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incoming calls from TELCO fail via my E1 line

m.gnahoui
Level 1
Level 1

Hi Cisco community,

I have a problem with calls coming from outside via my E1 line.
Outgoing calls from my network go through while incoming calls fail.

TELCO (E1 with SDAs) --> CUBE (ISR 4331) --> CUCM (version 12.5)

I tried to contact the TELCO but they told me no problem on their side. Is the error level? Help me please. I can't identify the problem.

When I examine, I manage to see at the debug isdn level, **error**: l3_badpeermsg and the debug voip ccapi Disconnect Cause=102.

mgnahoui_0-1709808470948.png

Attached are the debug files and my full configuration

Any feedback/guidance you can provide would be greatly appreciated

 

1 Accepted Solution

Accepted Solutions

You have to check the CUCM logs to dig deeper.
The issue can be a lot of things.

The working call from CUCM to CUBE uses TCP.
But the non-working call from CUBE to CUCM uses UDP. Maybe you should configure the dial-peer to use TCP too.
"session transport tcp"

View solution in original post

6 Replies 6

b.winter
VIP
VIP

Can you post the output of the following debugs:
debug isdn q931
debug ccsip messages
debug voice translation
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

m.gnahoui
Level 1
Level 1

Hi @b.winter , see below.

The IP-address 10.205.0.21 (CUCM ?) is not responding to the INVITE and therefore, CUBE resends the INVITE mulitple times.

Since after a certain amount of time the PSTN provider doesn't receive anything back from CUBE, the provider hangs up the call
bwinter_0-1709810267054.png

 

m.gnahoui
Level 1
Level 1

CUCM is 10.205.0.21. What could be the cause and how to fix it ? Outgoing calls work well though.

You have to check the CUCM logs to dig deeper.
The issue can be a lot of things.

The working call from CUCM to CUBE uses TCP.
But the non-working call from CUBE to CUCM uses UDP. Maybe you should configure the dial-peer to use TCP too.
"session transport tcp"

Thank you so much. I was able to solve it by adding this :

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!

And I applied the "session transport tcp" and "voice-class codec 1" to all dial-peer voice XXX voip

dial-peer voice 121 voip
corlist outgoing APPEL-INTERNE
description description PHONE USER 1...
destination-pattern 1...
session protocol sipv2
session transport tcp
session target ipv4:10.205.0.21
dtmf-relay rtp-nte sip-kpml sip-notify
voice-class codec 1
no vad
!