11-27-2015 12:25 AM - edited 03-17-2019 05:02 AM
Hello.
Gateway router 2911 have got e1 to pstn, 4fxs module and sip connection to internal voice network. Problem is that its unable to make outgoing calls from analog telephones, connected to fxs ports in any directions. After dialing first digit got fastbusy tone.
Incoming calls from pstn or internal network works fine. Parts of config listed below
card type e1 0 0
!
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
!
!
!
isdn switch-type primary-net5
!
voice-card 0
!
!
!
voice service pots
!
voice service voip
address-hiding
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
h323
no call service stop
sip
early-offer forced
midcall-signaling passthru
!
!
hw-module pvdm 0/0
!
!
controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-10,16
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
!
voice-port 0/0/0:15
!
voice-port 0/1/0
cptone RU
timeouts initial 5
timeouts interdigit 2
caller-id enable
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
!
!
!
!
!
mgcp profile default
!
!
dial-peer voice 1 pots
!
dial-peer voice 2 pots
description E1
translation-profile incoming 2
translation-profile outgoing 2
destination-pattern 8.
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
direct-inward-dial
port 0/0/0:15
forward-digits all
!
dial-peer voice 5 pots
description MODEM1
destination-pattern 722900
port 0/1/0
!
dial-peer voice 11 voip
destination-pattern .
session protocol sipv2
session target ipv4:172.28.241.251
dtmf-relay cisco-rtp h245-alphanumeric h245-signal rtp-nte
codec g711alaw
no vad
11-27-2015 12:36 AM
Hi Evgeny,
Could you provide the below debugs for one test call from analog phone along with the call details to check and also the translation profile configuration.
debug voip ccapi inout
debug isdn q931
Thanks
Rajan
11-27-2015 12:55 AM
All calls, except outgoing from fxs, works fine.
for exaple call from internal network from sip trunk to pstn e1 works fine. There is no problem with connection to pstn or any other routing.
Problem only with outgoing fxs call - got fast busy after i dial only 1 digit on analog telephone
11-27-2015 01:36 AM
Can you share the below debugs for one failed call
debug voip ccapi inout
debug isdn q931
11-27-2015 10:16 AM
Hi,
you can start adding cptone RU under fxs Voice ports.
Than activate a debug vpm signal and test a call.
Please post the output of the debug here.
Thanks
Regards
Carlo
11-27-2015 12:38 AM
Also let us know the below:
Whether the call from IP phone to PRI works or not ?
Prefix 8 needs to be sent to the provider or needs to be stripped and sent out. As per your config, 8 is sent now.
11-30-2015 03:20 AM
The problem is solved -
dial-peer destination pattern need "T" in end for calls from FXS.
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