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Beginner

PSTN to SIP server

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

11 REPLIES 11
Engager

What is the exact call flow

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 ??

Beginner

Pstn provider=CME(2900)=Ivr

Pstn provider=CME(2900)=Ivr 

Is a trunk sip

Engager

I didn't get you call flow.

I didn't get you call flow. Which trunk you've in CME for PSTN callers to reach CME? FXO or PRI?

Highlighted
Beginner

is a PRI interface Serial0/1

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

 

 

Engager

Can you please share the

Can you please share the output of debug isdn q931 and debug ccsip messages... Also share the output of show run.

Thanks

Beginner

isdn switch-type primary-net5

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

!

!

Engager

Hi,I asked you for debugs

Hi,

I asked you for debugs (isdn q931) and ccsip messages...

 

Beginner

Ok attached the debug logs

Ok attached the debug logs respectively

VIP Mentor

Divine,The asterisk server is

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.

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Beginner

How can this be enable

How can this be enable

VIP Mentor

On CUBE you can do it

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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