[OpenSIPS-Users] error migrating database to 3.5

Hello, I used opensips-cli to migrate database from 3.4 to 3.5 but I got an error: ERROR: unsupported migration flavour: 3.4_to_3.5 I used opensips-cli from github https://github.com/OpenSIPS/opensips-cli $ opensips-cli --version OpenSIPS CLI 0.2.0 I checked modules/database.py and 3.4_to_3.5

Re: [OpenSIPS-Users] registrant example

Hello. I have a strange problem. I'm using docker official image. It seems unable to load the configuration file for some reason. opensips  | Sep 28 15:44:28 [1] WARNING:core:exec_preprocessor: no output from the preprocessor! Does it print to standard output? opensips  | Sep 28 15:44:28 [1]

Re: [OpenSIPS-Users] forwarding custom variables

Yes. It's possible. The "variable" must be in a custom header. Ciao s Il 15/03/2017 19:41, Roberto Cantalapiedra ha scritto: Hi, I would like to know if it is possible this scenario: - Caller initialize the call and sending specific variable to opensips. - opensips match one of those variab

Re: [OpenSIPS-Users] how to catch hangup event inside opensips

please explain better Il 25 Gennaio 2017 13:24:57 CET, Khalil Khamlichi ha scritto: >there is one more question we still struggling with, how to run a route >at >each hangup, we never found any way to do it inside opensips. right now >we >catching the event upon cdr insert in db. -- Inviato dal

Re: [OpenSIPS-Users] Registrar with IP authentication - selecting variables from DB

Use 0.0.0.0/0 for those without IP filter. s Il 13/01/2017 12:09, maatohewetbi ha scritto: I think You don't understand. My Opensips should work in this scenario: 1. When user wants to register, I have to check whether his sip login is in address table (which can be stored in context_info for

Re: [OpenSIPS-Users] tls_mgm

l. So what was your solution to solve the conflict? Deployed a custom deb? Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 08/20/2016 11:14 PM, Stefano Pisani wrote: Hello Răzvan, in my previous email I told you that I have upgraded openssl to 1.0.2h

Re: [OpenSIPS-Users] tls_mgm

Hello Răzvan, in my previous email I told you that I have upgraded openssl to 1.0.2h then the error was different. After that I realized that there was a conflict between ubuntu openssl package and new openssl. Finally, after fixed that, tls_mgm module is working properly. I'm using ubuntu 16.

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

PM, Stefano Pisani wrote: It seems to be the last available version for ubuntu 16.04 LTS. What can I do? #apt-get install openssl Reading package lists... Done Building dependency tree Reading state information... Done openssl is already the newest version (1.0.2g-1ubuntu4.1) Thanks Stefano

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

, Răzvan Crainea ha scritto: Seems to be a similar problem to the 1.0.1e-fips library. Could you try to upgrade the openssl package? Thanks, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 08/17/2016 12:48 PM, Stefano Pisani wrote: I miss the last two lines (gdb) info line

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

7;gdb ./opensips' and run the following commands: info line *0x7f0ce10d8a70 info line *0x7f0ce10d8550 Thanks, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 08/17/2016 12:06 PM, Stefano Pisani wrote: Hi Răzvan. These are the new logs. I hope it helps to fix this. Aug 17 05:0

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

wing commands: info line *0x7f0ce10d8a70 info line *0x7f0ce10d8550 Thanks, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 08/17/2016 12:06 PM, Stefano Pisani wrote: Hi Răzvan. These are the new logs. I hope it helps to fix this. Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initi

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

e following commands: info line *0x7f0ce10d8a70 info line *0x7f0ce10d8550 Thanks, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 08/17/2016 12:06 PM, Stefano Pisani wrote: Hi Răzvan. These are the new logs. I hope it helps to fix this. Aug 17 05:04:40 [3997] INFO:tls_mgm:mo

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

did not push the changes. Could you please re-clone and try again? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 08/16/2016 07:26 PM, Stefano Pisani wrote: I have this result. # ./opensips -V version: opensips 2.3.0-dev (x86_64/linux) flags: STATS: On

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

m, or you are using a wrong path. Could you please double-check? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 08/16/2016 06:56 PM, Stefano Pisani wrote: I cloned the source from |git clone https://github.com/OpenSIPS/opensips.git opensips_head the error appear

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

extra debugging in the master branch that might help you debug this. Could you please take the latest git version/deb and try to run again? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 08/16/2016 10:52 AM, Stefano Pisani wrote: I'm trying to enable wss on open

[OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS version: opensips 2.2.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, B

Re: [OpenSIPS-Users] opensips transparent technology

Where is their real phone number? Do you have it in a database? You can change the From header to show the real phone number. Il 24/01/2016 12.22, MichaelLeung ha scritto: Hi all i was trying to make my opensips users to sent their real phone number when they call . what is the name of thi

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 method

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 o

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

Re: [OpenSIPS-Users] Can OpenSIPS can be used as a WebRTC gateway for JsSIP client and WebRTC client?

Take a look to OverSIP. Il 05/01/2016 07.42, suganthi karthick ha scritto: Hi all, I need to implement a WebRTC gateway for an existing conference bridge. The WebRTC gateway has to support Signaling, ICE, DTLS-SRTP. The webrtc clients can be JsSIP or any JSON based webrtc client. The confer

Re: [OpenSIPS-Users] Try to setup ENUM NRENUM support...

Ciao Michele, per uscire con il contact corretto devi usare l'advertised_address / advertised_port se il serves ha solo un IP interno. Questo fa nascere vari problemi perché la correzione viene fatta anche per verso i client interni. Non c'è il concetto di Lan come per asterisk, almeno finora.

Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

Stun/Turn/Ice are usefull where Client is behind a NAT and OpenSIPS has public IP. You could use nathelper modules instead of Stun, to set the right IPs in the messages from client. If OpenSIPS is behind a NAT too (it has private IP) you must use RTPProxy too with a proper configuration. Il 2

[OpenSIPS-Users] Save in on_reply route

Hi Guys, I'm facing an issue. I'm using opensips as proxy for REGISTER too. So I need to save the registration only when 200 OK returns from media server. I do save("location","r") in onreply_route and it works but there is a problem. The original contact was fixed using fix_nated_contact() but

[OpenSIPS-Users] Save with AOR

Hi Guys, I'm using OpenSIPS 1.11. save("location","r","sip:user@1.1.1.1:3") in REPLY ROUTE is not working in the location table I find always a different value from the AOR I set. Is it a bug? Thanks s ___ Users mailing list Users@lists.opensip

[OpenSIPS-Users] transformation

Hello guys, I have problems with these to simple transformations. $var(uri) = "$ct.fields(uri)"; $var(ct) = "$(var(uri){uri.user})" + "@" + "$(var(uri){uri.host})"; what's wrong with them? s ___ Users mailing list Users@lists.opensips.org http://list

Re: [OpenSIPS-Users] forwarding calls from Asterisk to OpenSIPs

Just define a peer in sip.conf then Dial(SIP/peer_name/OpensipsAccount) on extensions.conf. Very simple. regards, s Il 07/05/2015 08:32, Julian Kay ha scritto: Hello Everyone; I want to be able to forward calls from Asterisk to a phone registered with OpenSIPs, can anyone give me some infor

Re: [OpenSIPS-Users] SIP and RTP Proxy without local user base

Remember that if the audio path is p2p the DMTF tones do not work anymore, so you can't transfer a call, etc. You should use SIP INFO if supported. For the audio path give a look to the asterisk options /directmedia/=yes and directrtpsetup=yes. You could not need to use openSIPS and rtpproxy.

Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

You should say something more about your issue. Il 14/03/2015 17:22, mahan77 ha scritto: Hi Danilo, I’m having problem with OpenSips => Asterisk connection. Can you able to mail me your working OpenSips scripts. mail at Sathees.co.uk appreciate sathees ---

Re: [OpenSIPS-Users] OpenSipS as a simple proxy

The purpose is to create a privileged network path beetween your server and the local clients. Is it? Il 04/02/2015 20.58, Dovid Bender ha scritto: Hi, We have a cluster in the US running custom software. We have clients in countries where their ISP’s traffic out of the country is not the b

[OpenSIPS-Users] ACK never leaves opensips

I have a strange issue with an ACK that never leaves Opensips. It disappears. This is the ACK message incoming dumped with ngrep U publicIP1:32769 -> publicIPOpenSIPS:5172 ACK sip:s@publicIP2:6050 SIP/2.0. Via: SIP/2.0/UDP 192.168.4.53:32769;branch=z9hG4bK-nt6kbhw2yq7b;rport. Route: . From: "1

Re: [OpenSIPS-Users] Some phones can call, others not...

In the first case the last UDP packet (INVITE with authentication header) is fragmented. It's very common to lost fragmented UDP packet. s Il 28/11/2014 09.42, Michele Pinassi ha scritto: Hi all, i'm experiencing a strange issue. Some VoIP phones, like mine (5002), cannot call other phones (l

Re: [OpenSIPS-Users] URGENT! uac_auth PSTN gateway authentication issue

Check if the cseq was incremented by one in the second try. Use ngrep. Il 24/08/2014 22.24, Satish Patel ha scritto: Hi, my Opensips (UAC) registered to PSTN gateway and now i am trying to call using my SIPphone which is register to opensip but no success. I am getting 407 Proxy authentica

Re: [OpenSIPS-Users] 403 Rely forbidden

is what I get: # opensipsctl domain add 10.10.10.3 INFO: execute '/sbin/opensipsctl domain reload' to synchronize cache and database [root@lion opensips]# opensipsctl domain reload 500 command 'domain_reload' not available What's wrong again? Thank you!

Re: [OpenSIPS-Users] 403 Rely forbidden

fined, but don't know how to achieve that. Any tips? Thank you! On 04/25/2014 09:14 AM, Stefano Pisani wrote: If you do not define a local domain OpenSIPS can't understand that SIP messase is for him. Il 25/04/2014 13.36, i...@vintageelectronics.ca ha scritto: Most likely not, a

Re: [OpenSIPS-Users] 403 Rely forbidden

G4bK-313736-c420c7a03fba6256d5863512fbdd6683.. Server: OpenSIPS (1.10.1-tls (x86_64/linux))..Content-Length: 0 #^Cexit 52 received, 0 dropped On 04/24/2014 06:26 PM, Stefano Pisani wrote: Could you dump the SIP dialog using ngrep on opensips server? and post it? Il 25/04/2014 00

Re: [OpenSIPS-Users] 403 Rely forbidden

Could you dump the SIP dialog using ngrep on opensips server? and post it? Il 25/04/2014 00.17, i...@vintageelectronics.ca ha scritto: My installation is essentially vanilla, whatever came with the rpms except for listen port/IP. Where in which file should I seek the ruri header and what should

Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't I?

believe the answer is yes. The above header field is in the OK message. Cordially, Peter Nayland Kust Director of Technologies BusinesSuites 24624 Interstate 45 North, Suite 200 Houston, TX 77070 peter.k...@businessuites.com -Original Message- From: Stefano Pisani [mailto:stefano.pis

Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't I?

d what the best/safest and most stable method is for altering that header, if necessary. Cordially, Peter Nayland Kust Director of Technologies BusinesSuites 24624 Interstate 45 North, Suite 200 Houston, TX 77070 peter.k...@businessuites.com -Original Message----- From: Stef

Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't I?

BYE was never received. Check the Contact header in OK message. Is it right? Check also the request route. Are they present? Probably NOT because BYE go to The UAC and not to the PROXY. Cheers, s Il 29/03/2014 17.17, Peter Kust ha scritto: Also, this is how the SIP messaging is proceeding, st

Re: [OpenSIPS-Users] RES: RES: Error in Module Permissions

So I was right :-) Il 18/03/2014 20.11, Alcindo Schleder ha scritto: Hi all.. I found the error. The version of the module was incorrect. The opensips was loaded through the rpm package version 1.9.2. I downloaded the source and recompiled ... voilà it worked. []s *De:*users-boun...@lists

Re: [OpenSIPS-Users] Error in Module Permissions

missing loadmodule? Il 18/03/2014 18.03, Alcindo Schleder ha scritto: I'm trying to use the function and check_address get an error compiling the script. Script line 390. if (check_address("1", "$si", "$sp")) { xlog("IP Allow Routing to $si");

Re: [OpenSIPS-Users] too many hops

It seems an addressing issue. Could you post your opensips.cfg? Are you able to log the sip session using ngrep? Post it. Il 10/03/2014 11.03, Mike Claudi Pedersen ha scritto: im trying to establish connection between 2 phones user1: 43384001 user2: 43384002 i have user 2 added to usrloc with t

Re: [OpenSIPS-Users] Redirect

Simply replace the $ru ($ru = NEWURI) and the call goes to the right destination If the remapping is fixed you and use ALIAS. for example. Il 10/03/2014 08.11, Mike Claudi Pedersen ha scritto: Can someone please help me ind the right direction, i need to implement a system to rewrite the desti

Re: [OpenSIPS-Users] Renegotiation

You can trap the 415 response from the called peer and send the call through asterisk to get transcoding. Il 27/02/2014 16.51, Jorge Ortea ha scritto: Hi all, I have a scenario with OpenSIPS 1.8 and Asterisks 1.4. Proxy SIP has two ways to manage a call, the first is B2BUA and second is be

Re: [OpenSIPS-Users] Adding Proxy-Authorization header

You can use module UAC_AUTH Il 24/02/2014 16.18, Diego Barberio ha scritto: Hi all, I have opensips registered to an IP-PBX using registrant module and I want to make an outbound call to that PBX through the proxy. I'm sending and INVITE from my application to the proxy with a From that is

Re: [OpenSIPS-Users] Initializing SIP messages from routing

You can develop something using perl and Net::SIP module. Il 13/02/2014 19.45, Jayesh Nambiar ha scritto: Hi, CRBT is caller ring back tone. What you are primarily looking at is sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both legs of the call. So when you ge

Re: [OpenSIPS-Users] media server behind nat

There are a problem in the network configuration. Opensips has only one network card, in this way it cound only use one IP but it actually has two IPs (one external and one interna). You need to enable a second network card with a public IP (in the DMZ) and use mhomed=1 in configuration to chan

Re: [OpenSIPS-Users] media server behind nat

Please post a link with your network drawing. Your description is unclear. Il 05/02/2014 20.19, Tony Ward ha scritto: Hello, I currently have a media server behind a nat firewall with calls delivered via a PSTN Trunk. I want to add a 2nd media server and route calls to either depending upon

Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

if( ( $(avp(sourceip_mask){ip.pton}) & $(avp(sourceip){ip.pton}) ) == ( $(avp(sourceip_mask){ip.pton}) & $(si{ip.pton}) ) ) { xlog("L_INFO", " ip $si belongs to $au\n"); } else { xlog("L_INFO", " ip $si does not belong to $au\n"); sl_send_reply("403", "Forbidden"); exit; } Why

Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

why you are using "[]"? use "()" instead. Il 01/02/2014 17.44, Edwin ha scritto: This helped a bit, so I came up with: $var(sourceip_net) = $(avp(sourceip_mask){ip.pton}) & $(avp(sourceip){ip.pton}); $var(si_net) = $(avp(sourceip_mask){ip.pton}) & $(si{ip.pton}); if($var(sourceip_net) == $var(

Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

$var(mask) = "255.255.0.0"; $var(ip) = "192.168.2.134"; $var(net) = $(var(mask){ip.pton}) & $(var(ip){ip.pton}); if ($(var(net){ip.ntop}) == "192.168.0.0") xlog("IP is in 192.168.0.0/16 network\n"); Il 30/01/2014 20.47, Edwin ha scritto: Stefano, I know, and maybe I will use the permissi

Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

You can create an external perl script that to the job. Il 30/01/2014 12.38, Edwin Haselhoff ha scritto: Stefano, I tested the permission module but changes to the table are not 'real time', I have to reload the table every time (or did I miss something?). Stefano Pisani schre

Re: [OpenSIPS-Users] How to connect opensips to esternal VOIP server?

UAC_REGISTRAR is not enought becouse you need something else to make call with proxy autentication like UAC_AUTH and failure_route Il 30/01/2014 17.40, Nikita Tarasov ha scritto: Where are any ability to make calls from opensips with registration? For example registered with uac_registrant? *

Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

Hi, use module permission. s Il 30/01/2014 12.21, Edwin Haselhoff ha scritto: Hi all, For security reasons I want to check if the $si ip is part of ip and subnet of a subscriber so added '$(avp(sourceip)' and '$(avp(sourceip_mask)' to the subscriber table. (I know I can use permissions modul

Re: [OpenSIPS-Users] changing header contact

If you have public and private IP on different eth ports you can use mhost = 1 Il 27/01/2014 20.58, discodo...@aol.com ha scritto: Hello all, I am trying to setup a opensips server that has 2 ip's one internal and one external IP. I have set the rewritehostport to an external IP. When I se

Re: [OpenSIPS-Users] Ref: how can we replace URI Header more than once per call?

The uac_replace_to can be used once. Are you sure you need to replace to header? Could be enough set a new RURI instead. What do you want do exactly? regards, Il 09/12/2013 12.57, AMPTEL PTY LTD | RuvixTel ha scritto: Hi all Just wondering, if anyone able to assist us with below: We are us

Re: [OpenSIPS-Users] Can I use alias_db to change credentials?

STN is there a module I can use, (using opensips 1.8.x), and how exactly do I use it? (I know how to pull the username and password and destination number from the database) an example would greatly help me. Thanks Walter. On 24/11/13 4:46 pm, Stefano Pisani wrote: You can query a ad

Re: [OpenSIPS-Users] Can I use alias_db to change credentials?

You can query a ad using avp_query and get new credentials, then use them in the routing script. This is not complicated. Il 24/11/2013 01.38, Walter Klomp ha scritto: Hi, I want to enable call forwarding feature to pstn, but the outbound calls need to be authenticated to the outbound proxy

Re: [OpenSIPS-Users] Re A bridge Call via Opensips

That is the parallel forking. You should see documentation about it. The simplest solution is to register the 3 SIP phones using the same SIP account (ID:200) Il 11/11/2013 07.01, steven chew ha scritto: Hi All, How are you? I would like to know how to configure a bridge call in opensips.cf

Re: [OpenSIPS-Users] Modify/Edit INVITE

take a look to textops module Il 09/11/2013 16.06, Aziz Bayd ha scritto: Hello. I have a question about OpenSIPS configuration. Can I edit or modify the INVITE message content? For example, remove "To" or "Via" fields in the INVITE content. If yes, how can I do? Thanks. 2013/11/9, Aziz

Re: [OpenSIPS-Users] Multiple aliases for one sip account

I agree. Il 30/10/2013 01.38, Dario Busso ha scritto: La tua richiesta è veramente poco chiara. Your question is really confused. May you try to describe it better? -ddB Il 28/ott/2013 18:00 "Manuela Pigini" > ha scritto: Hello, we are developing a voip c

Re: [OpenSIPS-Users] Opensip as transaprent inbound proxy for asterisk

Just FYI you need to write down some code in asterisk to manage the new header obviusly Il 11/10/2013 02.30, bluerain ha scritto: Just FYI, I tried, I insert your line in the method invite and right before the routing, Asterisk didn't seem to care. It still care about the prior Hop IP. So wh

Re: [OpenSIPS-Users] Opensip as transaprent inbound proxy for asterisk

You do not need to manipulate core variables. You have to add a header to pass the source ip to asterisk. esample append_hf("X-src-ip: $si\r\n") Il 10/10/2013 02.05, bluerain ha scritto: Are you sure? Can you tell my which function call in opensips? I know how to manipulate the core variabl

Re: [OpenSIPS-Users] Opensip as transaprent inbound proxy for asterisk

opensips can add an header with the real IP and asterisk can use that header to know the real IP Il 09/10/2013 17.02, bluerain ha scritto: I've try to search on internet but not much info. I currently have Asterisk server setup to have sip trunk with customers on a "peer" type. This way, no re

Re: [OpenSIPS-Users] calling external command with sudo

are you sure to know how to configure sudo? please post the user used by openSIPS and sudo configuration (sudoers) Il 20/09/2013 19.11, Dragomir Haralambiev ha scritto: Hello everyone, I am trying to execute 'iptables' from opensips in the script, which works if opensips runs as root. However

Re: [OpenSIPS-Users] WebRTC : Integration with opensips.org free VoIP service & Tutorial

, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 07/04/2013 07:35 PM, Stefano Pisani wrote: I tried to call a SIP URI but it do not seems to be working. I used crome. The connection works but it cannot place the call. s Il 04/07/2013 15.55, Vlad Paiu ha scritto: Hello, The free

Re: [OpenSIPS-Users] WebRTC : Integration with opensips.org free VoIP service & Tutorial

I tried to call a SIP URI but it do not seems to be working. I used crome. The connection works but it cannot place the call. s Il 04/07/2013 15.55, Vlad Paiu ha scritto: Hello, The free VoIP service offered by opensips.org has now been enhanced in order to support WebRTC calls. In order to t

Re: [OpenSIPS-Users] uac_replace_to problem

Hi , have you changed to twice? This happens if you try to use uac_replace_to (or uac_replace_from) twice. s Il 19/06/2013 15.18, M.Khaled W Chehab ha scritto: while i am using uac_replace_to in failover route branch i can find that TO header is not changed(sip user part ) but appended an new

Re: [OpenSIPS-Users] OpenSips and Ekiga

Could you get some log using ngrep? regards s Il 04/06/2013 00.50, Alberto Ayala ha scritto: Hello subscribers I'm trying to integrate OpenSips and Ekiga I was able to install using EPEL packages in a CentOS 5.x machine opensips mysql is working. (I create two users for testing purpouse) wh

Re: [OpenSIPS-Users] How to get only the ip from contact?

Use the core variables $(ct.fields(uri){uri.host}) or something like that best regards s Il 29/05/2013 15.43, microx ha scritto: Hi all, An example contact uri of an INVIT looks like "sip:111@61.60.228.221:5060". OpenSIPS seems not to have a specific variable to store the IP part. Is there an

Re: [OpenSIPS-Users] How do I remove an unwanted header w OpenSIPS?

Use remove_hf("X-BroadWorks-DNC:network-address"); to remove the unwanted header s Il 26/04/2013 19.30, Stacy Trippe ha scritto: We are running into a problem with the VPN overhead,

Re: [OpenSIPS-Users] How to protect OpenSIPS from undesidered requests (DoS attack?)

Il 06/03/2013 19:58, leo ha scritto: I've also added Nick's suggestion: if ($ua =~ "friendly-scanner") { xlog("L_ERR", "Attack attempt - Request dropped"); drop(); } But i don't have neither those events in the opensips.log file. it depends where in the

Re: [OpenSIPS-Users] mediaproxy behavior

Use rtpproxy :-) s Il 16/01/2013 09:53, Jorge Ortea ha scritto: Hi all, I use OpenSIPS + Mediaproxy and several asterisk behind. I have the next problem: I would like SIP Proxy with a public IP and the asterisks in private network, but it isn't possible because mediaproxy only can forward R

Re: [OpenSIPS-Users] NAT issues on client and server

Could you explaint your scenario better? The server is in a private network? The cliets are in different private networks? s Il 04/01/2013 05:23, Mark Currie ha scritto: Thanks for the advice Flavio. Currently I am actually pretty close with my NAT'ed OpenSIPS and NAT'ed clients. I am assumi

Re: [OpenSIPS-Users] android native sip client

Hi Nick, I'm using Sipdroid without any issue. s Il 24/12/2012 06:58, Nick Chang ha scritto: Hello Do everyone use android native sip client with opensips?? I try it. I can dial to B phone. B can answer. But it is interrupt immediately. If I change to linphone. It's OK. Thanks Nick _

Re: [OpenSIPS-Users] call duration problem

l-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event); \

Re: [OpenSIPS-Users] call duration problem

Try using this option modparam("acc", "early_media", 0) regards, s Il 24/07/2012 19:55, Francisco Franco ha scritto: Hi, I have opensips 1.6 runing with mediaproxy and have a problem with call duration accounting. The session duration that is stored in database is total time from call st

Re: [OpenSIPS-Users] private domain to IP resolution

Take a look to drouting module Il 18/07/2012 13:59, asd asd ha scritto: Hi Rebecca, thank you for the proposal. The requirement is to keep all the configuration inside opensips so there would be no need to correlate with any other external system. All the proposed solution require an externa

Re: [OpenSIPS-Users] private domain to IP resolution

Just create your own DNS. Simply. s Il 17/07/2012 21:18, asd asd ha scritto: Hi, thanks for an idea, but this would not be manageble for a farm of opensips nodes. I'm looking for a a solution using standard opensips modules or at least a db solution using standard opensips db access functi

[OpenSIPS-Users] ERROR:core:pv_printf: buffer overflow

Hello folks, I get this error using avp_query. How I can increase the buffer size? Jul 12 17:00:28 ks363596 opensips[24980]: ERROR:core:pv_printf: no more space for spec value [75][1192] Jul 12 17:00:28 ks363596 opensips[24980]: ERROR:core:pv_printf: buffer overflow -- increase the buffer size

Re: [OpenSIPS-Users] PROXY behind a NAT

Yes, you need advertised_address. Remember, you cannot have internal and external client. All client MUST be external Il 10/07/2012 21:59, Ignacio Gonzalez ha scritto: So adding this to my configuration file, do I fix the problem? Do I need to set advertised_address or not? 2012/7/10 Ali Pe

Re: [OpenSIPS-Users] Detecting retransmissions

You have to check for same callid, same from/to tag, and same cseq. If you have the same values of a previus record, discard it. s Il 08/04/2011 10:21, Pete Kelly ha scritto: Hi I am performing some database logging actions on responses in a reply route, is it possible to detect if a response

Re: [OpenSIPS-Users] Need ideas to tamper with CSeq

What are you trying to do exactly? s Il 31/03/2011 16:37, Cindy Leung ha scritto: I guess I wasn't being clear enough in the call flow. I assume the CSeq in the CANCEL has to be the same as the second INVITE. 1. Phone sends out INVITE #1, OpenSIPS responds with 401, Phone ACK'd. I believe

Re: [OpenSIPS-Users] Dedicated Presence Service

to do what? s Il 31/03/2011 11:07, Paris Stamatopoulos ha scritto: Anyone who could give a helping hand here? Regards, Paris ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___

Re: [OpenSIPS-Users] Modify a URI

You can do that in several ways. A simple way could be: $ru = "sip:12345#" + $rU + "@" + $rd other ways are possible Il 23/02/2011 19:31, Brian Artigas ha scritto: Can anyone give me a code example how I would take a URI and inject a string between the sip: and the actual number. I found the

Re: [OpenSIPS-Users] OpenSIPS handling B2B features

It's very simple setup a Conference server using OpenSIPS and Asterisk. So use asterisk. Regards, s Il 27/01/2011 17:39, Anca Vamanu ha scritto: Toyima, I am sorry, I don't have experience in setting up conference systems, so I can not make a recommendation. Regards, ___

Re: [OpenSIPS-Users] force_rtp_proxy no longer supported

My experience using the opensips 1.6.3 is that engage_rtp_proxy doesn't work, and force_rtp_proxy works fine, despite the documentation. Il 10/02/2011 14:57, chris ha scritto: Hi, Thanks for that. The new docs mention engage_rtp_proxy but this does not seem to be invoked or described in the mo

Re: [OpenSIPS-Users] How to test if a message is from myself

Hi Dave you could try if ($si == $hdr(X-src-ip)){...} Il 03/02/2011 12:59, Bogdan-Andrei Iancu ha scritto: Hi Dave, Unfortunately does not work with variables. Regards, Bogdan Dave Singer wrote: Wow I missed that one. Thanks. Does that work for PVs so I can test other IPs like one from ano

Re: [OpenSIPS-Users] call forwarding with replace from uri

Hi Jesse, in your script you are replacing from header twice. Double check to your script and delete the second uac_replace_from. This function can be used just once a phone call. ciao s Il 02/02/2011 16:41, Jesse Cloutier ha scritto: Thanks for the answer, I was not very clear in my first email

Re: [OpenSIPS-Users] Lookup contact from user part of RURI

Hi, you could set OpenSIPS to not use domain part of uri, so your issue is solved. stefano Il 02/02/2011 15:30, Nauman Sulaiman ha scritto: Hi, using opensips 1.6.2. I am trying to use the user part of incoming RURI to look up a contact, reason being is full RURI is incorrect, this is due to

[OpenSIPS-Users] RTP Proxy with 2 opensips

Hello, I have this scenario: UAC --> Opensips1 --> Opensips2 + RTPProxy --> Internet Opensips1 is on LAN, Opensips2 is dual homed (LAN + Public IP) and on the same machine there is a RTP Proxy in Bridge Mode. RTP Proxy doesn't work. It uses, as callee IP, the opensips1 IP instead of UAC IP.

Re: [OpenSIPS-Users] Opensips B2B + RTP proxy in bridged mode

Could you post your cfg? regards, s Il 13/12/2010 11:48, beci345 ha scritto: Hello to all, i'm using the Opensips as proxy in multihomed mode (one public IPaddr and one internal), with relaying RTP traffic through RTP proxy in bridged mode: UA --->OpenSipsIP1-OpenSipsIP2 >UA (signalling) UA

[OpenSIPS-Users] opensips cross border configuration

I guys, I'm try to configure opensips 1.6.3 + rtpproxy on a dual homed server with one interface on private network and one on public network. I want to make possibile the calls from inside to outside and viceversa. There is a smart way to do that? I'm going to write a cfg to manually change the

Re: [OpenSIPS-Users] registration to other SIP proxies?

I developed my perl script using Net::SIP to do that (and also to play audio message without asterisk). s Il 02/12/2010 15:24, Erik Dekkers ha scritto: Bogdan, Would it be possible to run sipak from the opensips.cfg script? I'm also looking for something like this. Kind regards, Erik ---

[OpenSIPS-Users] Opensisp behind a Firewall and NAT

Hallo, my architecture needs the Opensips behind a firewall than does static NAT between the private IP of the opensips and the public ip. I would use advertise_host to give correct public IP only for the messages directed outside and not for those directed inside; in different words, if an int

Re: [OpenSIPS-Users] rewritehost() and AVP

use $du = $avp(...) regards, s Il 18/11/2010 10:38, Anton Zagorskiy ha scritto: Thanks, this is work. Can you explain why rewritehost() doesn't accept AVP? Where AVP doesn't work too? WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagor

[OpenSIPS-Users] Two OpenSIPS one database

Hello, I need to divide the load on two opensips (using DNS round robin) I would use only one database (mysql). my worries are about the writing on database. Both opensips can use the db to record the user locations without problems? thanks stefano ___

Re: [OpenSIPS-Users] Two OpenSIPS one database

want. That being said, how are you wanting to round robin? On Tue, Oct 26, 2010 at 3:42 PM, Stefano Pisani mailto:stefano.pis...@omnianet.it>> wrote: Hello, is it possible to share the database beteween two opensips? I would to use round robin to divide the load on m

[OpenSIPS-Users] Two OpenSIPS one database

Hello, is it possible to share the database beteween two opensips? I would to use round robin to divide the load on more servers. Thanks s ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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