[OpenSIPS-Users] Write siptrace on "local" route

I'm trying to capture siptrace on local route like this: local_route { sip_trace(); } So when dialog module hangups a call, I'll get the siptrace of the BYE. This is not working. Is there any workaround? I'm using Opensips 1.11.6. Thanks ___

[OpenSIPS-Users] Problem running Media Proxy on updated Debian Jessie

Just reporting, I found a problem when running Media Proxy module with updated Debian Jessie distro. There is some error at Python... This is the error at syslog: Mar 9 16:37:26 mp5 media-relay[2252]: error: Unhandled error in Deferred: Mar 9 16:37:26 mp5 media-relay[2252]: AttributeError:

Re: [OpenSIPS-Users] Getting AVP or Transaction data on local_route

other way, you can get above done by storing avp value along with call > id and fetching desired value in local route. > On 17-Mar-2016 8:04 pm, "Daniel Zanutti" <daniel.zanu...@gmail.com> wrote: > >> I'm trying to get an AVP variable at local_route, but all AVPs

Re: [OpenSIPS-Users] Opensips doesn't react to 200 on BYE

Hi Daniel SIPP is a litte tricky to work with proxies, you need to double check routes and Via headers. Here is what I'm using on BYE: BYE [next_url] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] [routes] From:

Re: [OpenSIPS-Users] Opensips doesn't react to 200 on BYE

w on? > > > Thanks again. > > > Atenciosamente, > > Daniel Moreira Yokoyama. > @dmyoko > http://twitter.com/dmyoko > > TrafficTalks > Um podcast sobre cinema feito a partir de conversas de trânsito. > http://traffictalks.com.br > > > 2016-03-17 10:55 GMT

[OpenSIPS-Users] Getting AVP or Transaction data on local_route

I'm trying to get an AVP variable at local_route, but all AVPs I have are nulled. When dialog module hangs the call (due to timeout), I configured to send BYE to both sides. I need that the first BYE be accounted and second not... So i did set an AVP and tried to get on the second call of

Re: [OpenSIPS-Users] Query about avp in siptrace module

Still am seeing AVP is NULL in my traces. Can you please clarify me on > what am missing out. > > > > Regards, > Agalya > > > > > > *From:* users-boun...@lists.opensips.org [mailto: > users-boun...@lists.opensips.org] *On Behalf Of *Daniel Zanutti > *Sent:* Monday, M

Re: [OpenSIPS-Users] Query about avp in siptrace module

ptrace module? It would > be great if you help me out on this. > > > > Regards, > > Agalya > > > > *From:* users-boun...@lists.opensips.org [mailto: > users-boun...@lists.opensips.org] *On Behalf Of *Daniel Zanutti > *Sent:* Tuesday, March 22, 2016 12:22 PM > >

Re: [OpenSIPS-Users] Query about avp in siptrace module

Correcting: *$avp(traced_user) = "1";* On Mon, Mar 21, 2016 at 2:19 PM, Daniel Zanutti <daniel.zanu...@gmail.com> wrote: > Hi > > AVP is a kind of variable on Opensips. > > When you configured the siptrace module to use the AVP, you must ensure > that the

Re: [OpenSIPS-Users] Query about avp in siptrace module

Hi AVP is a kind of variable on Opensips. When you configured the siptrace module to use the AVP, you must ensure that the AVP has some kind of value before routing the package, otherwise it will be NULL. Just add this on first line of main route: *$avp(traced_user) = 1;* On Mon, Mar 21,

Re: [OpenSIPS-Users] Query about avp in siptrace module

> Just adding the line *$avp(traced_user) = "1";* in config file will > solve the issue? Or we need to include > > *modparam("siptrace", "traced_user_avp", "$avp(traced_user)") also in the > config file?* > > Regards, > Agalya > > *From:

Re: [OpenSIPS-Users] Query about avp in siptrace module

abase", 0) > > modparam("siptrace", "trace_flag", 22) > > modparam("siptrace", "trace_on", 1) > > modparam("siptrace", "hep_version", 2) > > > > route{ > > > > $avp(tra

Re: [OpenSIPS-Users] Delay requests to avoid CPS provider's CPS limits

Hi Did you take a look at async function of opensips 2.0? I think you may insert some delay with this. Em 27/03/2016 7:03 PM, "Mateusz Bartczak" escreveu: > Hi > > I'm looking for a solution to nicely handle call spikes > > The issue is like this: > > I have customer

[OpenSIPS-Users] BYE generated locally going to private IP, not public IP

Hi I'm facing an strange issue when my Opensips instance hangs up a call, generating BYE to both sides (timeout on dialog module or rtpproxy). The BYE is sent to both sides but A side is behind NAT and the BYE is sent to the local IP address and not to the public one. See trace bellow: Customer

Re: [OpenSIPS-Users] BYE generated locally going to private IP, not public IP

ng traffic to public > Internet, should not use at all private IPs. > > Bottom line, the broken link in your scenario is the 172.20.17.11 proxy > before your opensips. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >

Re: [OpenSIPS-Users] Fix_nated_SDP works very well. How to do more changes in the SDP?

Hi Rodrigo In theory you can modify any SIP/SDP using text ops: http://www.opensips.org/html/docs/modules/1.6.x/textops.html#id293600 I don't know any other way... Regards On Mon, Apr 18, 2016 at 6:17 PM, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > Fix_nated_sdp()

Re: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi Cesar For realtime querying, you can use dialog profiles. Set a profile (for example: inbound) on all calls like this: set_dlg_profile("inbound"); Then read this using FIFO. I just didn't get the "Kamailio Sip Server running Opensips". Regards On Wed, Jul 27, 2016 at 2:29 PM, Cesar Alberto

Re: [OpenSIPS-Users] Get concurrent calls from sip server.

On my sample, you should run: opensipsctl fifo profile_get_size inbound Using dlg_list you should get something like this: # opensipsctl fifo dlg_list dialog:: hash=84:1411689852 state:: 4 user_flags:: 0 timestart:: 1469646635 datestart:: 2016-07-27 16:10:35

Re: [OpenSIPS-Users] Can I count the number of active calls in a particular time interval through opensips.

I use something different. I defined a profile with value: modparam("dialog", "profiles_with_value", "customercps"); Then use this profile in a counter with customer_id + current timestamp: set_dlg_profile("customercps", "$avp(customer_id)$Ts"); This way I can count current calls on current

Re: [OpenSIPS-Users] OpenSips rejecting too many calls

Hi Chandan I would suggest upgrading to 1.11.X first, it will be very straightforward. If you still have problems, migrating to 2.2 will probably not solve them and you may find new ones. Regards On Wed, Aug 10, 2016 at 9:16 AM, Chandan PR wrote: > Hi Guys, > > We are

Re: [OpenSIPS-Users] Opensips switch from SQL to NoSQL

Hi Feroze I think you should first understand exactly what is written to MySQL and answer yourself why do you want to move to NoSQL. Opensips has many modules that access DB structures like ACC, Subscribers, Address, Location, Loadbalancer, cachedb, etc. Which do you want to move to NoSQL? And

[OpenSIPS-Users] ACC failover solution

Hi I need some help to provide a failover solution to do accounting on our calls. If ACC fails to write on MySQL accounting system (Like MySQL), the record is lost. Is there any failover solution on the ACC module, to record on another kind of DB, only if MySQL failed? I know I could write on

Re: [OpenSIPS-Users] ACC failover solution

rw@host2/testa > ") > > Your acc db url would look something like this: > modparam("acc", "db_url", "virtual://accounting”) > > jarrod > > On Jul 2, 2016, at 4:57 PM, Daniel Zanutti > aniel.zanu...@gmail.com> wrote: > > Hi > > I need

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

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

[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] 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

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] Help with big amount of data for routing

Hi Miha I have a similar situation, but around 20 M routes. The native routing mecanims wasn't performing well, so I developed a custom mecanism using Opensips scripting. Everything is stored on MySQL database. The best approach was use avp_db_query to get the route, the primary key (and index)

[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

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

Hi Short: Do you guys see any problem on removing the to-tag of all 1XX messages? Long description: If user A calls number . Opensips forward to gateway B which replies with 183 session progress, then refuses with 503. Opensips then call gateway C which replies with 183 then 200 OK. User

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

Hi Ben and Alex Thanks for the quick response. This is exactly what I was worrying about, get unto some unpredictable state like A rejecting some packages. Agree that putting a B2B in front/back of opensips would solve but it´s another server =( Thanks for the advices guys! On Fri, Oct 21,

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

could run on the existing server. There > shouldn’t be any need for a B2B server and a proxy server; the B2B would > replace the proxy. Unless you just can’t change the config of that server > for some reason? > > > > > > Ben Newlin > > > > *From: *<users-boun

[OpenSIPS-Users] Doubt on B2B Scenario

Hi I'm having a problem with a specific client. It ignores Record-route and Via fields order and send a BYE to last destination, not the first one. This clearly breaks RFC and there's no way to solve this with his bug software. To solve this, I was planning to implement a B2B scenario to it, so

Re: [OpenSIPS-Users] Doubt on B2B Scenario

on dialog module: > http://www.opensips.org/html/docs/modules/2.2.x/topology_hiding.html > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 20.10.2016 23:21, Daniel Zanutti wrote: > > Hi > > I'm having a p

Re: [OpenSIPS-Users] Doubt on B2B Scenario

. Is it possible? Thanks On Thu, Oct 20, 2016 at 6:46 PM, Daniel Zanutti <daniel.zanu...@gmail.com> wrote: > Hi Bogdan > > I'll take a look deeply. > > Thanks > > Em 20 de out de 2016 6:39 PM, "Bogdan-Andrei Iancu" <bog...@opensips.org> > escreveu: > >>

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] 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

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

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] 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

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] 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

, 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

; } > } > 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(

[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] 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 >

[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] 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 >> >>

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

[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] 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.

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

[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

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 &

[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

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

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

[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

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

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

[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

[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] 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] 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] 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

[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

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] 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] 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] 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] 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

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 > 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] 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] 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

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] 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

[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] 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] 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 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] 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

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] 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] 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] 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] 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] 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] 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] 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] 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

, 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

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