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No Voice-call signaling - secondary router

Symptom: Incoming calls ring on the phone but user cannot 'answer' the line, it just keeps ringing.  Then spits this error: VOICE_IEC-3-GW: SIP: Internal Error (ACK wait timeout): IEC=1.1.129.7.67.0 on callID 80063 GUID=CC0DCEB1C5A411E8B7C2DD4A89C36D36

 

Setup: 2921 Primary router/circuit - FXO Card with analog line connected as inbound.  HSRP is setup to fallback to secondary router with a secondary circuit and tunnel back to CallManager site.  We've tried an 819, 891 and a 2901 router as the secondary with the same results, however when we use a 4331 router calls process just fine.

 

We have more than one site having the same issue, all our sites with 4331 routers do not have the issue.  Are we running into a hardware limitation of some kind or do the 2900 series routers require a different configuration?  I read about HA CUBE configuration but that requires two like for like routers which we don't have at some of our sites.

 

Call flow: PSTN à FXO à 2921 àbackup router àDMVPN àSIPàCUCM à AA(dial-by-ext)à IP phone

 

voice service voip
ip address trusted list
ipv4 x.x.x.x x.x.x.x
allow-connections sip to sip
fax protocol t38 nse version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
modem passthrough nse codec g711ulaw
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
rel1xx disable
header-passing
registrar server

 

The call hits this dial-peer to route to auto-attendant:

 

dial-peer voice 1001 voip
description ** Outbound call leg to UCM **
preference 1
destination-pattern +xxxxxxx....$ (removed actual digits for privacy)
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip options-keepalive
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte sip-kpml
ip qos dscp cs3 signaling
no vad

 

Again - all incoming calls work correctly while on the primary router or if we use a 4331 as a secondary.

 

Thanks.

 

 

 

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Beginner

Re: No Voice-call signaling - secondary router

I may have stumbled into a fix for this; posting for others to reference.

 

The primary dial-peer for CUCM call leg was configured as you see below:

 

dial-peer voice 1001 voip

description ** Outbound call leg to UCM **

preference 1

destination-pattern REMOVED

session protocol sipv2

session target ipv4: REMOVED

voice-class codec 1

voice-class sip options-keepalive

voice-class sip bind control source-interface Loopback0

voice-class sip bind media source-interface Loopback0

dtmf-relay rtp-nte sip-kpml

ip qos dscp cs3 signaling

no vad

 

By removing the 'rtp-nte' from the dtmf-relay, it allowed calls to be answered while routing through the secondary router.  Not sure why that would fix my issue but there it is.