search for: ser

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? whic...

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...

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 duplicated effort and lack of specialised response. T...

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...

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...

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-v...

...erase R folder installed on my /usr/lib and went to synaptic and uninstall all the cran files. The idea was to start over again. But I try to install by apt and appears: Os pacotes a seguir t?m depend?ncias desencontradas: r-base: Depende: r-base-core (>= 2.7.0-1gutsy0) mas 2.5.1-1 est? para ser instalado r-recommended: Depende: r-base-core (>= 2.7.0-1gutsy0) mas 2.5.1-1 est? para ser instalado Depende: r-cran-boot (>= 1.2.19) mas n?o vai ser instalado Depende: r-cran-cluster (>= 1.9.6-2) mas n?o vai ser instalado Depende: r-cra...

Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone -> SER -> ASTERISK -> SER -> PSTN 2) sipphone -> SER ->ASTERISK ->PSTN on the first option i am trying to return the call to the ser af...

Hi, I am puzzled with serialize(). It comes down generating identical hash codes for (apparently) identical objects using digest::digest(), which in turn relies on serialize(). Here is an example illustration the issue: ser <- function(object, ...) { list( names = names(object), namesRaw = charToRaw(names(ob...

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 tell Asterisk to forward...

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...

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and clients are registering successfully. However I want clients to authenticate before they...

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why...

Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP server....

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 Message ----- From: braincrew.com To: serusers@iptel.org ; asterisk-users@lists.d...

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated...

...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 ,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 t...

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)....

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 Med...

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...