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Trying to find a solution to a sticky problem here. We have 3 OpenSER systems. Phones register with the OpenSER systems, and after they authenticate the user, pass the registration info using OpenSER's send() command to all Asterisk boxes sitting behind them. Each asterisk system then knows about every phone. For this to work, I had to turn off authentication in Asterisk for both

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 dimensioning this system to work

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

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

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

Hello Masters 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

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip

I'm taking the liberty to announce this event on the Asterisk mailing list, as Asterisk and OpenSER form a valuable combination in SIP architectures. The second edition of OpenSER Summit will take place in San Jose, USA ,on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of

Hi, I am trying to integrate asterisk with openser for a simple call. I am facing some issues with Asterisk. Below is the explanation: I have a UA1 sending invite to UA2 through Openser and Asterisk with the below sequence. Sequence is UA1->OpenSER->Asterisk->Openser->UA2 When Asterisk gets the INVITE, the INVITE contains two Via headers, one of the UA1 and

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

Hi, Using openser 1.1.0-dev8 as a registrar/proxy in from of Asterisk. Recently I have been getting errors from Asterisk due to corrupted From: headers, which appear to be caused by uac_replace. Here is a section of the debug log: Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: DBG:uac::restore_from_reply: removing <From: <sip:lenc_domain.com@sip.domain.com>;tag=635c3ce6 > Mar

Hi, does anybody know if asterisk can authenticate on a NOTIFY send to a peer. I use OpenSER as SIP-Proxy and asterisk as voicemail system with ODBC Support for voicemail-messages, voicemail-users and sip-peers/users. My SIP-Users register with OpenSER. Asterisk has two views on the OpenSER database for voicemail-users and sip-peers. Call-Routing, leaving voicemail-messages in database an MWI

Hi All, I am stuck with an issue in the Openser+Asterisk Forking. In this solution we are using Openser as the Registrar. Hence it will store all the contact bindings along with the q values for a given user, say ua1. The current setup is such that the INVITEs are sent to Asterisk by Openser and Asterisk sends out the INVITE. Now if ua1 is registered with two different contacts having

Sorry if this hit the list twice, sent out yesterday, but didn't see it show up. Hi All, Can Asterisk be used as a SIP proxy, blah, blah, blah??? I've glanced over questions like this through the years, with a good idea on what a SIP proxy is and what Asterisk is and IS NOT. I never really took the time to lab-up SER and test drive it to see what advantages might be gained from using

Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based proxy / call routing setup? I need to get simple CDRs; not for detailed settlement/rating, but just for reconciliation with an ultimate TDM carrier just to make sure we only get billed for what we're actually using. I'd use the often-heralded approach of dumping a call from OpenSER into Asterisk and having it

This is a SER/Polycom question, but I hoped we may have some SER guru's here... I have a series of Polycom phones that are tying to register with OpenSER. The phone sends a REGISTER message, and OpenSER replies with Unauthorised (all normal). The phone re-sends the REGISTER with the credentials, and OpenSER sends Ok. Here's where it goes downhill. The polycom's appearance display

Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC ----> openSER - - - ->

Hi! Meanwhile also the location is fixed: it is happening at metalab (http://metalab.at/) - a place for geeks. Thus, we meet there at Thursday, 20.9.2007, 19:00 CEST (=local Vienna time). Metalab is located next to the city hall: http://metalab.at/wiki/Lage Metalab is no pub/restaurant. Thus, don't come hungry! Nevertheless liquid food (drinks) is available. We meet in the library (in

I've just finish setting up OPENSER with Asterisk 1.2.2 In OPENSER, i have set extension 400 to push to asterisk, which in turn run apps VoicemailMain() I noticed after the INVITE came to asterisk, it reply to OPENSER with " We're at 203.125.68.66 port 16520 ". Right after that , it will keep on " Retransmitting #1 (no NAT) to 203.125.68.66:5060: " , all the way until