search for: mediaproxy

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to di...

...Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients ,SER is running on port 5060 and Asterisk on 5065, here i need to forward pstn calls to asterisk and i am planning to connect asterisk to a Cisco Gateway , when sip client calls to pstn SER will recieve...

...lasso > > CC: serusers@iptel.org > > Asunto: Re: [Serusers] NAT considerations... > > > > > > Giovanni Balasso wrote: > > > > >Just some thoughts based on my experience... > > >After months trying to make everything work using > > rtpproxy-mediaproxy with > > >almost everything accomplished but video, I tried to switch > > to stun solution. > > >All my problems are gone now, I have audio, video, presence > > and instant > > >messages working like a charm. And most important media > > server doesn'...

...use IP tables to only allow connections from the OpenSER systems, but that doesn't always work. When a caller transfers a call, the phones will send a REFER message directly to Asterisk, so all the phones would have to also be in the ip tables allow list. Not an elegent solution. We could run mediaproxy on OpenSER and force all RTP streams back through it. Might work, but it might also break other stuff. We could then configure ip tables to only allow RTP streams from the OpenSER systems. It might be possible to configure OpenSER to perform the logic necessary to make it talk to Asterisk properly...

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

Alex thank you for your response. In this case we are USING INBAND, though I have tried both. Nothing works. Yes ser is configured with mediaproxy. Thank you, -JK JK, In-band or RFC2833 DTMF signaling? Also, unless you have SER configured with a media proxy, the actual "call" is not running through SER. It's a signaling proxy only. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670...

Hello i am trying to use this exmple with SER-0.9.3 but still NATED Clients are not working any other requirement http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper ----------------------------------------------------------- # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters

hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . try to call them !! i'll offer you some money . You can not Call them for some advices ... It's really a bad product don't waste your time to setup it. this enterprise must

...e.com/srvres-MTAuMjE4LjIwLjE0Mzo5ODEy" xmlns:ses="http://www.google.com/session"><ses:candidate name="rtp" address="74.125.155.126" port="19295" username="DPf6ayX6cVfawsYS" preference="1.0" protocol="udp" network="mediaproxy" generation="0" password="" type="relay"/><ses:candidate name="rtp" address="74.125.155.126" port="19294" username="DPf6ayX6cVfawsYS" preference="0.6" protocol="tcp" network="mediaproxy" g...

....168.0.12:5070 | | | I | SIP Phone --------|___| | S | | K | SER running on ... |___| global:61.194.32.77:5060 private:192.168.0.12:5060 ipv6:[2001:268:304:a300::10]:5060 *For NAT solution, MySTUN & MediaProxy is employed* In the above setting, can UA (SIP phones) on global network use conference call of Asterisk? In other words, can SER relay from its global connection to local private network? If possible, what can be the syntax in ser.cfg? Thanks in advance and deep sympathy to Tsunami victims in S...

...s? They claim to be b2bua devices but that's a very wide category and only implies that the media stream passes through it - exactly what can be done with fairly simple OSS stuff. In short, what advantage does such a setup have over, for example, an all-IAX setup, or STUN, or a setup with SER/mediaproxy as a SIP server and asterisk behind it? thanks, yair

Hi, We're deploying a small VoIP solution for a group of teleworkers. Naturally, this exposes us to all sorts of fun, most of which we seem to have working properly. However, some NAT issues are still bugging us and we have noticed that often these situations didn't exist when users were connected directly to our VoIP provider, voiptalk.org. They have something which they call a

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Hi all, Is there a solution to solve this ? ASTERISK 1.2.4 || Internet===SER/OPENSER=====Nat==[private net] || sip agents rtpproxy/mediaproxy Sip agents use SER/OPENSER as an outbound sip proxy and asterisk as a registar server, pbx functions, ... SER/OPENSER look for domains in URI. if domains are handled by SER/OPENSER so requests are forwarded to ASTERISK. If domain in URI is not handled by SER/OPENSER and/or ASTERISK I wis...

...i All, i am stuck in NAT issue on ec2 cloud computing from last 2-3 days , may be some of you are doing setup and integration on cloud. below is my setup details which may help you to suggest me solution. Asterisk version : 1.6.2.6 1) Kamailio server having public_ip as well local ip .i am using mediaproxy [also tried rtpproxy] . 2) Asterisk server having public_ip as well local ip. setup: *UAC -----> KAMAILIO -----> ASTERISK* UAC registered to kamailio registration is successful. once it dial PSTN number i forwarded a call to asterisk server and then is created problem because i am not g...

Hi Guys, Soon, I'll be starting a new section related to Asterisk (around 4 years of full time experience with Asterisk, Trixbox, SER, OpenSer, MediaProxy, AGI*) so let me know if you like to see some topic coming. Cheers Arun -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110220/3a4f158f/attachment.htm>

I'm chasing down some DTMF interop issues would like to hopefully rule out Asterisk in the following configuration: RTP path is: Linux/PC/Mac SIP clients -> [MediaProxy as needed] -> Asterisk 1.8.7 -> SIP termination provider(s) DTMF is strictly RFC2833 with no in-band. Asterisk stays in the media path for application reasons and is "Locally bridging SIP/foo and SIP/bar" Asterisk is NOT configured to act on any DTMF while bridging the call (no o...

...servers behind a SER server (in a local ip range). The SER server is also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER server? If not, how will the reach the * server, since they're only reachable VIA the SER router. Here's is scheme: ----- IP Phone A (Behind NAT router) (ext 100, Ast...

..."/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/nathelper.so" #loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/maxfwd.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db....

Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _________________________________________________________________ The