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davidnewt
join:2009-08-10
Conroe, TX

davidnewt

Member

[Asterisk] Help on Asterisk configuration

I have used sipsorcery for my home phone service for some time and I am very satisfied. Now, I want to try some new stuff. I have installed Asterisk on my WL520GU router (with Tomato) following Mango's instruction (Thanks a lot, Mango). I have struggled for this weekend to try to make it work. Now, It works half way: I can place calls but can not receive calls right now. I am wondering if someone could help to find out what is wrong. Here is my setting:

The WL520gu router where Asterisk is installed is set as a WDS node with IP 192.168.1.10 (the main router is 192.168.1.1). I tried to register 3 phones (2 xlite softphones and 1 pap2t). Here is my sip.conf and extensions.conf. Since I am new to Asterisk, I removed most of those advanced settings and only try to use the basic functions.

sip.conf:

; SIP Configuration for Asterisk
;
[general]
context=sip       
allowguest=no          
allowoverlap=no        
bindport=5060         
bindaddr=0.0.0.0       
srvlookup=yes         
disallow=all           
allow=ulaw             
alwaysauthreject=yes  
canreinvite=no         
nat=yes                
session-timers=refuse
externip=xx.xxx.xxx.xxx
localnet=192.168.1.10/255.255.255.0 
 
; Register to sip providers
;
register => xxxxx:xxxxxx@newyork.voip.ms:5060
register => xxxxxx:xxxxxxx@sipsorcery.com:5060
register => xxxxx:xxxxxx@nonoh.net:5060
 
maxexpirey=3600  
defaultexpirey=1200
 
;sip providers
[voipms]
canreinvite=no
context=sip
host=newyork.voip.ms
secret=xxxxxxx
type=friend
username=xxxxxx
disallow=all
allow=ulaw
fromuser=xxxxxx
trustrpid=yes
sendrpid=yes
insecure=port,invite
nat=yes 
;
[sipsorcery]
type=friend
secret=xxxxxx
username=xxxxxx
host=sipsorcery.com
;
[betamax]
canreinvite=no
context=sip
host=nonoh.net
secret=xxxxxx
type=friend
username=xxxxxx
disallow=all
allow=ulaw
fromuser=xxxxxxx
trustrpid=yes
sendrpid=yes
insecure=port,invite
nat=yes 
;
;pap2t
[pap2t]
type=friend
username=pap2t
secret=pap2t
host=dynamic
port=5060
context=sip
canreinvite=yes
dtmfmode=rfc2833
 
;xlite on main computer
[xlite]  
type=friend 
username=xlite
secret=xlite 
context=sip
canreinvite=no 
host=dynamic 
dtmfmode=rfc2833 
qualify=200 
mailbox=1003 
nat=1 
 
;xlite on laptop
[xlite2]  
type=friend 
username=xlite2
secret=xlite2 
context=sip
canreinvite=no 
host=dynamic 
dtmfmode=rfc2833 
qualify=200 
mailbox=1004 
nat=1 
 

extensions.conf:

; extensions.conf - the Asterisk dial plan
;
[general]
static=no
writeprotect=no
autofallthrough=yes
clearglobalvars=yes
priorityjumping=no
 
[globals]
PHONE1=SIP/xlite
PHONE2=SIP/pap2t
PHONE3=SIP/xlite2
 
FWDUSERID=11001
FWDUSERNAME=myname
 
MYNAME=myname
 
[sip]
exten => 1111,1,Dial(SIP/pap2t,20,tr)
exten => 1234,1,Dial(SIP/xlite,20,tr)
exten => 4321,1,Dial(SIP/xlite2,20,tr)
 
include => voipms-outbound
include => voipms-inbound
 
[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@betamax)
exten => _00.,n,Hangup()
 
; inbound context example
 
[voipms-inbound]
exten => xxxxxxx,1,Answer()
 
 

The 3 phones can call each other without problem. The only problem I have now is I can not get incomming calls from outside. All the sip providers are registered. One odd thing is the in sipsorcery, the contact URI is sip:s@myipaddress, instead of the username@myipaddress.

Thanks a lot in advance.

DogFace056
join:2005-12-09
Cary, NC

4 edits

DogFace056

Member

Replace the "register" statements with:
register => xxxxx:xxxxxx@newyork.voip.ms:5060/extension
register => xxxxxx:xxxxxxx@sipsorcery.com:5060/extension
register => xxxxx:xxxxxx@nonoh.net:5060/extension
 
where "extension" is a placeholder for the actual extension you want the incoming calls to ring, eg "1111". If not explicitly specified, Asterisk uses the default extension "s", which would normally go to the automated attendant. But you don't have one set up.

"Extension" is the user id specified in the SIP "contact" header when registering. It's the id by which you're known locally to your system, not the provider's. Unfortunately, some providers for absolutely senseless reasons require that this "extension" match the user name that you register as with the provider--a most imbecilic policy, as it's none of their business to impose what you should be known as to your own system.

Alternatively, you can leave the register statements unchanged, and add the following under the [sip] context in extensions.conf to have your PAP2T ring:
exten => s,1,Dial(SIP/pap2t,20,tr)
 
davidnewt
join:2009-08-10
Conroe, TX

davidnewt

Member

Click for full size
Thank you very much for your reply. I have tried to put extension at the end of the registration, now, the asterisk CLI can catch the call, but give the error message as in the image.

register => xxxxxxx:xxxxxx@sipsorcery.com:5060/1111
 

Please let me know if anything else I should do. Thanks a lot.
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo to davidnewt

Premium Member

to davidnewt
As you said in your post, your Asterisk PBX system has no problem to register to the intended VoSP servers, except no incoming calls. This may be caused by either a NAT/Firewall router or a misconfigured dialplan context to process the incoming call. A quick reading through your sip.conf and extension.conf files indicates both configuration files look alright to me, except I would write it a bit different way. Anyway, here is what I meant:
externip=xx.xxx.xxx.xxx   ; It is better to use externhost
externhost=your.FQDN      ; For your FQDN, I would use a free dynDNS.org
localnet=192.168.1.10/255.255.255.0 ; I would use 192.168.1.0/24
 
; Register to sip providers  
;  
register => xxxxx:xxxxxx@newyork.voip.ms/SIP
register => xxxxxx:xxxxxxx@sipsorcery.com/SIPSorcery
register => xxxxx:xxxxxx@nonoh.net/Nonoh
;
[sipsorcery]
context=sip     ; <<<< == Add this context
type=friend
secret=xxxxxx
username=xxxxxx
host=sipsorcery.com
   :
[pap2t]
context=phone  ; Should be a different context from inbound
   .
   :
[xlite]
context=phone  ; Should be a different context from inbound
   .
   :
[xlite2]
context=phone  ; Should be a different context from inbound
 
Then, change your extension.conf to match the context as shown below:
[internal]
exten => 1111,1,Dial(SIP/pap2t,20,tr)
exten => 1234,1,Dial(SIP/xlite,20,tr)
exten => 4321,1,Dial(SIP/xlite2,20,tr)
 
[sip]
exten => SIP,1,Dial(SIP/1111)
exten => SIPSorcery,1,Dial(SIP/1234)
exten => Nonoh,1,Dial(SIP/4321)
 
[phone]                     ; <<<<==== Add this section for outbound context
include => internal
include => voipms-outbound
;include => voipms-inbound  ; <<<<==== Remove/comment this line
 
[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@betamax)
exten => _00.,n,Hangup()
 
Once you have done the above, then issue module reload on your asterisk CLI to reload both sip.conf and extension.conf files. Please be sure to back up your configuration files before you make any changes. This way, if the changes don't work, you can always restore your configuration back to it was before.

DogFace056
join:2005-12-09
Cary, NC

DogFace056 to davidnewt

Member

to davidnewt
The "type" entries for your providers have to be set to "peer", not "friend". The "friend" type causes asterisk to require authentication of each caller. Not only would you have to create a "friend" section for each and every possible caller, but the originating provider would have to know what credentials you'd assigned that caller in order to be able to authenticate it. Trunks are not meant to be set up as type "friend".
davidnewt
join:2009-08-10
Conroe, TX

1 edit

davidnewt

Member

Thanks a lot for all the replies. I have made some progress, but still have problem: my ipcomms DID can work now, but my voip.ms DID can not work. To make things simple, I keep only 2 providers (ipcomms and voip.ms) and only 2 phones (one pap2t and one xlite). I tried to make both voip.ms and ipcomms DIDs have the exact configuration, but one works the other not, even though it shows registered in voip.ms website. Another thing: if I use the type=peer, then, I have 1-way audio for incomming calls from ipcomms DID, with type=friend, the audio is normal. Also, voip.ms and ipcomms do not support DNS SRV lookup. In asterisk CLI, the voip.ms DID calls do not show up at all. Here is my configuration:

sip.conf:

; SIP Configuration for Asterisk
;
[general]
context=sip      
allowguest=no          
allowoverlap=no        
bindport=5060          
bindaddr=0.0.0.0      
srvlookup=no          
disallow=all           
allow=ulaw,g729            
alwaysauthreject=yes   
canreinvite=no         
nat=yes               
session-timers=refuse
externhost=myname.dyndns.org
localnet=192.168.1.0/24
 
; Register to sip providers
register => xxxxxx:xxxxxxx@newyork.voip.ms:5060/555111111
register => xxxxxxxx:xxxxxxxx@sipconnect.ipcomms.net:5060/666111111
 
maxexpirey=3600  
defaultexpirey=1200
 
;sip providers
[voipms]
context=sip
canreinvite=no
host=newyork.voip.ms
secret=xxxxxx
type=peer
username=xxxxxx
disallow=all
allow=ulaw,g729
fromuser=xxxxxx
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes ; Uncomment this if your box is behind a NAT
;
[ipcomms]
context=sip
username=xxxxxxxx
type=friend
secret=xxxxxxxx
host=sipconnect.ipcomms.net
fromuser=xxxxxxx
allow=ulaw,g729 ; we support ulaw,alaw,ilbc,gsm,g723.1,g726,g729a
trustrpid = yes
sendrpid = yes
canreinvite = no
insecure=invite
nat=yes
 
; My SIP phones in the house/office are listed below
;
;pap2t
[pap2t]
type=friend
username=pap2t
secret=pap2t
host=dynamic
port=8060
context=phone
canreinvite=yes
dtmfmode=rfc2833
nat=1
 
;xlite on main computer
[xlite]  
type=friend 
username=xlite
secret=xlite 
context=phone
canreinvite=no 
host=dynamic 
dtmfmode=rfc2833 
qualify=200 
mailbox=1003 
nat=1 
 
 

extensions.conf

;
[general]
static=no
writeprotect=no
autofallthrough=yes
clearglobalvars=yes
priorityjumping=no
 
[globals]
PHONE1=SIP/xlite
PHONE2=SIP/pap2t
 
[internal]  
exten => 1001,1,Dial(SIP/pap2t,20,tr)  
exten => 1002,1,Dial(SIP/xlite,20,tr)  
   
[sip]  
exten => 666111111,1,Dial(SIP/pap2t,20,tr)  ;ipcomms DID
exten => 555111111,1,Dial(SIP/pap2t,20,tr)  ;voipms DID
   
[phone]                      
include => internal  
include => sip
include => voipms-outbound  
   
[voipms-outbound]  
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)  
exten => _1NXXNXXXXXX,n,Hangup()  
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)  
exten => _NXXNXXXXXX,n,Hangup()  
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)  
exten => _011.,n,Hangup()  
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)  
exten => _00.,n,Hangup()  
 
beaver
join:2006-03-12
Beaverton,OR

beaver to davidnewt

Member

to davidnewt
david,

I am watching closely what you are building.
One thing I am not clear. On your PBX server, how do you use the google voice call-back? Do you run a script (in Perl, C++, Ruby...)? or simply use sipsorcery (I could see you were trying it).
If you use sipsorcery, then its reliability is still an issue. maybe running a local sipsorcery is a better fix.
if not, how do you use the google voice call-back?

I could setup a pbx box home then lump all VSPs on it. But it will still have the known issues.
davidnewt
join:2009-08-10
Conroe, TX

2 edits

davidnewt

Member

beaver:

I am still using the sipsorcery for my home phone service. I am quite satisfied with SS and made donation to Aaron for the new servers. The reliability of SS is not a big concern for me since I have backup when it is down with close monitoring (forwarding incoming calls directly to SPA3000 and use ATA dialplan to forward the outgoing calls through one of the VSP in the gateways, this way works perfect except that I need to pay for outgoing calls). Running local SS is not good choice for energy efficiency. Now I am just trying to play Asterisk for fun at this time. So far, outgoing calls are ok, incoming calls are ok for ipcomms DID, but voip.ms DID not work. I hope the experts on Asterisk here could help.
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo to beaver

Premium Member

to beaver
said by beaver:

One thing I am not clear. On your PBX server, how do you use the google voice call-back? Do you run a script (in Perl, C++, Ruby...)? or simply use sipsorcery (I could see you were trying it).
If your Asterisk PBX system (asterisk v1.6.x) is hosted on a Linux computer, you may want to check out this GV DialOut AGI scripts written by Paul Marks in a Python scripting language.
davidnewt
join:2009-08-10
Conroe, TX

davidnewt

Member

After 2 days' struggle, I finally made the incoming calls from both voip.ms and ipcomms work. Now the Asterisk server on the WL520gu router works fine with the basic functions I have tried (incoming and outgoing calls) and I will try to explore Asterisk more such as DISA etc. Thanks everyone for the help here. This is the best forum.
beaver
join:2006-03-12
Beaverton,OR

beaver

Member

Congratulations!
I am looking for a WRT54G router at local craigslist. I plan to build my PBX (linux) using it. ASUS router is an option.
Thanks mazilo for the Python scripting language. I save a copy of the script. At this time, I even don’t know Python language yet. Just like before I started my sipsorcery, I didn’t know any on Ruby language.
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo

Premium Member

said by beaver:

I am looking for a WRT54G router at local craigslist.
IIRC, a WRT54G router only has 4/16MB Flash/RAM. You will soon find out it will have barely enough space/RAM to hold/run an asterisk-1.4.x. You also notice from the GV DialOut AGI script page, it requires a new bridge() function (only available on asterisk-1.6.x and onward, IIRC) to bridge both the outgoing/incoming calls to make a GV call looks/feels like business as usual. OTOH, the GV DialOut AGI script is a Python base script. This means a Linksys WRT54G router needs more space/RAM to hold/run a Python scripting interpreter to process the GV DialOut AGI script that it may not have. I am sure some readers here may have ported this GV DialOut AGI script to other scripting languages that use a fraction of additional RAM. So, unless you have a plan to modify the Linksys WRT54G router, i.e. add more RAM and space through a USB port or an SD card, you probably want to toss out and let this idea vaporize into the thin air. OTOH, if you really want to have an Asterisk PBX system hosted on a Linux embedded system, perhaps this PlugPBX (a turn-key project) may not be a bad idea, except it is kind of pricey, YMMV. For that kind of money, perhaps one can probably suggest to get a used Asus eeePC, too! For me, I just bought me some Hitachi SimpleNET NAS and Patriot Box Office 1080P HD Media Player (PBO) from a local Fry's Electronics store for $30/each and $50/each (after a $25 rebate), respectively. I hope one day to turn these devices into an Asterisk PBX system with an OpenWRT firmware (as mentioned here and here). As of this writing, there is still no support from OpenWRT community on either device yet, FYI.
Thanks mazilo for the Python scripting language.
Funnily, I didn't write the scripts, except just to let you know where to get it. So, credits should go to Paul Marks who wrote the scripts.
davidnewt
join:2009-08-10
Conroe, TX

davidnewt to beaver

Member

to beaver
said by beaver:

Congratulations!
I am looking for a WRT54G router at local craigslist. I plan to build my PBX (linux) using it. ASUS router is an option.
Thanks mazilo for the Python scripting language. I save a copy of the script. At this time, I even don’t know Python language yet. Just like before I started my sipsorcery, I didn’t know any on Ruby language.
Thanks. As far as I tried, the Asus WL520GU router is better than WRT54G because there are different versions of WRT54G. The other choice is Asus WL500gp which has more flash size and memory and ideal for an Asterisk box (DD-WRT jumbo with build-in Asterisk can be installed).
davidnewt

davidnewt

Member

I need more help on the Asterisk configuration. Now I am trying the DISA function. In the Modules.conf, I have the following line:

load => app_disa.so
 

In the sip.conf, I have set a DID with DISA-in context:

[ipcomms]
context=DISA-in
username=mydid
type=friend
secret=mypwd
host=sipconnect.ipcomms.net
.
.
.
 

In the extensions.conf, I have the following:

[DISA-in]
exten => mydid,1,Answer()
;exten => mydid,n,Playback(enter-password)
exten => mydid,n,DISA(no-password,DISA-out)
 
[DISA-out]
exten => _18[0678][0678]NXXXXXX,1,Transfer(SIP/${EXTEN}@voipms)
 

With this simplest DISA configuration, I assume when I call mydid, I should get a dialtone from DISA application. The problem now is that after I call mydid, the line seems connected, but there is no sound. The Asterisk CLI shows the DISA is executing, but just hang there and doing nothing. Can anyone tell me what is wrong? Thanks a lot in advance.
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo

Premium Member

For your DISA-out context, try the following to see if that will work:
[DISA-out]
exten=>_NXXXXXXXXX,1,GoTo(1${EXTEN},1)
exten=>_1NXXXXXXXXX,1,Transfer(SIP/${EXTEN}@voipms)
 
davidnewt
join:2009-08-10
Conroe, TX

davidnewt

Member

said by mazilo:

For your DISA-out context, try the following to see if that will work:
[DISA-out]
exten=>_NXXXXXXXXX,1,GoTo(1${EXTEN},1)
exten=>_1NXXXXXXXXX,1,Transfer(SIP/${EXTEN}@voipms)
 

Thanks for the reply. I will try that. The problem is that it can not reach the DISA-out context yet. It seems stopped at the DISA call.
Stewart
join:2005-07-13

Stewart

Member

When your config asked it to first play a password request, did it? If not, but the log shows the Playback as having executed, you likely have a NAT related problem with the audio, which you could troubleshoot with captures, etc.
davidnewt
join:2009-08-10
Conroe, TX

davidnewt

Member

said by Stewart:

When your config asked it to first play a password request, did it? If not, but the log shows the Playback as having executed, you likely have a NAT related problem with the audio, which you could troubleshoot with captures, etc.
No, the caller can not hear anything. It only has dead silence. Since I am using "no-password", it should directly give the caller the new dialtone. There is no error message from the captures.
Stewart
join:2005-07-13

Stewart

Member

You config showed
;exten => mydid,n,Playback(enter-password) 
If you uncomment that, Asterisk should play a password request. If the Asterisk log says that is happening, but you don't hear anything, then you can debug this as a simple "no audio" problem. If the log doesn't show the Playback, then at least you'll know that the problem is not related to DISA. If the password request is heard (but not the dialtone), that is beyond my knowledge, though an Asterisk guru will likely know what's wrong.
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo to davidnewt

Premium Member

to davidnewt
David,

You created a [DISA-in] context, but there is no where on your sip.conf (the register) line points to the extension (mydid) on this context (see example below). As such, your Asterisk PBX system uses the default extension.

sip.conf:
register=><usernam>:<password>@example.sip.com/myDID
 

extensions.conf:
[# DISA-in]
exten => myDID,1,Answer()  
exten => myDID,n,DISA(no-password,DISA-out)
 
davidnewt
join:2009-08-10
Conroe, TX

davidnewt

Member

Thanks both Mazilo and Stewart. Actually I have that registration line for the DID and the incoming call has reached the DISA-in context.

BTW, as Stewart mentioned, I tried to use Playback() function. This is another problem for me: there is no sound files in the var/lib/asterisk directory for the built I installed on the router, so I downloaded a package of sound files (with .g711u format) into var/lib/asterisk/sounds/en/ directory and renamed one of the file to enter-password.g7711u. But the Playback() function gives an warning message saying the file can not be found. Where these files should be in and how the program can access these sound files? I don't know if this is related to the problem I have with DISA call.
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo

Premium Member

said by davidnewt:

Where these files should be in and how the program can access these sound files?
The directory is pointed by the variable astvarlibdir on your /etc/asterisk/asterisk.conf file.
davidnewt
join:2009-08-10
Conroe, TX

1 edit

davidnewt

Member

said by mazilo:

The directory is pointed by the variable astvarlibdir on your /etc/asterisk/asterisk.conf file.
I got lost now. No matter where I put the sounds files, the Playback() never be able to find the files and always give "No such file or directory" error. The file Asterisk.conf is below. I have tried to remove the (!) in the first line, but it is the same error. I have tried to put the sounds files both in /opt/var/lib/asterisk/ and /opt/var/lib/asterisk/sounds/.

I think the DISA problem may not be related to the sound problem.

[directories](!) ; remove the (!) to enable this
astetcdir => /opt/etc/asterisk
astmoddir => /opt/lib/asterisk/modules
astvarlibdir => /opt/var/lib/asterisk
astdbdir => /opt/var/lib/asterisk
astkeydir => /opt/var/lib/asterisk
astdatadir => /opt/var/lib/asterisk
astagidir => /opt/var/lib/asterisk/agi-bin
astspooldir => /opt/var/spool/asterisk
astrundir => /opt/var/run/asterisk
astlogdir => /opt/var/log/asterisk
.
.
.
 
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo

Premium Member

said by davidnewt:

I got lost now. No matter where I put the sounds files, the Playback() never be able to find the files and always give "No such file or directory" error. The file Asterisk.conf is below. I have tried to remove the (!) in the first line, but it is the same error. I have tried to put the sounds files both in /opt/var/lib/asterisk/ and /opt/var/lib/asterisk/sounds.
On your /etc/asterisk/modules.conf file, make sure you have the app_playback.so module loaded. At any rate, be sure the PlayBack() function works with other context.
I think the DISA problem may not be related to the sound problem.
Worse comes to worst, remove the PlayBack() line and see if you can get the dial tone. When you call in and there is no dial tone, try to dial a number and see if the call will go through.
davidnewt
join:2009-08-10
Conroe, TX

davidnewt

Member

Thanks.

Yes, the Playback module is loaded. Here is the modules.conf.

[modules]
autoload=no                    ; Only load explicitely declared modules
load => format_pcm.so          ; Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.711
load => codec_ulaw.so          ; mu-Law Coder/Decoder
load => app_dial.so            ; Dialing Application
load => app_macro.so           ; Extension Macros
load => app_playback.so        ; Sound File Playback Application
load => app_mixmonitor.so      ; Record calls
load => app_setcallerid.so     ; Set CallerID Presentation Application
load => app_disa.so            ; DISA - used for calling card-type projects
load => app_transfer.so        ; Transfer calls - good for use with DISA so that you don't proxy audio
load => func_timeout.so        ; Adjust timeout; handy for use with DISA.  But not essential if you can dial quickly.
load => func_callerid.so       ; Caller ID related dialplan functions
load => func_logic.so          ; GotoIf() and friends
load => func_strings.so        ; String handling functions
load => pbx_config.so          ; This loads your dialplan
load => pbx_spool.so           ; This is needed to make call files work
load => chan_sip.so            ; SIP
load => res_musiconhold.so     ; Music-on-Hold
load => func_shell.so          ; Execute shell commands and use the output in the dialplan.  (Useful for formatting things with gnu-sed.)
load => func_channel.so        ; Find information about the channel.  (Used with our implementation of DISA.)
 

I tried to remove the Playback() before DISA call, then, when I dial the DID for DISA, I can only get dead silence. The CLI shows DISA is started, but just hang there doing nothing, and the line is connected from the caller's record. I can make calls and receive calls without problem.
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo

Premium Member

said by davidnewt:

I tried to remove the Playback() before DISA call, then, when I dial the DID for DISA, I can only get dead silence. The CLI shows DISA is started, but just hang there doing nothing, and the line is connected from the caller's record.
I ran out of idea, unfortunately. If you find the solution, please kindly post here.
davidnewt
join:2009-08-10
Conroe, TX

1 edit

davidnewt

Member

Thanks anyway. I suspect that the version for the router is not complete. I will try to use a full version on a computer to see if I have the same problem.
beaver
join:2006-03-12
Beaverton,OR

beaver

Member

So you decided to move to a computer?
I am reluctant to use a PC, but if I do, possibly I will try sipsorcery local version first.
among sipsorcery, pbxes and voxalot. sipsorcery is the most powerful one. If it runs reliably, it is the best one. unfortunately, its reliability seems not as good as pbxes or voxalot. I hope its local version doesn't have the reliability issue, at least better than the the main server.
davidnewt
join:2009-08-10
Conroe, TX

davidnewt

Member

no, I will not switch to PC for regular use. I am still playing Asterisk. But right now, I have difficulties to make DISA work for the version installed on router.
davidnewt

davidnewt

Member

I finally made the DISA work by doing "ipkg upgrade" in optware shell and reinstall the sound files. So far so good. I will continue to explore Asterisk and try more functions. Thank everyone who give me help here!