Messages

Hi,

Yes, I too figured out that the prefix was the problem. If I set the prefix to anything else than 0, I can place international calls.

regards

Hi,

I have trouble placing international calls. If I connect to asterisk and set verbosity to 3 I get the following output when placing the call

Code Select Expand

  == Using SIP RTP CoS mark 5
[2014-03-24 20:18:07] NOTICE[2009]: pbx_realtime.c:371 realtime_exec: No such application 'Meetme' for extension '000NNNNNNNNNNN' in context 'from-lmce-custom'
  == Spawn extension (from-internal, 000NNNNNNNNNNN, 1) exited non-zero on 'SIP/202-00000029'
    -- Executing [h@from-internal:1] Macro("SIP/202-00000029", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/202-00000029", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/202-00000029", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/202-00000029", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/202-00000029", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/202-00000029", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/202-00000029", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/202-00000029' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/202-00000029'

the first 0 is the prefix I set for the phone line and the following 00 is my normal international prefix. NNNNNNNNN is the phone number (including country code).

I place the call from a SIP phone connected directly to the lmce subnet. National calls work both fix and mobile. I haven't done enough testing to say for sure that the problem is related to international calls only, but it seems like it. Maybe the three 0's mean something to asterisk, so it does something I did not intend?

Any solutions or further troubleshooting ideas are highly appreciated!

Regards
Rikard

Setting a prefix does not change anything. Outgoing calls are still prepended with my area code. I call can dial local calls, but all other calls won't work. 

OK. Not sure I understand. Outgoing calls actually work. The problem is that my area code is prepended to the number I dial which means I always call someone else than I wanted...

What should I put in the prefix field?

Why?   

I didn't do anything special when I configured my SIP provider. See attached picture.

Yes, I have looked at it. I just couldn't find the dial rules in there. I need some help to orient myself in the db. Do you know of any documentation? Couldn't find anything on the wiki.

I need my problem solved that asterisk prepends my area code to all outgoing calls. This prevents me from making outgoing calls to e.g. cell phones. My guess is that this is because of some misconfiguration of the dial rules. The phone line wizard just can't solve this.

Yes, I saw that one. It gives me the possibility to enter prefix (which I have left blank) and emergency numbers (which I filled in).

This is somewhat limited functionality compared to what the dial rule language has to offer. Or did I miss something?

Regads

I made a fresh install of 10.04 which worked like a charm (thanks l3mce!).

I have a problem with outgoing calls. For some reason my area code is prepended to all numbers I dial, which makes outgoing calls impossible. I don't know how the system figured out my are code and why it decided to prepend it.

Since freepbx is no longer available I have no idea how to fix this problem. I would guess I have to fiddle with either the dial patterns or dial rules, but I do not know how to do that. I have digged around in the asterisk mysql database without really finding my way.

Advice would be appreciated! Maybe someone has a link that describes the asterisk real time database? I have searched the wiki without result.

Regards

Hi all,

I am running WMP11 on Vista. It shows the myth tv UPnP server but not the LMCE media tomb server. I can see it in the network neighborhood and access the web interface, but WMP won't connect to it.

I am running LMCE 8.10 on my core. I tried searching in this forum, in media tomb forum and in WMP forum. No real success...

Someone else had this problem?

Regards

Yep, I can now see that it works. Thanks a lot. However, the asterisk security kind of bothers me. Did you read this?

http://forums.asterisk.org/viewtopic.php?p=159984

Seems like all extensions created by lmce is of type friend. Looks like an unnecessary security risk. I changed them to peer. The system still works and now it should supposedly be more secure. (However not 100%).

This little lesson has taught me that what is installed default in lmce is a real security nightmare...

Hi all

When I receive a call on my MD during media playback, the soft phone does not produce any sound. There is no ring sound, and if I answer the call I can not hear the other person. If there is no media playing, the sound works fine. I have looked in the logs for the relevant devices, without finding any errors.

Does anybody have any advice for trouble shooting?

I am running 8.10 with latest updates.

regards

Thanks for all advice guys! The intrusion was done in my asterisk server, hence I am responsible. A peek in the asterisk logs confirmed that it was a brute force attack, fail2ban should solve this. I followed the instruction on the wiki provided by pw44. To test the asterisk jail I tried to register to an extension with a SIP soft phone on a computer in my local network. I registered three times with incorrect password. The attempts were correctly logged in the asterisk log, but looking in the fail2ban log I could see that the ban did not kick in. Do I have trouble shooting to do, or is there an explanation to this? (I did not include the computers IP to the ignore list).

regards

Or like me, who have two voip services:
1) sipgate, with no credit, only to receive calls,
2) voipcheap, with € 10,00 credit, to place calls.
If someone is able, bypassing fail2ban and firewall to place calls, it will stop in € 10,00
But even € 10,00 i'm not willing to give away to some jerk, so fail2ban, firewall and strong and log sip extension passwords are in use.

You removed 5060 rules, but did you block incoming traffic from outside to this port?

I thought that was the purpose of the firewall rule itself. How do I do that?