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(sorry, posted without subject) 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

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

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

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 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 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

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, 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 guys, I'm new to Asterisk, so I'm hoping someone can guide me :-) Currently, I am having the configuration as follows : PSTN -> Cisco router -> Sip Express Router -> Asterisk Voicemail I'm able to get the part from PSTN to Sip Express Router working, but I can't integrate Asterisk with Sip Express Router (SER). Basically, SER does all the registering and forwarding

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

Hi guys, I currently have a sip proxy server (sip express router) which registers the sip phones. I need to add voice mail capability, i.e. sip express router will forward all incoming calls to Asterisk if the user does not pick up the call in 15 seconds. The voicemail recording stops when the user hangs up. However, the recording does not end if the user presses the # key, i.e. it is ignoring

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, 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

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?

hi list, is anyone succesfully using asterisk with libretel port-of-call (www.libretel.com)? If so, i would be grateful for configs..i set up libretel to forward to mynumber@myserver.com:5070 (asterisk is running on 5070 and SER on 5060) and when i call the number i see SIP messages with ngrep but the asterisk CLI doesn't seem to catch them. I assume i need to register...is this even possible

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

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

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

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

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to