PSTN to SIP server

[divine007]

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

[Vivek Batra]

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 ??

[divine007]

Pstn provider=CME(2900)=Ivr 

Is a trunk sip

[Vivek Batra]

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

[divine007]

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

[Vivek Batra]

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

Thanks

[divine007]

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

!

!

[Vivek Batra]

Hi,

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

[divine007]

Ok attached the debug logs respectively

[Ayodeji Okanlawon]

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.

[divine007]

How can this be enable

[Ayodeji Okanlawon]

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