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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
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Cisco ATA 186 SIP and Asterisk - HowTo

ATA186 configuration.


The ATA configuration is well described by Cisco you will find pointers on the following page
Asterisk phone cisco ATA18x

For a 5mn quick start do the following :

Connect your ATA on the Ethernet, it will get an IP address via the DHCP. Look in your message log to see which IP adress has been allocated
Connect with a web browser on your adaptor http://the-ata-ip-address/dev to get access to the configuration.
Set UID0: to your-login-line1 ; This will be reuse later in asterisk configuration as the end point ID (Channel in sip.conf)
Set PWD0: to the associated passwd
repeat for UID1 and PWD1 for line 2    ; Note the UID1 MUST be different from UID0

Set UseLoginID: to 0 ; If you want a loginID different of your end point ID set to 1 and fill in LoginID0: and LoginID1:
Set GkOrProxy: to the IP address of your Asterisk server (or SIP proxy/gateway)
Save your configuration.

You can see my ATA186 configuration in the joined file sip.ata186.conf.pdf





/etc/asterisk/sip.conf


[general]
 context=internal              ; Default context for incoming calls
 allowguest=yes                  ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
 bindport=5060                  ; UDP Port to bind to (SIP standard port is 5060)
 bindaddr=0.0.0.0              ; IP address to bind to (a good idea to restrict that to what you really want)
 srvlookup=no                    ; Enable DNS SRV lookups on outbound calls
 pedantic=no                     ; Enable slow, pedantic checking for Pingtel; This section declares a Cisco ATA with 2 analogue lines attached.

; This section declare the Cisco ATA186 Lines 1 and 2

[ata1-line1]
 type=friend
 secret=ata1-line1
 qualify=yes
 nat=no
 host=dynamic
 caninvite=no
 context=internal

[ata1-line2]
 type=friend
 secret=ata1-line2
 qualify=yes
 nat=no
 host=dynamic
 caninvite=no
 context=internal

; This section declares 4 SIP phone runing on PC client (e.g kphone) Useful for test purpose

[phone1]
 type=friend
 secret=phone1
 ;qualify=yes
 nat=no
 host=dynamic
 caninvite=no
 context=internal

[phone2]
 type=friend
 secret=phone2
 ;qualify=yes
 nat=no
 host=dynamic
 caninvite=no
 context=internal




/etc/asterisk/extension.conf

 
 [internal]                                ; As declared in /etc/mgcp.conf calls coming from the ATA will be presented with the context.
 ;
 include => emergency               ; ANY good numbering plan SHOULD have an entry for emergency services in any context.
 include => your-extra-context      ; Add here any context(s) that you need/want to

 ; Integrate a Cisco ATA186 configured in SIP mode.
 ;Integrate a SIP Cisco ATA 186 (see related conf.pdf)
 exten => 111,1,Dial(SIP/ata1-line1,45,o)                  
 exten => 112,1,Dial(SIP/ata1-line2,45,o)

 ;for info integrate a MGCP Cisco ATA 186 (see related conf pdf)
 exten => 121,1,Dial(MGCP/aaln/1@000e83e530ae,45,o)          ; dial end point aaln/1 in MGCP Channel (see mgcp.conf) 000e83e530ae, rings to to 45s , pass the CallerID as received in the context.
 exten => 122,1,Dial(MGCP/aaln/2@000e83e530ae,45,o)          ; dial end point aaln/2


You are done.



See Also


Created by Dominique Le Foll, Last modification by Dominique Le Foll on Tue 13 of Jun, 2006 [17:40 UTC]

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