Hi Guys, My name is Marcias, and I am setting up for the first time an Asterisk PBX, I am learning as I go along. I have been able to download and install Asterisk, Libpri and I have been able to get Asterisk up and running. I have several questions: 1 .I can call the Asterisk server from my Xten phone and it picks up. I have 3 computers (one of them being the asterisk server) I can seem to call from one computer to the other through the Asterisk server (All this is local within my network) but as soon as I pick up I cant hear anything on the other side. ANy ideas of whay this would be happening? 2. On another note, I have a website that Steve, was kind enough to re-direct me to. www.onlamp.com/lpt/a/3956 . Very nice website on how to setup everything, but it states that I am supposed to have an addmailbox script under /usr/src/asterisk. I don't seem to have this? So when I call in to listen to the voice mail that i have left I get the server informing me that I have voicemail, but I can't seem to listen to them. So I am not sure if I am missing something in the setup of the voicemail. 3. My next step is to learn how to change the welcome greetings and and then eventually how to hook up my server to another server or to hook a FWD number to it? Thanks for your help in advance, Marcias email: martinez@aei.ca FWD: 260032 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040404/31319f01/attachment.htm
I am only just starting out with * myself, but believe it or not I had the same problems not more than a couple of days ago. 1) With the X-Lite clients I was able to connect a call amongst them, but unable to hear a thing. (Same problem I suspect). The problem ended up being that the * server was not sending which audio protocol back to the client. (It was only send the DTMF protocol, which means that if you hit a number it would be heard.) I added the following lines to my sip.conf file and everything worked properly: [general] ; normal general settings go here disallow=all allow=ulaw allow=alaw allow=gsm 2) The scripts have been moved to the /usr/src/asterisk/contrib/scripts/ subdirectory. Once you run the script it will prompt you for the context, which I have left blank, and the extension. 3) I don't know because I haven't gotten that far. Hope this helps, Robert Jackson -----Original Message----- From: Marcias Martinez [mailto:martinez@aei.ca] Sent: Sunday, April 04, 2004 5:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Please help Hi Guys, My name is Marcias, and I am setting up for the first time an Asterisk PBX, I am learning as I go along. I have been able to download and install Asterisk, Libpri and I have been able to get Asterisk up and running. I have several questions: 1 .I can call the Asterisk server from my Xten phone and it picks up. I have 3 computers (one of them being the asterisk server) I can seem to call from one computer to the other through the Asterisk server (All this is local within my network) but as soon as I pick up I cant hear anything on the other side. ANy ideas of whay this would be happening? 2. On another note, I have a website that Steve, was kind enough to re-direct me to. www.onlamp.com/lpt/a/3956 . Very nice website on how to setup everything, but it states that I am supposed to have an addmailbox script under /usr/src/asterisk. I don't seem to have this? So when I call in to listen to the voice mail that i have left I get the server informing me that I have voicemail, but I can't seem to listen to them. So I am not sure if I am missing something in the setup of the voicemail. 3. My next step is to learn how to change the welcome greetings and and then eventually how to hook up my server to another server or to hook a FWD number to it? Thanks for your help in advance, Marcias email: martinez@aei.ca FWD: 260032
I am facing problem in playing a wav or gsm file on asterisk. The error i
get whenever i tried is
*CLI> -- Executing BackGround("SIP/1235-98f6",
"/etc/asterisk-1.2.0/sounds/vm-goodbye.gsm") in new stack
Jan 13 20:08:57 WARNING[6181]: file.c:508 ast_openstream_full: File
/etc/asterisk-1.2.0/sounds/vm-goodbye.gsm does not exist in any format
Jan 13 20:08:57 WARNING[6181]: file.c:820 ast_streamfile: Unable to open
/etc/asterisk-1.2.0/sounds/vm-goodbye.gsm (format ulaw): No such file or
directory
Jan 13 20:08:57 WARNING[6181]: pbx.c:5747 pbx_builtin_background:
ast_streamfile failed on SIP/1235-98f6 for
/etc/asterisk-1.2.0/sounds/vm-goodbye.gsm
== Auto fallthrough, channel 'SIP/1235-98f6' status is
'UNKNOWN'
-- Saved useragent "ZyXEL P2000W VoIP Wi-Fi Phone" for peer 1235
Can somebody help out
Hello all,
Iam using this Asterisk server since three weeks and i have to clarify some
thing about Asterisk
here is my problem Iam trying to use my Asterisk as a
gateway to pstn and SER as a proxy and redirection server so,here in SER i
had added three or four users by using the command serctl add ,and whem i
check in mysql database i can view these list of people which i registered
here in SER
so do i need those people to register again in ASterisk or the ser
just look in the database and make rtp sessions when call is being made .Iam
not able to get the point here clearly
If for example i want to forward only pstn calls to asterisk and
remaining all sip sessions will made by SER .just configuring in SER works,
because asterisk is non stateless server and we will register peers using
domain as well as ip address but in SER we will register peer only by giving
serctl add name password e-mail
but there is no ip address to bind ,so here caller can call from any place
using the username and password it works? or not.?
and if i want to add more than name password
e-mail i.e like
username etc.. how i have to enter in to the database or is there serctl
command to make this work ....
May be my doubts here very fun to most of professionals in
Asterisk even then please try to help me .. so that I can move further in
ASTERISK and SER
integration with purpose .
Thank You.
Regards ,
Ravi.
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