similar to: Sticky Problem SER/Asterisk

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

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, 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

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

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?

Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in

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>

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

Hi, we a have a SER (OpenSER) in front of 2 real-time Asterisk. SER simply forward SIP messages to 1 of the Asterisks: UA --> SER --> Asterisk We have a problem with REGISTERs: Asterisk answers with 200 OK, but changes the Contact header, inserting the IP of SER instead of the original IP (the IP of the UA). It seems that performs a sort of NAT-traversal, but all the elements are on

Hi Friends, I want to setup multiple SIP accounts. How can I do this? I have installed Asterisk, created Asterisk SIP extensions and registered in www.sipgate.co.uk. Now, what I have to do? 1) Am I need to install SER or OpenSER in my server along with Asterisk? 2) If yes, can you please recommond SER or OpenSER? 3) I searched in Internet. But, I didn't find good tutorial for this. Can you

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in

I'm reading some information that recommends using SER / OpenSER for large installation to offload SIP traffic from the Asterisk server. http://www.voip-info.org/wiki/view/Asterisk+at+large However, it looks like the information might be dated. I'm looking at a potential 750 SIP phone and 150 Analog installation, all internal network, PRI trunks, and am trying to nail down an

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

I am curious if it is advisable to use implement Asterisk as a Session Border Controller for a VoIP reseller environment. Users will terminate calls SIP to my server, which will authenticate them via RADIUS, perform a LCR lookup, select an appropriate trunk (based on LCR), and terminate the call (update RADIUS accounting at end of call). All while acting as a B2BUA to prevent the users from seeing

HI, I'm not able to find some good doc or manual regarding Integration of Asterisk with SER. Bacially, I want to forward my calls from SER to asterisk. If some one already done this please guide me. thanks in advance arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently I"m using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call