similar to: SER/OpenSER, I Finally Get It.............General Observation

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>

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

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?

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

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

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

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

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'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

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

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'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:

Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because "OpenSER does only signalling while Asterisk does all". My question is: If Asterisk also does only signalling

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

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. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

Hey, You know, the Digium guys said both are good. They said the the DNS method is better because you dont have the extra point of failure (SER) but said the SER method is better in that it gives you more exact control in the handling of the calls and registration. They did acknowledge there would be a possible downtime only for incoming calls to users with dynamic IPs if the

Does any one know if there's an mechanism (internal to asterisk or otherwise) to replicate dynamic SIP device registrations across a pool of asterisk servers? I'm in the process of creating a asterisk cluster using a SIP hardware load balancer and so far this is one of the challenges I'm facing. One thought I'm currently investigating is to use openSER to intercept and

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)