similar to: How to exit from console?

Hi, all I am building a new server. Have installed FC 6 and put in TDM400 card. Checked out latest asteriusk code, run make install in zaptel directory. So far all is fine. Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver is found on the system. Is there are any known problems with FC6? I did not have much trouble running

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

Hi, all I am doing "prrof of concept" system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be

Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by

Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a

I have a number of Polycom phones to setup with my * server. For my initial few phones I hand wrote configs. Does anyone here who uses Polycom phones have some form of management utility for automating their setup? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

G'Day All, Can anyone give me some direction in setting up the TFTP server on my RadHat ES3 box? I did quite a bit of reading, but I think I am more unsure now than before. I found the information nebulous. TFTP is already installed. I am trying to determine where the root directory for the tftp services is located so I can copy the CISCO 7960 firmware files onto it. Thanks.... Ferg

Hi, all I have installed TDM400 card. I can see it is there (lspci). But Asterisk does not find it. phonebox2*CLI> zap show status No Zaptel interface found. I assume that driver is not loaded, but I am sure I have installed it (I compiled zaptel and then re-build asterisk and did make install for both zaptel and asterisk). When asterisk is started I get: Jan 2 06:28:08 WARNING[3473]:

Hi, all I finally got myself one of those SIPURA boxes. It is labeled as Linksys, but this is actually a SP3000 box. Anyway, unit has lots of configuration parameters. Not all are obvious. At the moment it registers against my *, but all the calls I do from analog phone connected to it, go to VoIP channel. As this part is still in testing, I want all the outgoing calls got to PSTN by default

I'm having a problem with a new installation of asterisk 1.2.5 with a digium dual port T1 (span 1 connected to an outside line, and span 2 connected to a CAC access bank I channel bank with 24 fxs ports). When I start Asterisk (either from safe_asterisk or asterisk -vvvc) it will immediately exit after it initializes. It will start the logger, register applications and functions, register

Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP

Hi all, I have installed some Asterisk machine, all with the same problem. My typical configuration is: - Asterisk 1.2.14 (or 1.4.0beta3) - CentOS 4.4 server. The problem is this: When I start Asterisk with the default init script (/etc/init.d/asterisk start) distributed with the source, and kill (or kill -9) Asterisk-pid, then safe_asterisk doesn't correctly work (it dies and not restart

Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3.

Hi, I configured Asterisk to run as "asterisk" user, but I see that a user can anyway get a root sheet using !command in CLI. I understood that it's something related to safe_asterisk and TTY console, but modifying the script safe_asterisk I wasn't able to disable this root access. Can someone help me? Thanks. -------------- next part -------------- An HTML attachment was