09-30-2015 09:07 AM - edited 03-17-2019 04:26 AM
I have a Cisco 2911 i have conf with pstn provider
did translation and created a trunk sip an sent extension to IVR server
but from GSM number when i dial the ivr number it drops like after 8 or more seconds
but when i call directly from a cisco ip phone it passes with no drops
what could be there cause for this
09-30-2015 10:18 AM
What is the exact call flow when you dial number from GSM?
What is the trunk type in gateway?
Do you mean PSTN/GSM -> Cisco Gateway -> SIP Provider ??
10-01-2015 12:39 AM
Pstn provider=CME(2900)=Ivr
Is a trunk sip
10-01-2015 12:43 AM
I didn't get you call flow. Which trunk you've in CME for PSTN callers to reach CME? FXO or PRI?
10-01-2015 04:00 AM
is a PRI
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bchan-number-order ascending
isdn sending-complete
no cdp enable
!
dial-peer voice 5077 pots
description "incoming Calls from CAMTEL-MTN-ORANGE'
translation-profile incoming INCOMING-CALLS-New
incoming called-number 5077
direct-inward-dial
10-01-2015 09:15 PM
Can you please share the output of debug isdn q931 and debug ccsip messages... Also share the output of show run.
Thanks
10-02-2015 03:29 AM
isdn switch-type primary-net5
!
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
no call service stop
sip
!
!
voice translation-rule 5
rule 1 /5077/ /2001/
!
voice translation-profile INCOMING-CALLS-New
translate called 5
!
voice-card 0
!
controller E1 0/1/0
clock source internal
pri-group timeslots 1-31
!
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bchan-number-order ascending
isdn sending-complete
no cdp enable
!
voice-port 0/1/0:15
!
dial-peer voice 5080 voip
description IVR
destination-pattern 2001
session protocol sipv2
session target ipv4:192.168.10.1
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 5077 pots
description "From GSM"
translation-profile incoming INCOMING-CALLS-New
incoming called-number 5077
direct-inward-dial
!
!
10-02-2015 03:31 AM
Hi,
I asked you for debugs (isdn q931) and ccsip messages...
10-02-2015 08:24 AM
10-02-2015 09:29 AM
Divine,
The asterisk server is dropping the call.. You need to investigate why on that side..
Oct 2 15:03:11: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:243223186@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK66e3e09c;rport
Max-Forwards: 70
From: <sip:2001@192.168.10.1>;tag=as277e686c
To: <sip:243223186@192.168.1.1>;tag=1C3C450-1E1F
Call-ID: 7BCE0AC6-684D11E5-932FDCE6-F497A57D@192.168.1.1
CSeq: 102 BYE
User-Agent: FPBX-12.0.76.1(11.19.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
You may also want to enable PRACK on both asterisk and your voice gateway.
10-05-2015 12:58 AM
How can this be enable
10-05-2015 01:51 AM
On CUBE you can do it globally or at the dial-pee level..
voice service voip sip rel1xx require 100rel
Dial-peer level:
dial-peer voice 1000 voip voice-class sip rel1xx require 100rel
I don't know how you enable it on asterisk..You will need to investigate this
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