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CISCO DID Voice Gateway (3600) PROBLEM

smy
Posts: 4
 Member Since:
2009-06-17
Submitted by smy on Thu, 06/18/2009 - 12:05am.

Hi everyone,

I have 2 separate problems with the following configuration, the infrastructure is extremely simple, yet I couldn't get it to work properly, please help....

1) INBOUND DID Calls from Telco to Asterisk..

CISCO 3600 (w/T1 DID from Telco) ===> LAN ===> Asterisk===>IP PHONE (ext 404)

I've already setup the cisco to pass all the DID calls to Asterisk via internal LAN (Thus no NAT), and through network packet inspection, the SIP signals got through to Asterisk with no problems. Yet when I inspect the log file @ "/var/log/asterisk/full", it says "Call from 'CISCO-3660' to extension '404' rejected because extension not found."

I have already setup proper inbound routing and trunk settings with the follow USER Details...
context=incoming
type=friend
host=192.168.11.1
nat=no
insecure=very
canreinvite=yes
qualify=yes
Disallow=all
Allow=ulaw&alaw
DTMFMode=rfc2833

ANY HELP???

As for problem 2)
IP PHone (404) ==> ASTERISK ===> LAN ===> CISCO(T1 Telco) {The same PRI is used for inbound and outbound}

Calls are made from an endpoint (Internal LAN) and got through to CISCO... the call actually completed by ringing and PSTN number, yet absolutely no audio on both ends after pickup. At first I thought it was the codec problem, but I have configured everything to restrict to G.711ULaw and still no luck. As for Endpoint, I've tried using xLite and other sorts of IP Phones, yet the result is still the same.

PS: I'm using CISCO IOS 12.2 if that's any help!

I've been searching all the related forums and done a healthy dose of reading on this issue, yet I couldn't really find any real answers which can help me sort this out. Perhaps some of you have a working scenario related to mine, I would love for you to share your experiences and a help hand.

Thanks

‹ CallerID Name Missing on DID calls IAX2 to ZAP route, ss-noservice ›

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SkykingOH
Posts: 9541
 Member Since:
2007-12-17
There is no context
Thu, 06/18/2009 - 8:29am

There is no context incoming. If you want to be able to route the calls with inbound routes the context is from-trunk to have access to internal extensions without routing the context is from-internal.

--

Scott

aka "Skyking"


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smy
Posts: 4
 Member Since:
2009-06-17
Thanks for the pointer
Thu, 06/18/2009 - 9:33pm

Thanks for the pointer Scott!! But it doesn't seem to work even after I changed the configuration! Anything else that I might missed in the configuration? Perhaps something that I might need to pay attention to?


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SkykingOH
Posts: 9541
 Member Since:
2007-12-17
I have no way of knowing you
Thu, 06/18/2009 - 11:27pm

I have no way of knowing you need to post your 'show run' and the peer settings from your trunk so I can look them over. Please use the [code]tags[/code] to make it readable.

--

Scott

aka "Skyking"


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smy
Posts: 4
 Member Since:
2009-06-17
Here's my Cisco config (..
Fri, 06/19/2009 - 10:39am

Here's my Cisco config (.. with some irrelevant settings excluded!)

Current configuration : 4851 bytes
!
! Last configuration change at 11:16:15 UTC Fri Jun 19 2009
! NVRAM config last updated at 11:16:43 UTC Fri Jun 19 2009
!
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Jervois
!
enable secret 5 $1$jSYz$BB.gdfpmqC7h.nLTffaZ//
!
memory-size iomem 10
voice-card 4
!
ip subnet-zero
!
!
ip name-server 202.180.160.1
!
ip cef
isdn switch-type primary-dms100
isdn voice-call-failure 0
!
!
voice call send-alert
voice rtp send-recv
!
voice service pots
!
voice service voip
 signaling forward unconditional
 sip
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
!
fax interface-type fax-mail
mta receive maximum-recipients 0
!
controller T1 4/0
 framing esf
 clock source line primary
 linecode b8zs
 pri-group timeslots 1-24
 no yellow generation
 no yellow detection
!
controller T1 4/1
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
 no yellow generation
 no yellow detection
!
interface FastEthernet0/1
 ip address 192.168.11.1 255.255.255.0
 ip nat inside
 duplex auto
 speed auto
!
interface Serial4/0:23
 no ip address
 no logging event link-status
 isdn switch-type primary-dms100
 isdn incoming-voice voice
 isdn map address . plan unknown type unknown
 isdn send-alerting
 isdn sending-complete
 fair-queue 64 256 0
 no cdp enable
!
interface Serial4/1:23
 no ip address
 no logging event link-status
 isdn switch-type primary-4ess
 isdn incoming-voice voice
 no cdp enable
!
ip classless
no ip http server
!
call rsvp-sync
!
voice-port 4/0:23
 disc_pi_off
 no comfort-noise
 cptone HK
 bearer-cap Speech
!
voice-port 4/1:23
 translate calling 1
 translate called 1
!
mgcp profile default
dial-peer cor custom
!
dial-peer voice 3 pots
 description PSTN_INCOMING_MATCHING
 incoming called-number .T
 direct-inward-dial
 port 4/0:23
!
dial-peer voice 4 pots
 description PSTN_OUTBOUND
 huntstop
 destination-pattern .T
 port 4/0:23
 forward-digits all
!
dial-peer voice 5 voip
 huntstop
 destination-pattern ...
 progress_ind setup enable 3
 progress_ind progress enable 8
 signaling forward unconditional
 voice-class codec 1
 session protocol sipv2
 session target ipv4:192.168.11.23
 session transport udp
 dtmf-relay rtp-nte
 ip qos dscp cs7 media
 no vad
!
dial-peer voice 6 pots
 incoming called-number .T
 direct-inward-dial
 port 4/0:23
!
gateway
!
sip-ua
 sip-server ipv4:192.168.11.23:5060
!
!
line con 0
line aux 0
line vty 0 4
!
ntp server 131.215.254.2 prefer
!
end

While the following is my PEER settings for the trunk......

context=to-pstn
type=peer
host=192.168.11.1
nat=no
qualify=yes
insecure=very
canreinvite=yes
Disallow=all
Allow=ulaw&alaw

and the following is my USER Details....

context=ext-did
type=peer
host=192.168.11.1
nat=no
insecure=very
canreinvite=yes
qualify=no
Disallow=all
Allow=ulaw&alaw
DTMFMode=rfc2833

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smy
Posts: 4
 Member Since:
2009-06-17
I've just sorted out the
Sun, 06/21/2009 - 7:42pm

I've just sorted out the inbound "Extension not found" problem by trial & error!! I basically swiched the DTMFMode from rfc2833 to inband, then all inbound calls are now able to connect. I'm not sure if this is the proper way to sort it out, but it works.... is there a better way to do this? any comments?

Yet the 2-way audio problem still exists, any ideas?


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