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

We face similar issues, we had 1 * server per VLAN, where each * had the same IP address, we achieve this with NAT. <> Asterisk 1 <> Asterisk 2 SIP LB <---> NAT <> Asterisk 3 Your NAT device needs to be SIP aware, in our case

Re: [SR-Users] Websocket TLS Issue

Are you using self-signed certs? or public certs signed by public CA. On Thu, Feb 2, 2017 at 1:34 PM, Ludovic Gasc wrote: > Hi, > > It might be a stupid question, but why you don't have WebSockets without > TLS between HAProxy and Kamailio ? > I've a similar setup to enable us

Re: [SR-Users] Rewrite From domain

2nd try. Thanks On Fri, Dec 30, 2016 at 1:28 PM, Gonzalo Gasca Meza <gascagonz...@gmail.com> wrote: > Hi Igor, > Please look for "Masking Twilio call from:" > > > On Sun, Dec 25, 2016 at 12:47 PM, Igor Olhovskiy <igorolhovs...@gmail.com> > wrote: > >

Re: [SR-Users] Rewrite From domain

Regards, Igor > > On 23 дек. 2016 г., 10:51 +0200, Gonzalo Gasca Meza < > gascagonz...@gmail.com>, wrote: > > Hi Daniel, > > I added xlog and $mb here is the debug: > > I still cant figure out how to overwrite the From field. > > http://pastebin.com/vMruA0tP >

Re: [SR-Users] Rewrite From domain

, > > for the case when it doesn't work, do you see the xlog message printed? > > Cheers, > Daniel > > On 21/12/2016 08:37, Gonzalo Gasca Meza wrote: > > Hi Daniel, > > Thanks for the advise, I'm using the following configuration, > > # Manage

Re: [SR-Users] Rewrite From domain

om() > exported by uac module. > > The best place to do updates to headers for outgoing traffic is in a > branch_route block. > > Cheers, > Daniel > > On 13/12/2016 12:05, Gonzalo Gasca Meza wrote: > > Hi all, > > I'm using Kamailio to forward calls between 2

[SR-Users] Rewrite From domain

Hi all, I'm using Kamailio to forward calls between 2 Service Providers and I need to rewrite the From header "domain" URI. Example: From: "+188" to From: "+188" *> *Call flow:* Phone A --- >

Re: [SR-Users] Kamailio 4.4 VoiceMail

Thanks Daniel On Thu, Nov 10, 2016 at 8:04 AM, Daniel Tryba <d.tr...@pocos.nl> wrote: > On Sun, Nov 06, 2016 at 02:22:06AM -0800, Gonzalo Gasca Meza wrote: > > Currently in sample configuration script, seems to be that value: > $avp(oexten) > > is used to redire

[SR-Users] Kamailio 4.4 VoiceMail

I have the following call flow: INVITE -> sip:gonz...@sip.parzee.io -- TLS/TCP/UDP -> *KAMAILIO* - DB Lookup -> INVITE sip:gonzal...@test.external.com;transport=tls (Phone1) If Phone1 is Busy or No answer, I want call to go to VM. Phone1, is not registered to Kamailio, nor I'm using Realtime

Re: [SR-Users] Kamailio DB lookup

It worked perfectly with alias_db_lookup and add transport in domain field. Thanks On Sat, Nov 5, 2016 at 10:00 AM, Gonzalo Gasca Meza <gascagonz...@gmail.com> wrote: > Thanks for replying, > > What I want to achieve is once I receive the SIP INVITE, I need to > translate it

Re: [SR-Users] Kamailio DB lookup

as well as adding the Transport in order to have Kamailio force the new SIP request in TLS. Is this the right way? Thank you -Gonzalo On Sat, Nov 5, 2016 at 6:01 AM, Anthony Joseph Messina < amess...@messinet.com> wrote: > On Friday, November 4, 2016 1:58:48 AM CDT Gonzalo Ga

[SR-Users] Kamailio DB lookup

Hi Kamailions, Kamailio 4.3 will receive calls (SIP TLS) for users in this domain: *sip.parzee.io . *Example sip:gonz...@sip.parzee.io When call is received by Kamailio I need to overwrite R-URI and transport. I should keep a mapping per user. ( gonzalo -> gonzalo58)

Re: [SR-Users] How should a PBX behave when a user dials a DID configured for the same or another tenant on that PBX?

Hi Antonio, In general PBXs (Example: Cisco Callmanager, Asterisk) handles extensions and routes to PSTN differently. Example: Extension 1. 4082186571 Extension 2. 4082186572 Route Pattern (SJ Local) 408XXX There are normally 2 approaches for the problem you describe. a) Context access b)

Re: [SR-Users] IPSec supporting open source SIP Server

We recently implemented strongswan project which supports ipsec. https://www.strongswan.org/ On Jun 2, 2015 6:59 AM, Måns Nilsson mansa...@besserwisser.org wrote: Subject: [SR-Users] IPSec supporting open source SIP Server Date: Mon, Jun 01, 2015 at 02:51:00PM +0530 Quoting Priyaranjan Nayak (

Re: [SR-Users] Video Key-Frame Request using RTCP FIR or SIP INFO message

Hi Muhammad Can you comment if initially endpoints are receiving what is necessary to start sending Media at signaling level. For example: Successful ICE and SRTP-SDES/DTLS negotiation. I see two issues here: a) Establish a successful call b) Once call is established how to deal with packet

Re: [SR-Users] Need help on WebRTC with Kamailio as proxy

Are you terminating media in Kamailio or just handling WS communication? If yes which version of Kamailio and rtp-proxy ? Have you tried passing media directly between Browser and Kamailio with any TURN server? Are you using latest Chrome version or FF ? A working sample config using the