similar to: re: asterisk as VM for SER

hello, if anyone is using asterisk as a voicemail system for SER I would be grateful if i could see a working ser.cfg and extensions.conf of such a setup. I am having some issues with rollover to voicemail when busy, and in setting up a VM extension for users to retrieve their mail without having to enter their own extension. When i get this working i'll write it up clearly for the wiki

hello all, i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to 1.23 requests from SER to asterisk die quietly, no

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. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there

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

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

Guess I got where your confusiion lies.... you DON'T set up the ALL contexts the users will have acces to in sip.conf, in this file you will only set 1 single context. All other contexts you want the users to have acces to, you will have to add them in the extensions.conf So what you have to do is the following: -user 2092, set it the createmenu context in sip .conf - in extensions.conf

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a

Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and

I have SER running with Asterisk, both on seperate servers. If I call another SIP number from my SIP phone SER looks up the phone number to see if it's online. If it's not online it forwards the call to Asterisk. How do I configure the extensions.conf file so that calls being forwarded to Asterisk destined for VoiceMail do not conflict with normal outbound calls destined for the PSTN?

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

I ma new to Asterisk. I'd like to setup * as voicemail system for SER. Let's say I have an phone number registered in ser as 5554321. When somebody dial to ser for this number and nobody answer, the ser will forward the call to asterisk and get into voicemail box 5554321. I already have asterisk up and running with mysql setup for asterisk voicemail. Can somebody show me how to do it? Or

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

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem

Hi, I downloaded the stable branch of asterisk a couple of month ago, and I'm using it as a SIP UA voicemail server with SER, and my setup works fine. I do have a list of phones defined in voicemail.conf, in the sip.conf file I only have the setup of asterisk as a peer registering to ser. The extensions.conf file contain the extensions that link to the voicemail application. This setup is

Hi, I'm currently using SER, and the voicemail system there is not stable, and is lacking IVR. I'm wondering if I could use asterisk as a voicemail system only, where calls will get redirected by ser to asterisk and users will be able to leave a message. Is this setup workable, and have anybody done that ? Thanks. Samy.

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

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };

hello all, i need some help and after trying the wiki i'm even more confused than i was. i'm trying to set up call forwarding and running into problems... i want the most basic call forwarding imaginable. 1. caller dials extension (say, 154) 2. dialplan is updated to forward caller's extension (based on CALLERIDNUM) to voicemail, instead of ringing his endpoint. 3. caller is

hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered