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Hi, I would like to use Asterisk as VoiceMail system ... the only issue I have is with DTMF recognition. Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ... inband : works, but has some glitch ... not always good ... don't know why. rfc2833 : doesn't seem to work .. info : said to be not working

Hi, I 've a proxy on my network where some calls are routed to .... And as well some extensions on my Asterisk Server. What I would like to do is to accept all incoming calls from the proxy, wherever they are coming from or going to ... but, as soon as I receive a call with the same number as one extension defined in Asterisk (but through the Proxy !) , it refuses the call, saying that there

Hi, I would like to return different values/cause to another SIP Server with Hangup cmd. I tried to put different values in Hangup(xx) ... but it always returns the same value ! How can I send back different error cause ? Thanks, Jean-Marc -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

I 'm wondering ... I have tried to use Asterisk external IP for some times ... but it never affects the VIA SIP Field .... Is It normal ? When reading many books in SIP, this should be external IP, no ? Did somebody manage to put/force the external IP in this VIA header ? If yes, how ? If not, any ideas how to reach this goal? Thanks, JM -------------- next part -------------- An HTML

Hi, My case is a little bit complicated. I would like to use my Asterisk Box for 2 different services/providers : - Voicemail server for one - SIP Registrar and Proxy for some other extensions The problem is that Voicemail service is for another provider which has defined Extension like ABC ... We are connected to them through a SIP Trunk. Everything works fine .... Except IF ABC is also defined

Question is quite easy: How am I supposed to configure Asteirsk to have the same extension, in 2 differents domains. In the general section of sip.conf, I add the domains, But how to say to Asterisk : user1@domain1 > Pasword1 user2@domain2 > Pasword2 Thanks for your help !!!!! JM -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have tried to use the Record Command in Asterisk, Here is the configuration : exten => record,1,Answer ... exten => record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV) exten => record,n,Playback(vm-goodbye) exten => record,n,system(/usr/local/bin/send-recording.pl --to ${EMAILADDR} --file /var/spool/asterisk/record/${CALLFILENAME}.WAV) exten => record,n,Hangup

Hi, I use in Production : Asterisk 1.2.9.1 We Use Asterisk as a SIP Transit Server to record centrally all the calls. The call flow would be: incoming calls : PSTN -> GW -SIP-> Asterisk(Record) -SIP-> Softswitch -> IP Phone outgoing calls : IP Phone -> Softswitch -SIP-> Asterisk(Record) -SIP-> GW -> PSTN Dial plan in Asterisk is quite simple: [record] exten =>

Hi, I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages? I tried to use the "minmessage", but, couldn't. Is that the way ? I was thinking of using the "h" Dialplan, and launch some script, but

Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages? I tried to use the "minmessage", but, couldn't. Is

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

Hi I was trying to have realtime voicemail working with ODBC Driver. Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page ) But I do not manage to make it work with ODBC. Outside Asterisk, ODBC works fine, I can access my databases & tables ! Asterisk fails to