A1.6-b9 can't make calls?

[mystere]

Ok, I setup 1.2 today and decided to try out 1.6 to test TLS support. The TLS seemed to work fine (phone registered) but I couldn't make any calls.

After messing with it for a while, I couldn't get it to register standard sip phones at all, so I reinstalled completely (reformat) then tried to just make standard SIP work. Phone registers, but trying to make any calls (*43, *60, *65) all come back as "404 Not Found".

Any idea what could be the problem or where to start looking?

I followed Nerdvittles setup (everything except password changes), and I have no trunks or routes setup. Just trying to get the the extension to work. I've tried X-Line, Zoiper, and even went and bought EyeBeam (to test TLS).

I've turned off the firewall, and the system is pretty basic vanilla with no modifications other than updating to the lastest FreePBX and adding in AsteriskInfo.

There's not a lot that could be wrong here.

 

[mystere]

Nobody has any suggestions? I'm at a loss as to where to look.

This is all there is in the log

{May 17 15:57:26] NOTICE[4659] chan_sip.c: Peer '20' is now Reachable. (108ms / 2000ms)
[May 17 15:57:26] NOTICE[4659] chan_sip.c: Peer '10' is now Reachable. (10ms / 2000ms)
[May 17 15:58:07] NOTICE[4659] chan_sip.c: Call from '10' to extension '*43' rejected because extension not found.

The extensions appear to be there i extensions_additional.conf, also notice above it says both 10 and 20 are reachable, but I can't call one from the other either, same response, 404 not found and "rejected because extension not found".

Both extensions have from-internal as their context.

pbx*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
20/20 10.0.0.10 D 8170 OK (20 ms)
10/10 10.0.0.10 D 7648 OK (12 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

[10]
type=friend
secret=test
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=never
mailbox=10@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/10
context=from-internal
canreinvite=no
callgroup=
callerid=device <10>
accountcode=
call-limit=50

[20]
type=friend
secret=test
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=never
mailbox=20@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/20
context=from-internal
canreinvite=no
callgroup=
callerid=device <20>
accountcode=
call-limit=50

 

[mystere]

I know 1.6 is beta and is likely to be unstable, but I should be able to at least get an extension to work. That's pretty basic functionality, isn't it?

 

[darmock]

First you should try using extensions that are 4 digits long. (humor me here)

Second how are you creating the extentions? In freepbx or by hand?

Third does this number not seem to be excessive?

10ms / 2000ms


In the last 2 beta releases I have had no problems with making calls


Tom

 

[darmock]

I know 1.6 is beta and is likely to be unstable, but I should be able to at least get an extension to work. That's pretty basic functionality, isn't it?

do the following

1. Log into the CLI as user root
2. cd /usr/src/asterisk/configs
3. cp amd.conf.sample /etc/asterisk/amd.conf
4. chmod 664 /etc/asterisk/amd.conf
5. chown asterisk:asterisk /etc/asterisk/amd.conf

6. amportal restart
7. check to see if it is working asterisk -rvvvvvvv
8. sip show peers

Not sure why that is not being installed by asterisk as it really should be. In earlier versions of asterisk if a conf file was not present builtin defaults would take over. Not anymore it seems. Sip won't load into asterisk without this file being present.; I am modifying the load file for 1.6 to ensure it is installed.


Tom

 

[mystere]

I'm not sure what the second time is in 10ms / 2000ms. 10ms doesn't seem excessive, but the 2000ms seems like it might just be a timeout value or something?

I've done what you suggested. Still no difference. I've changed to extension number 1001. No difference. Returns "not found" for any extensions, *43, *60, *65, other registered extensions, etc..

Name/username Host Dyn Nat ACL Port Status
1001/1001 10.0.0.10 D N 17626 OK (106 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]

SIP seems to be loaded, as i can perform SIP commands, and the phone is registered, it just returns "not found" for any extension dialed.

Not sure why it can't find even the speak your extension. context is from-internal and all the extension*.conf files look correct.

 

[mystere]

Ok, I got it working finally.

I setup a special error log file, and checked that. It had a number of files that were #included but it couldn't find. I just touched them to create empty documents.

These files were:

[May 17 15:57:26] ERROR[4659] config.c: The file 'features_general_custom.conf' was listed as a #include but it does not exist.
[May 17 15:57:26] ERROR[4659] config.c: The file 'musiconhold_custom.conf' was listed as a #include but it does not exist.
[May 17 15:57:26] ERROR[4659] config.c: The file 'meetme_additional.conf' was listed as a #include but it does not exist.
[May 17 15:57:26] ERROR[4659] config.c: The file 'queues_general_additional.conf' was listed as a #include but it does not exist.
[May 17 15:57:26] ERROR[4659] config.c: The file 'extensions_override_freepbx.conf' was listed as a #include but it does not exist.
[May 18 22:42:49] ERROR[32103] config.c: The file 'features_applicationmap_custom.conf' was listed as a #include but it does not exist.
[May 18 22:42:49] ERROR[32103] config.c: The file 'queues_custom_general.conf' was listed as a #include but it does not exist.
[May 18 22:42:49] ERROR[32103] config.c: The file 'globals_custom.conf' was listed as a #include but it does not exist.
[May 18 22:42:49] ERROR[32103] config.c: The file 'extensions_additional.conf' was listed as a #include but it does not exist.
[May 18 22:43:31] ERROR[32260] config.c: The file 'features_applicationmap_custom.conf' was listed as a #include but it does not exist.
[May 18 22:44:50] ERROR[32422] config.c: The file 'queues_custom.conf' was listed as a #include but it does not exist.
[May 18 22:45:36] ERROR[32582] config.c: The file 'queues_post_custom.conf' was listed as a #include but it does not exist.

I'm also getting this when restarting amportal, which doesn't seem to affect anything yet:

STARTING FOP SERVER
/etc/profile: line 30: /dev/null: Permission denied
/etc/profile.d/colorls.sh: line 3: /dev/null: Permission denied
/etc/profile.d/colorls.sh: line 4: /dev/null: Permission denied
/etc/profile.d/colorls.sh: line 17: /dev/null: Permission denied
/etc/profile.d/colorls.sh: line 18: /dev/null: Permission denied
/etc/profile.d/colorls.sh: line 19: /dev/null: Permission denied
/etc/profile.d/colorls.sh: line 20: /dev/null: Permission denied
-bash: cannot redirect standard input from /dev/null: Permission denied
FOP Server Started

 

[darmock]

Hmm those files are supposed to be created when you install the system. Then you also need to run update-scripts16 and update-fixes16 which should bring you up to date on all of those files.

I just did yet another default install and added the amd.conf file. I can add both sip and iax stuff fine. Hooked up a spare stanaphone sip trunk just fine made a call from a spare GXP2000 out via the sip line and an iax line. I also have a spare digium tdm400 card in the box and it also works from a sip phone dialing out and receiving calls.

1.6 does not have flite or festival working a known bug I am afraid so NONE of the applications that would make use of it will work AT ALL.

I did a simple setup thru fpbx point default inbound route to extension 7100. Set up outbound to go thru sip trunk then iax then zap trunk works as expected.

Beyond that I am afraid you will need to go to Digium.

The FOP server error seems to be cropping up on a few systems. Related to the FOP panel which is NOT a freepbx product and just happens to be included with FPBX. So until someone finds a fix you will have to live with it.

I have several other patches which I am going to push out tomorrow related to cdr on Ast 1.6 but too tired tonight.

Tom

 

[gbrook]

Interestingly I tried a minimum install of the FreePBX Modules once and found that unless you add some of the additional Modules (I never found which one) the _custom files did not get installed until one of the additional modules was installed.

Cheers
Garry

 

[mystere]

Well, I did a default install and it did *NOT* install those files. Booted, typed in ks16, went through the setup, configured networking, ran update-scripts16, ran update-fixes16, configured zaptel to use ztdummy (i have no hardware, just SIP), ran genzaptelconf, added the extensions via freepbx, couldn't dial anything.

We must be doing something different because those files did not get installed when I did it. I'm using the 1.2 iso i downloaded last week.

I fixed the FOP error. For some reason, /dev/null only had root permissions.

I don't care about festival or cepstral or flite or any of that stuff. I'm just trying to do basic testing, ideally with TLS. I may setup a secondary production server on 1.6 strictly for TLS and link it to my production 1.4 server for everything else. That way, if 1.6 flakes it won't take everything else with it.

 

[mystere]

Which modules might those be? Any idea? That would really help. Thanks.

 

[mystere]

By the way, when I run update-scripts16 it says the version is 1.2.3 released on 050808 and when i run update-fixes16 it says it's version 1.3.8 released on 051708, i thought i read yesterday that there was supposed to be a 1.3.9 on the 18th.

Also, after running update-fixes, the /dev/null error comes back.. so something in update-fixes must be changing the permissions on /dev/null

 

[darmock]

1.3.9 is not soup yet. There are a number of new patches to be added so be patient.

I will look into the /dev/null error thanks for pointing it out.

Tom