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Comfort noise Support in 11.10.2?

November 17, 2014, 2:24 pm
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Is there anyway to add comfort noise support to Asterisk Version 11.10.2 for silence on SIP channels? I really need it.

Statistics : Posted by medicranger • on Mon Nov 17, 2014 4:24 pm • Replies 1 • Views 29
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SIP trunk doesn't reconect after internet problem

November 17, 2014, 2:17 pm
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At a guess, you have a dynamic address and your address has changed.

Statistics : Posted by mmg_asterisk • on Mon Nov 17, 2014 4:17 pm • Replies 1 • Views 33
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SIP trunk doesn't reconect after internet problem

November 17, 2014, 2:17 pm
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Hello, thanks for reading

I've a problem when internet connection interrupt, and after a while it comes back.
When it happens, my trunk doesn't work as usual.
If I execute sip show registry, I see it registered, but when I try to make a call, I obtain "all lines are busy" message, although I can recieve incoming calls.
I can only fix this problem executing "module reload"

Does anyone have an answer?

My config:

username=TRKxxxxx-xxx
type=friend
trustrpid=no
sendrpid=yes
secret=xxxxxxxxxxxxxxx
qualify=yes
maxexpiry=120
insecure=invite
host=trk.cpsnet.com.ar
fromuser=TRKxxxxx-xxx
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ilbc

register string:
TRKxxxxx-xxx:xxxxxx@trk.cpsnet.com.ar/TRKxxxxx-xxx

Thanks again!!
Marco

Statistics : Posted by mmg_asterisk • on Mon Nov 17, 2014 4:17 pm • Replies 1 • Views 33
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Asterisk AJAM issues through virtual host.

November 17, 2014, 4:20 pm
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Hi all

I wanted to be able to host AJAM on a different server so I used proxy pass to achieve this, I did the following:

I added:
Code: <VirtualHost *:80>
ServerName localhost
ProxyPass /proxy http://localhost:8088/asterisk
ProxyPassReverse /proxy http://localhost:8088/asterisk
</VirtualHost>


to my httpd.conf file in /etc/httpd/conf. I am now able to access the static files from http://server-ip/asterisk/static instead of http://server-ip/:8088/asterisk/static. So for example I went to http://server-ip/asterisk/static/mantest.html and I was able to access the page, but mantest isnt able to actually make any requests. I then went to http://server-ip/asterisk/rawman?action ... t=password and I got this:

Quote:Service Temporarily Unavailable

The server is temporarily unable to service your request due to maintenance downtime or capacity problems. Please try again later.

Apache/2.2.15 (CentOS) Server at server-ip Port 80


Any know what I am doing wrong? Thanks.

Statistics : Posted by wmarshall • on Mon Nov 17, 2014 6:20 pm • Replies 0 • Views 13
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RTP port-range

November 18, 2014, 2:00 am
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Use the CLI to confirm the port numbers actually used; maybe the file was misparsed and you have defaults.

Also note that you must not restrict remote port numbers, as they are controlled by the remote end.

Statistics : Posted by leegethas • on Tue Nov 18, 2014 4:00 am • Replies 1 • Views 63
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RTP port-range

November 18, 2014, 2:00 am
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I've been messing with Asterisk for a few weeks now. So far, I really like it. I'm hooked!

I'm running my own asterisk box at home for a few days now. Just to get the hang of it. See if any problems come up, and if I can tackle those.

One problem I had, was that a caller could hear me, but I couldn't hear the caller. Eventually I figured this had to do with blocked RTP-ports. In rtp.conf I have this:
Code: [general]
rtpstart=7078
rtpend=7110

But tcpdump showed me that calls were using ports much higher. And sure enough, after opening UDP-port 10000-20000, everything worked fine.

The thing is, I don't have this port-rage defined anywhere. Not that I know of, anyway. So, why are incoming calls using these ports anyway, expecting them to be open? Is it mandatory/standard to have this port-range open?

Statistics : Posted by leegethas • on Tue Nov 18, 2014 4:00 am • Replies 1 • Views 63
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Forcing hangup of existing extension from dialplan

November 18, 2014, 7:48 am
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You can't hangup and extension because an extension can have thousands of calls, and there would be no way to distinguish between them.

I don't believe you can hangup a device (and even devices can have thousands of calls, if they represent trunks, and upwards of two, even if they represent phones).

If nothing else, you can hangup a channel by channel redirecting it to dialplan that runs Hangup().

The forum for support questions is Asterisk Support.

Statistics : Posted by alisterr • on Tue Nov 18, 2014 9:48 am • Replies 1 • Views 54
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Forcing hangup of existing extension from dialplan

November 18, 2014, 7:48 am
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Hi,

I'm trying to do the following:

Have an extension that may be called using its normal number (6010, for example), or via a modified one (106010, for example) which will cause the destination phone to auto-answer. In the latter case, when the system I control that communicates with Asterisk starts up, I'm currently dynamically creating special extensions in the dialplan via AMI to perform the operation:
Code: command: dialplan remove extension 106010@exten-rules"
command: dialplan add extension 106010,1,SIPAddHeader,\"Call-Info: answer-after=0\" into exten-rules"
command: dialplan add extension 106010,2,Macro,\"internalcall,6010,SIP/6010\" into exten-rules"
command: dialplan add extension 106010,3,Voicemail,\"6010,u\" into exten-rules


This works fine - a call to 106010 causes the Grandstream phone I'm using at extension 6010 to auto-answer, except that ideally what I'd like to happen is for any existing call in progress for 6010 to get hung up when Asterisk gets a call to the 106010 auto-answer one.

I'm using Asterisk 1.8, and can't see how this can be achieved. The SoftHangup application isn't present, which is what I was drawn to attempting first, and I can't seem to execute any CLI commands from the dialplan that might do the job (not system ones, but Asterisk ones).

Does anyone have any cunning (dial)plan to do this, please?

Statistics : Posted by alisterr • on Tue Nov 18, 2014 9:48 am • Replies 1 • Views 54
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How to use a ring group in a extension

November 18, 2014, 10:20 am
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Ring groups are FreePBX concepts, not Asterisk concepts.

Statistics : Posted by real_skydiver • on Tue Nov 18, 2014 12:20 pm • Replies 1 • Views 41
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How to use a ring group in a extension

November 18, 2014, 10:20 am
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Hi folks,

I have created a ring-group 600 (Applications -> Ring Group). I can call this ring-group fine from a IP phone. But if I call this group trough a extension from outside (ISDN) I get this error message (log):

VERBOSE[24575] pbx.c: -- Executing [47@isdn-in:1] Dial("CAPI/ISDN1#02/47-0", "SIP/600,45,tT") in new stack
WARNING[24575] chan_sip.c: Purely numeric hostname (600), and not a peer--rejecting!
WARNING[24575] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

The extension looks like this:

exten => 47,1,Dial(SIP/600,45,tT)

How can I use/name this ring group in the extension?

BTW: if I replace the extension with a number of a existing (IP) phone, it works just fine.

Statistics : Posted by real_skydiver • on Tue Nov 18, 2014 12:20 pm • Replies 1 • Views 41
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SIP trunk doesn't reconect after internet problem

November 17, 2014, 2:17 pm
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Set maxexpiry as low as you can get away with.

Statistics : Posted by mmg_asterisk • on Mon Nov 17, 2014 4:17 pm • Replies 3 • Views 117
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SIP trunk doesn't reconect after internet problem

November 17, 2014, 2:17 pm
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Thanks for your answer!

I don't think so, but, if this was the problem, how can I fix it automatically? without my intervention..

Regards

Marco

Statistics : Posted by mmg_asterisk • on Mon Nov 17, 2014 4:17 pm • Replies 3 • Views 117
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A little help: when a**gic phone answer,ip phones still ring

December 12, 2013, 8:20 am
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My conf files

/etc/asterisk/chan_dahdi.conf
Code: [trunkgroups]

[channels]
context=entrata
language=it
signalling=fxs_ks
rxwink=300              ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=2
pickupgroup=2
immediate=no

;Uncomment these lines if you have problems with the disconection of your analog lines
busydetect=yes
busycount=6

#include /etc/asterisk/dahdi-channels.conf


extension.conf
Code: [interni]
include => diamondcardterm
include => entrata

exten => 200,1,Dial(dahdi/4/outgoing_number) // dial 200 to dialout from dahdi channel 4
exten => 200,1,Set(LANGUAGE()=it)
exten => 200,2,Hangup

exten => 1001,1,Dial(SIP/1001,20,t,m)
;exten => 1001,2,Voicemail(1001@interni)
exten => 1001,3,Hangup

exten => 1002,1,Dial(SIP/1002,20,t,m)
;exten => 1002,2,Voicemail(1002@interni)
exten => 1002,3,Hangup

exten => 1003,1,Dial(SIP/1003,20,t,m)
;exten => 1003,2,Voicemail(1003@interni)
exten => 1003,3,Hangup

exten => 1004,1,Dial(SIP/1004,20,t,m)
;exten => 1004,2,Voicemail(1004@interni)
exten => 1004,3,Hangup

exten => 7500,1,VoicemailMain(@interni)

exten => 600,1,Answer()
exten => 600,2,Playback(demo-echotest) ; Let them know what
exten => 600,3,Echo()                  ; Do the echo test
exten => 600,4,Playback(demo-echodone) ; Let them know it
exten => 600,5,Hangup()


[entrata]
exten => s,1,Answer()
exten => s,2,GotoIf(${BLACKLIST()}?blacklisted)
exten => s,3,Dial(SIP/1003&SIP/1002&SIP/1001&dahdi/1,150,t,m)
;exten => s,3,Voicemail(1002@interni)
exten => s,4,Hangup()


Statistics : Posted by manduto22 • on Thu Dec 12, 2013 10:20 am • Replies 3 • Views 372
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A little help: when a**gic phone answer,ip phones still ring

December 12, 2013, 8:20 am
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No way?
I never solved this Image

Statistics : Posted by manduto22 • on Thu Dec 12, 2013 10:20 am • Replies 3 • Views 372
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A little help: when a**gic phone answer,ip phones still ring

December 12, 2013, 8:20 am
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I have set a PBX asterisk for home testing
i have 2 analogic phone connected on their fxs port
and 2 ip phones connected to pbx wich is connected to a fsx port.
When someone call,if i answer with ip phone,the analogic phone stop
ringin(and is ok),but if i answer with analogic phone the ip phone still ring.
My configuration

Code: [mycontext]

exten => 200,1,Dial(dahdi/1/outgoing_number) // dial 200 to dialout from dahdi channel 1
exten => 200,1,Set(LANGUAGE()=it)
exten => 200,2,Hangup

exten => 1001,1,Dial(SIP/1001,10,t,m)
;exten => 1001,2,Voicemail(1001@mycontext)
exten => 1001,3,Hangup

exten => 1002,1,Dial(SIP/1002,10,t,r,m)
;exten => 1002,2,Voicemail(1002@mycontext)
exten => 1002,3,Hangup

exten => 1003,1,Dial(SIP/1003,10,t,r,m)
;exten => 1003,2,Voicemail(1003@mycontext)
exten => 1003,3,Hangup

exten => 7500,1,VoicemailMain(@mycontext)

exten => 600,1,Answer()
exten => 600,2,Playback(demo-echotest) ; Let them know what
exten => 600,3,Echo()                  ; Do the echo test
exten => 600,4,Playback(demo-echodone) ; Let them know it
exten => 600,5,Hangup()

[from-pstn]
exten => s,1,Answer()
exten => s,2,Dial(SIP/1002&SIP/1001&dahdi/2,20,r,t,)
exten => s,3,Voicemail(1002@mycontext)
exten => s,4,Hangup()


Ip phones are grandstream 1400
card is TDM410P
A little help?
Thanks

Statistics : Posted by manduto22 • on Thu Dec 12, 2013 10:20 am • Replies 3 • Views 372
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Trixbox Voicemail Issue

November 19, 2014, 1:41 am
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It is also abandonware. Fonality don't support and they have even closed down their peer support forum.

Statistics : Posted by faizanelahi • on Wed Nov 19, 2014 3:41 am • Replies 2 • Views 91
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Trixbox Voicemail Issue

November 19, 2014, 1:41 am
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I'm not sure who's going to be able to give you advice on Trixbox. It's an extremely old and poorly implemented fork of Asterisk and of FreePBX. If you want a FreePBX interface on a supported platform, see the FreePBX distro from Schmoozecom - www.freepbx.org

Cheers

Statistics : Posted by faizanelahi • on Wed Nov 19, 2014 3:41 am • Replies 2 • Views 91
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Trixbox Voicemail Issue

November 19, 2014, 1:41 am
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Hello all,
I am integrating trixbox 2.8.0.4 with cisco call mananger express for voicemail feature.
Voicemail is working fine internally (extensions to extensions ).
But if anyone calling externally it is not going to voicemail after ringing.
It is giving a message.." The number is not in service "
Please guide..as i am totally new to this platform.
Further inputs will be provided on request.

Statistics : Posted by faizanelahi • on Wed Nov 19, 2014 3:41 am • Replies 2 • Views 91
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Remote Dialtone

November 10, 2014, 7:22 pm
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Image Okay, got this figured out -- kinda/sorta. Take it for what it's worth from a crazy, old guy.

System A wants to access local 7-digit dial tone on System B.

Outgoing call rule on System A looks like this:
Number begins with *01 (can be whatever you want).
Rest of the number must be between 5 and 8 digits (want 5 digits so they can dial extensions between the systems)
Before connecting the call trim 3 digits (removes the *01)

System B incoming call rules look like this, one for extension dialing and one for dialing outside, local 7-digit numbers:
Beginning of range 10000 -- end of range 99999. Voice Calls. Any Provider. Don't trim any digits. Don't add anything. This one is for inter-office extension dialing.

Beginning of range 90000000 -- end of range 99999999. Voice Calls. Any Provider. Trim 8 Digits -- basically the entire number. Add 19876 -- this is the number of a IVR I'm going to use here. It can be any IVR you want.

The IVR has two steps. First step sets a variable to the INCOMING_DID which is the number I dialed from the other system.

The second step uses the Send To External Number action and sends the variable containing the INCOMING_DID set in the first step.

Works for me. May be a hole in there somewhere but it works for what I want and limits remote dial tone to local numbers.

Statistics : Posted by jreed1949 • on Mon Nov 10, 2014 9:22 pm • Replies 2 • Views 144
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Remote Dialtone

November 10, 2014, 7:22 pm
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If both systems are Switchvox PBXes, Id' suggest using the Switchvox PEERING mechanisms rather than connecting through IAX. IAX is really for connecting Switchvox to non-Switchvox systems and I think you'll find that a lot of the "mess" is resolved by using the Switchvox peering options by default. Also, what version of Switchvox are you running on each end? This should absolutely be doable and we've used it for 'least cost routing' on a number of occasions.

Statistics : Posted by jreed1949 • on Mon Nov 10, 2014 9:22 pm • Replies 2 • Views 144
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