Problem with my asterisk1.4

please i have a serious problem with asterisk and i don’t understand why. The problem is that i can’t speak through the LAN with another user. The softphone that i use is xlite i saw this in the log files

[Sep 14 15:51:45] WARNING[20041] app_voicemail.c: No entry in voicemail config file for '1001' [Sep 14 15:52:03] WARNING[20048] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 14 15:52:03] WARNING[20048] app_voicemail.c: No entry in voicemail config file for '1001' [Sep 14 15:52:14] WARNING[20055] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 14 15:52:14] WARNING[20055] app_voicemail.c: No entry in voicemail config file for '1001' [Sep 14 15:52:27] WARNING[20060] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 14 15:52:27] WARNING[20060] app_voicemail.c: No entry in voicemail config file for '1001' [Sep 14 15:53:11] WARNING[20079] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 14 15:53:11] WARNING[20079] app_voicemail.c: No entry in voicemail config file for '1002' [Sep 14 21:24:29] ERROR[3353] asterisk.c: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. You have options: 1. You only have to compile Zaptel support into Asterisk if you need it. One option is to recompile without Zaptel support. 2. You only have to load Zaptel drivers if you want to take advantage of Zaptel services. One option is to unload Zaptel modules if you don't need them. 3. If you need Zaptel services, you must correctly configure Zaptel.
please explain me the problem and try to give me solutions…
I’m waititng[/code]

You have more than one problem. I count at least three.

You have a problem in sip.conf, or extensions.conf, or maybe no sip.conf at all.

You have a problem in voicemail.config (or extensions.conf) and probably no voicemail.config at all.

You have a problem with zaptel, and should, in any case, for a new installation, be using dahdi.

We’d need at least verbosity 3 output, and almost certainly will need the contents of the relevant configuration files.

If your firewall isn’t too extreme, I would suggest installing the sample configuration and getting to understand how it works. Also, if you haven’t already done so, you should read Asterisk: The Future of Telepony.

please i want to know if o am oblige to install and configure zaptel if i use asterisk only to communicate in a LAN?

You need Dahdi (you should not be using zaptel on a new installation) if you use meetme conferences, or you use a phone that uses silence suppression.

meetme won’t work at all without a low level driver to provide timing. Phones with silence suppression, may not hear tones and recorded announcements without it.

i have installed and configured very successfully with two users. We are using x-lite as softphone but when qe talk through the phone each other ear as the fog when i kooked the log i saw this

[Sep 16 12:50:10] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' [Sep 16 12:50:20] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' [Sep 16 12:50:30] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' [Sep 16 12:50:40] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' [Sep 16 12:50:50] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100'
this is the configuration of my sip.conf

[code]general]
[1001]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw

[1002]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
[/code]
i look forward to read you soon

This doesn’t make sense. Even allowing for typos and strange English, I still can’t guess all of it.

Best guess is:

…we talk through the phone to each other we hear (ungessable metaphor); when I looked at the log…

Try explaining very simply and also repeat in your own language, in case someone can translate it.

I’m tempted to ignore the unknown codec messages, for the moment, although the fix would be on the phone. I don’t get them with X-Lite 3.0 build 41150.

i will try excuse i don’t know English well the problem is that i don’t understand very well when i am speaking with a person through the phone there are sounds like fog idon’t understand very well. i think it deals with codec look at files in the previous messages

“sounds like fog” is not a normal metaphor, so can you decribe the sound more precisely.

Poor sound quality is normally the result of a poor network connection (high delay, variable delay, or high packet loss). Not having a timing source (no dahdi/zaptel) can cause breaks in the sound if the phone uses silence suppression.

Poor sound quality may also result from trying to run Asterisk on a virtual machine.

Mu-law is a simple, standard, codec that is used throughout the traditional phone system in North America.

please i want you just to tell me more about the messages below please

[Sep 16 12:50:10] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' [Sep 16 12:50:20] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' [Sep 16 12:50:30] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' [Sep 16 12:50:40] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' [Sep 16 12:50:50] NOTICE[7304]: rtp.c:1334 ast_rtp_read: Unknown RTP codec 126 r eceived from '192.168.10.100' i’m waiting

Please supply sip set debug and rtp debug output Warning. The latter will be huge. Make sure you include an entry that has codec type 126. For the SIP, you need to provide all the SDP and the debug output associated with processing the SDP, as a minimum.

However, I don’t believe these messages are important. They are only appearing every 10 seconds, and problems with mis-matched codecs show up in SDP (in the SIP packets) not in RTP.

Also see http://forums.digium.com/viewtopic.php?p=45558&sid=6cf39735bcac4a03e5738a36267f6960

ok thanks but your english is very high i don’t understand welle what you are saying

Please:

  1. configure Asterisk to record debugging output;

  2. enter the command “sip set debug on” on the CLI;

  3. reproduce the problem;

  4. Paste the output produced into this forum. It will make the posting smaller if you only include the relevant parts. The relevant parts are those relating to SDP.

  5. enter the command “rtp set debug on”;

  6. reproduce the problem;

  7. find RTP debugging output showing codec 126;

8 ) paste an example of just that output into the forum.

Or:

Don’t worry. It is very unlikely that this is your problem.