Re: [OpenSIPS-Users] Cache design

Hi John, The cachedb_x module do implement support for different noSQL like DBs (local - in opensips memory, mongoDB, memcached, redis, etc). The sql_cache is new transparent way of caching an SQL table in OpenSIPS memory (while reading data from an SQL table). the cache interface works

Re: [OpenSIPS-Users] Planing the future OpenSIPS releases

Hi Dragomir, What you mean ? All OpenSIPS versions do have the radius support; starting with 2.2 we will have the async support for radius too. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.01.2016 22:03, Dragomir Haralambiev wrote: Hi,

Re: [OpenSIPS-Users] MenuConfig Compile Fail on Solaris Sparc

I've put together a patch for all Solaris-related issues [1], for the latest 2.1 code. Please revert the previous fix we did here, and apply this patch with: git apply solaris_bad_compile.patch if you're running off a git repository, or with: patch -p1 < solaris_bad_compile.patch

Re: [OpenSIPS-Users] tomcat external app listening to OpenSIPS events

Hi Julian, So, you say OpenSIPS is actually connecting via HTTP to tomcat in order to deliver the event, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.01.2016 03:02, Julian Kay wrote: Hi; Thanks for the help! the event

Re: [OpenSIPS-Users] B2B BYEs

Hi Søren, With or without restarting the B2B isn;t the BYE sent by client to the B2B instance ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.01.2016 16:32, Søren Andersen wrote: Hello there, I’ve some strange issues with the B2B

Re: [OpenSIPS-Users] Userloc stats are showing zero registered users

Hi Husnain, The statistics (for usrloc) are computed based on the in-memory (cached) data. In db_mode 3 (db only), there is nothing cached in memory, so not statistics. if you want statistics, use db_mode 1 or 2 ; mode 1 updates the db table in realtime, but has a performance penalty ; mode

Re: [OpenSIPS-Users] uac_replace_from multiple times?

No you can't. Use a variable to store the from and replace it once, just before to send out the message. Il 13/01/2016 14.32, Søren Andersen ha scritto: Hello, I’m wondering if it’s possible to use uac_replace_from multiple times? – fx. Inbound call gets changed by uac_replace_from and

Re: [OpenSIPS-Users] uac_replace_from multiple times?

Hi, Thanks for your mail. - Actually I need to send the call out to my client without the prefix, and if the client don't answer the phone the call gets sent to another phone number via my ISP. Do you have a smart solution for this? /Søren Fra: users-boun...@lists.opensips.org

Re: [OpenSIPS-Users] presence xpidf

Hi Stas, You say you see the DOCTYPE line in NOTIFY packets and this is supported by OpenSIPS ? Now, on Polycom extension - if it is something end-2-end, it means it does not require a presence server and everything should be between end points by using SUBSCRIBE and NOTIFY (no PUBLISH, as

Re: [OpenSIPS-Users] Userloc stats are showing zero registered users

Hi Bogdan, Actually we have 2 registrar servers in production environment half of the users are registered with one server and other half on the second server so whenever inbound call from PSTN comes to our SBC, The SBC check the user in location table of both opensips servers and then route call

[OpenSIPS-Users] uac_replace_from multiple times?

Hello, I'm wondering if it's possible to use uac_replace_from multiple times? - fx. Inbound call gets changed by uac_replace_from and removed the +45 prefix. - But sometimes I need to forward the call back to my ISP, and they need to have +45 in the from header. But if I try to use the

Re: [OpenSIPS-Users] B2B BYEs

Hi Bogdan-Andrei, Only if I reload the B2B the BYEs is sent directly to the client. - But the funny thing is this only happens if the client is receives a call. If the client initialize the call everything works fine. /Søren Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sendt: 13.

Re: [OpenSIPS-Users] X-Auth-IP How?

Thank You. do you have an example as to how to have the Freeswitch ACL work with he via headers you discussed? I have followed many guides for setting up OpenSIPs as a load balancer for Freeswitch, but none of them addressed making the Freeswitch ACL functional. On 1/13/2016 12:49 PM, Stefano

Re: [OpenSIPS-Users] X-Auth-IP How?

Astpp comes with and opensips config file which add X-auth automatically... On 13 Jan 2016 18:13, "Tim King" wrote: > Thank You. do you have an example as to how to have the Freeswitch ACL > work with he via headers you discussed? I have followed many guides for > setting up

[OpenSIPS-Users] T_Relay Re-Invite

Hi everyone, I have a problem with t_relay. When the session is established (last 200 OK) the phone receive a in-dialog invite from the PBX directly and since there every packet don’t go thought OpenSIPS but goes directly to the PBX meaning that I have no way to do anything on the active

Re: [OpenSIPS-Users] T_Relay Re-Invite

Hi Eric, Thank you for the quick answer. With record_route it go thought the proxy but every in-session messages aren’t sent to the PBX or to the phone from the PBX. Just to make sure I’ve tried adding : xlog("DEBUG : METHOD $rm”); right after : route { and I don’t pick any log from it. Thank

Re: [OpenSIPS-Users] MenuConfig Compile Fail on Solaris Sparc

Liviu, I removed the previous change and applied the patch and here is the error that I got: Compiling net/net_udp.c In file included from net/../locking.h:66:0, from net/../statistics.h:143, from net/../pt.h:70, from net/net_udp.c:30:

Re: [OpenSIPS-Users] Planing the future OpenSIPS releases

Hi Bogdan, Thanks for your replay. Here is email describe Radius problem in Opensips 2.1. === Hi Arsen, If i completely understand the problem, the server doesn't provide the deserved response. The problem in this case is that rc_send_server

Re: [OpenSIPS-Users] tomcat external app listening to OpenSIPS events

HI, I really appreciate you taking the time!! yes OpenSIPS is connecting to Tomcat, the problem I see (I think) is when the event is being raised OpenSIPS is NOT sending the complete URI. Tomcat server returns an error of 404. the Tomcat logs seem to indicate OpenSIPS is only be sending

[OpenSIPS-Users] latest opensips 2.2 dev revision 055f4b1

I am seeing some strangeness in which opensips delivers a large 200 ok/w sdp at the size of 5461 then opensips proceeds by sending a fin and a rst. I can supply logs if needed. This ends up causing issues with chrome and sipjs it no longer renders media or sends it. Thanks, Tito

[OpenSIPS-Users] Opensips 1.8 ds_list behavior in dispatcher

Hello, I am observing some unusual behavior of the ds_list command when adding and removing gateways in the dispatcher table for opensips 1.8 I am running the following sequence of commands - ./opensipsctl dispatcher addgw 40 sip:2.2.2.4:5060 0 'test1' ./opensipsctl dispatcher addgw 40

Re: [OpenSIPS-Users] OpenSIPS 2.1.2 and 1.11.6 Releases

Razvan, We are faving an issue in 1.11.5, that is, intermittently once or twice a week opensips stops responding to sip messages. All active sessions are dropped. Thread is still running, network ports are still showing in LISTEN state, and no cli access. As this is happening in production

Re: [OpenSIPS-Users] uac_replace_from multiple times?

If you use uac_replace_from in a branch route then the changes are specific to that branch. If the call fails the header will be reverted and can be modified again in another branch. Ben Newlin From: > on behalf of

Re: [OpenSIPS-Users] opensips 1.11.5-tls , SNMP not working - No Such Object available on this agent at this OID

Hi Игорь Павлов, Tried to set up snmp from scratch. Followed the tutorial in OpenSIPS[0], managed to start snmp but had a hard time setting up the opensips mibs. Found out this tutorial [1] which told to put the mibs in "/var/lib/mibs/ietf/", and adapted the command from there

Re: [OpenSIPS-Users] presence xpidf

Hi Bogdan, I do not think the DOCTYPE is the problem here. What I see is that when I use MI to publish this application/xpidf doc, OpenSIPS does not want to parse the document, and if I understand correct, this is because this type of document does not have XML branch. You are right, about

[OpenSIPS-Users] X-Auth-IP How?

I have read countless articles now talking about using x-auth-ip as a method for using OpenSIPs as a load balancer serving to a cluster of Freeswtich servers and having a method to maintain the original IP address. Direct from the Freeswitch wiki it states: apply-proxy-acl Use the IP

Re: [OpenSIPS-Users] X-Auth-IP How?

what about appendHf("X-Auth-IP: $si"); in your script? however there are the "via" headers already to do this job Il 13/01/2016 18.43, Tim King ha scritto: I have read countless articles now talking about using x-auth-ip as a method for using OpenSIPs as a load balancer serving to a cluster

Re: [OpenSIPS-Users] X-Auth-IP How?

to be precise: append_hf("X-Auth-IP: $i\r\n"); according to documentation Il 13/01/2016 18.43, Tim King ha scritto: I have read countless articles now talking about using x-auth-ip as a method for using OpenSIPs as a load balancer serving to a cluster of Freeswtich servers and having a

[OpenSIPS-Users] Config Advertised_address makes OpenSIPS forward SIP ACK when it must not. How to fix? Help.

Dear OpenSIPS-users; My OpenSIPS is behind a NAT. The public IP is xxx.xxx.xxx.xx. So, I have configured advertised_address="xxx.xxx.xxx.xx" in my OpenSIPS. It seems that such new configurations takes appropriated effect, because a Record-Route header field in a SIP OK message (for a call