similar to: Asterisk and SER Integration together

Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, I'm not familiar with network configuration really, although I do understand most of the basics. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version:

Hi, I've been registering asterisk to ser. I'm using SER as the outbound SIP trunk for Asterisk. Users registered with Asterisk will use the SIP trunk to reach SER registered users and PSTN's. Now when I register Asterisk with SER, on my SER's location table I see these record: Username Column = asterisk Contact Column = sip:s@202.84.24.47 I have a script running that checks

Hi all, I would like to know if it is possible to use asterisk and ser together in a single computer system using ser as a sip proxy and forwarding any voice call request to asterisk for calling into the pstn gateway. (or any other alternative that is possible is also welcomed for suggestions). If it is possible can someone kindly show me the necessary configuration files or refer me to any page

Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea?

I have seen a lot of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I

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

Hello! Does anybody know of a way to access the Request URI in a SIP message? I've got the following problem/scenario: We have a SIP Proxy (SER) wich forwards SIP-messages for non-IP destinations to our Asterisk. There is no authentication done between Asterisk and SER. I've configured Asterisk to accept any request for a PSTN-line from SER's IP-address. Since we allow IP-to-IP

Hi, I'm trying to use Asterisk as a SUA which would allow me to register on a SER running on the same machine and to send simple test messages through the CLI to another SUA located on another computer, registered on a SER running also on this other computer. I need the CLI because I have to work remotely. Is Asterisk the good choice? What kind of sip.conf and extensions.conf do I need?

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of

Hey gang, I'm trying to work out all possible scenarios using SER & Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle.

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

Dear All, I have the following setup. Quad T1's<->Asterisk (PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet) | Local US Help Desk (Snom 200') This setup works well. I can pass calls from over the internet to the Asterisk PBX via SER using X-Ten Lit. I have a couple of questions; 1. How do I

Hello, This is the scenario i want to setup: Cisco ATA 186 -----------> SER -------------> Asterisk I want the Cisco ATA to register to Asterisk through SER. when the Cisco ATA place a call, SER querry a data base (MySQL or else), and if there is a valid Account for the ATA, the call go to Asterisk. Did someone know how to set SER to work like this with Asterisk? which version of SER

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

Hello, I have several * 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

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

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, I was wondering if someone could explain the relationship of Asterisk, SER & Jabber (XMPP) to me. I understand that there are facilities within Asterisk to use jabber to notify of incoming calls via XMPP clients, however I'm trying to work out exactly where the SER server would sit in all of this and what it's actual role would be. This might seem like a silly

Hello: I noticed in few references that asterisk and ser and complementary. Meaning asterisk handles connections to PSTN and voicemail but SER is better for routing SIP traffic. Is anyone using just asterisk for production purpose. Meaning serving a high number of callers. Is it mandatory to use SER behind asterisk? your feedback would appreciated. -Gaid -------------- next part

I have some simple questions and i need your help guys. I have ser server which working fine, between users. I am trying to add some more features to the ser. Most important is the IVR. I installed Asterisk and i am trying to register user in asterisk with no success. Part of ser.cfg file where i am trying to redirect the call to the asterisk.