I have Asterisk installed on a Gentoo box with 2 SIP phones configured. I am able to dial between the two using the extension I configured, so Asterisk appears to be working fine.
I then installed and tried both version .26 and .27 of the Flash Operator Panel, but can’t get it to work completely. I’m trying to drag one phone to the other to create a call, however when I do this the first phone rings and when I answer, the 2nd never rings and the first simply hangs up. I ran op_server.pl with debug on and it looks like the callerid variable is never getting set so it doesn’t know where to dial back to.
** DIGEST_EVENT: GOT SIP/D3877|ocupado1|&incoming,[]|1153925740.55-0|SIP/D3877-9666
** DIGEST_EVENT: canal: SIP/D3877
** DIGEST_EVENT: quehace: ocupado1
** DIGEST_EVENT: dos: &incoming,[] <------ BLANK???
** DIGEST_EVENT: uniqueid: 1153925740.55-0
** DIGEST_EVENT: canalid: SIP/D3877-9666
Below is my op_buttons.cfg:
[SIP/D3877]
Position=1
Label=“D3877”;
Extension=3877
Context=internal
Mailbox=100@internal
No_Rectangle=0
Background=bg.jpg
Privacy=false
callerid=“D3877”
[SIP/D3878]
Position=n
Label="D3878"
callerid="D3878"
Extension=3878
Context=internal
Mailbox=101@internal
Icon=1
Here is the output I get from the Asterisk console as well:
-- Executing Goto("SIP/D3877-9666", "internal") in new stack
Jul 26 09:55:45 NOTICE[11816]: pbx.c:1753 pbx_extension_helper: No such label ‘internal’ in extension ‘s’ in context 'default’
Jul 26 09:55:45 WARNING[11816]: pbx.c:6514 ast_parseable_goto: Priority ‘internal’ must be a number > 0, or valid label
== Spawn extension (default, s, 1) exited non-zero on ‘SIP/D3877-9666’
Any help is appreciated, thanks.
what that error indictates is that FOP is looking for something like so:
[default]
exten => s-internal,1,…
in extensions.conf or whatnot.
i would refer to the FOP documentation and find out if it requires a specific set of parameters set up to work with your particular asterisk setup. i myself have never messed with FOP, but i’m guessing it’s looking for a context that either doesn’t exist or doesn’t have the right info.
if you post your extensions.conf, we might be able to figure this out.
Here is my extensions.conf, it’s very basic as I am just starting to learn Asterisk:
[code][general]
static=yes
writeprotect=yes
[default]
exten => s,1,Goto(internal)
[globals]
HARD1=SIP/D3877
HARD2=SIP/D3878
SOFT1=SIP/D3876
SOFT2=SIP/D3875
[internal]
exten => 1001,1,Goto(conferences,1001,1)
exten => 3877,1,Dial(${HARD1},15,wt)
exten => 3877,2,Voicemail(u${HARD1}@internal)
exten => 3877,102,Voicemail(b${HARD1}@intenral)
exten => 3878,1,Dial(${HARD2},15,wt)
exten => 3878,2,Voicemail(u${HARD2}@internal)
exten => 3878,102,Voicemail(b${HARD2}@intenral)
[conferences]
exten => 1001,1,MeetMe(1001)[/code]
I just tried removing the [default] section all together and then got this in the Asterisk console:
> Channel SIP/D3877-dea7 was answered.
== Starting SIP/D3877-dea7 at internal,1 failed so falling back to exten ‘s’
== Starting SIP/D3877-dea7 at internal,s,1 still failed so falling back to context 'default’
Jul 26 10:22:45 WARNING[11859]: pbx.c:2357 __ast_pbx_run: Channel ‘SIP/D3877-dea7’ sent into invalid extension ‘s’ in context ‘default’, but no invalid handler
try this:
[default]
exten => s-internal,1,Goto(internal)
not sure it will work, but it might give us a bit more to work with.
Here’s what I got after making those changes:
> Channel SIP/D3877-e3a8 was answered.
== Starting SIP/D3877-e3a8 at internal,1 failed so falling back to exten ‘s’
== Starting SIP/D3877-e3a8 at internal,s,1 still failed so falling back to context 'default’
Jul 26 10:33:32 WARNING[11882]: pbx.c:2357 __ast_pbx_run: Channel ‘SIP/D3877-e3a8’ sent into invalid extension ‘s’ in context ‘default’, but no invalid handler
here we go:
FOP isn’t passing an extension, it’s just passing a context and a priority. the FOP config isn’t correct.
if you were to add this to your internal context
exten => i,1,playback(tt-weasels)
you would probably get some audio when attempting the call.
i just checked the docs for FOP, and i THINK you need to review the op_buttons.cfg file and double check that the extension value is set for each entry…
I appreciate the help.
I added the line you recommended however it doesn’t play a sound file. As soon as I pick up the first phone it hangs up immediately.
> Channel SIP/D3877-3884 was answered.
== Starting SIP/D3877-3884 at internal,1 failed so falling back to exten ‘s’
== Starting SIP/D3877-3884 at internal,s,1 still failed so falling back to context 'default’
Jul 26 11:08:12 WARNING[12190]: pbx.c:2357 __ast_pbx_run: Channel ‘SIP/D3877-3884’ sent into invalid extension ‘s’ in context ‘default’, but no invalid handler
since no extension is defined in the fop config, it’s looking for the s extension:
== Starting SIP/D3877-dea7 at internal,1 failed so falling back to exten ‘s’
when it can’t find s, it goes back to the default context
== Starting SIP/D3877-dea7 at internal,s,1 still failed so falling back to context ‘default’
since there is no s extension in the default context, the system errors out.
you need to verify your FOP config has an extension defined, because nothing is being passed.
barring that, you could rewrite your dialplan, but i don’t think that would do much good, as the routing from FOP is still going to be invalid.
check your FOP configs
I feel like I have tried everything and am still having no luck.
[SIP/D3877]
Position=1
Label="D3877"
Extension=D3878
Context=internal
Voicemail=3877
Mailbox=3877@internal
Icon=1
[SIP/D3878]
Position=2
Label="D3878"
Extension=D3878
Context=internal
Voicemail=3878
Mailbox=3878@internal
Icon=1
oh…take the D off the extension listing, like so:
Extension=3878
try that.
Removing the ‘D’ didn’t help. I do now notice whenever I try to connect the call this error appears from the Flash Operator Panel:’
Use of uninitialized value in split at ./op_server.pl line 6474.
Use of uninitialized value in concatenation (.) or string at ./op_server.pl line 6490.
Use of uninitialized value in concatenation (.) or string at ./op_server.pl line 6492.
Those variables that aren’t getting initialized appear to be the extension and the context, hence the whole reason this thing isn’t working. I’m just not sure why since I have them defined.
have you tried possibly downloading a clean version and reinstalling? also, you might check that your version of perl is up to date.
past that, this appears to be a problem with FOP, not asterisk - they have a mailing list, you might try sending a message out on that, with the error you pasted below.
/sorry, don’t know perl