pascu <d...@ag-projects.com> wrote: > > On 17 Mar 2017, at 3:54, Daniel Zanutti wrote: > > > Adrian > > > > You may be correct, overload can be the problem. But since the call is > already finished

Re: [OpenSIPS-Users] Monitoring Mediaproxy

Understood. Thanks for explanation. Regards On Fri, Mar 17, 2017 at 2:47 PM, Dan Pascu <d...@ag-projects.com> wrote: > > On 16 Mar 2017, at 15:58, Daniel Zanutti wrote: > > > Hi Dan > > > > This is exactly how I'm monitoring but looking to the dispatcher

Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

wrote: > Perhaps your virtual machine simply cannot handle the load. The commands > to close sessions may also be dropped or lost under such environment. > > Adrian > > > > On 16 Mar 2017, at 11:22, Daniel Zanutti <daniel.zanu...@gmail.com> wrote: > > Hi Dan >

Re: [OpenSIPS-Users] Monitoring Mediaproxy

that can send one and then based on that disable/enable > accordingly. Hopefully mediaproxy will not respond to the “SIP ping” if > frozen. > > > > Robert > > > > From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of > Daniel Zanutti > > Sent: Wednesday,

Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

(or the other endpoint's data gets filtered at some > firewall), and because it cannot learn both endpoint's addresses it cannot > setup the kernel conntrack rule to move data forwarding to the kernel. > > On 14 Mar 2017, at 13:38, Dan Pascu wrote: > > > > > On 13 Mar 2017

Re: [OpenSIPS-Users] Monitoring Mediaproxy

and then based on that disable/enable > accordingly. Hopefully mediaproxy will not respond to the “SIP ping” if > frozen. > > > > Robert > > > > From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of > Daniel Zanutti > > Sent: Wednesday, March 15, 2017 4:55 PM

Re: [OpenSIPS-Users] Monitoring Mediaproxy

How can this be done? Or do you mean SIP options? On Wed, Mar 15, 2017 at 5:45 PM, Johan De Clercq <jo...@democon.be> wrote: > Send options. > > On 15 Mar 2017 11:48 PM, "Daniel Zanutti" <daniel.zanu...@gmail.com> > wrote: > >> Hi >> >>

[OpenSIPS-Users] Monitoring Mediaproxy

Hi What's the best way to check if a mediaproxy is running fine? Monit is monitoring PID but how can I check the process has is not frozen? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

Hi guys I sent this email a few days ago, anyone from Mediaproxy team could take a look? I could debug it, just need some directions on where to look. Thanks On Tue, Mar 7, 2017 at 11:10 AM, Daniel Zanutti <daniel.zanu...@gmail.com> wrote: > I'm using mediaproxy on several inst

Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

Any idea guys? On Tue, Mar 7, 2017 at 11:10 AM, Daniel Zanutti <daniel.zanu...@gmail.com> wrote: > I'm using mediaproxy on several installations and have noticed that when > the machine is on high load (> 700 sessions), the media-relay process > starts to hang some sessions.

[OpenSIPS-Users] Mediaproxy hanging sessions on high load

I'm using mediaproxy on several installations and have noticed that when the machine is on high load (> 700 sessions), the media-relay process starts to hang some sessions. These sessions doesn't have any RTP being sent/received anymore and they never hangup. After some hours of frozen sessions,

Re: [OpenSIPS-Users] OpenSIPS Features

I think you have a wrong answer: 2) Yes, you can. But the SIP Signaling part is only half part of the problem. Checkout this sip client: http://icanblink.com/ It implements desktop sharing over SIP and yes, they use Opensips. Regards On Sat, Feb 25, 2017 at 2:04 PM, Voice TAC

Re: [OpenSIPS-Users] Change 200 OK of BYE message

ou sure you arm the onreply > route when handling the _BYE_ request ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 02/07/2017 01:48 PM, Daniel Zanutti wrote: > > The sip client is working fine, I can confirm t

Re: [OpenSIPS-Users] Change 200 OK of BYE message

there should be no > problem. It can of course also be that the client does something strange > ... > > 2017-02-06 21:43 GMT+01:00 Daniel Zanutti <daniel.zanu...@gmail.com>: > >> Hi Robert >> >> Yes, all messages are passing through the proxy, but when I receive the >&g

Re: [OpenSIPS-Users] Change 200 OK of BYE message

kow...@ets.org> wrote: > Did you use “record_route”? > > > > For reference: > > http://www.iptel.org/sip/intro/scenarios/rr > > > > > > Robert Mundkowsky > > > > *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of

[OpenSIPS-Users] Change 200 OK of BYE message

Hi I need to change something on the 200 OK of BYE message. Tried everything on Opensips but looks like this message doesn't follow standard message path. Neither Main Route or Reply route pass this message. Is there any way to do it? Thanks ___ Users

Re: [OpenSIPS-Users] How to Keep OpenSIPS in reverse path

I also suggest use topology_hiding(), i saw some equipments ignoring Via/record-route order. On Wed, Dec 14, 2016 at 3:53 PM, Muhammad Naseer Bhatti wrote: > > Hi Razvan, > I am not using REGISTER, but I guess add_path() wont’ work for me, I am > using record_route() for the

[OpenSIPS-Users] Github x gzip version

Hi Just a question about which version to use: Is it safe to use the latest Github version of 1.11.x or is safe to use the .tar.gz version? My point is: Can I trust github version to use in production? Thanks ___ Users mailing list

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