Re: Inbound Calls On New SIP Trunk Fail After 20 Seconds

Michael Mertens · ‎04-05-2021

This is a new SIP Trunk terminating on a new CUBE. Outbound calls work fine, however, inbound calls fail after 20 seconds with the outside caller beaning able to hear the inside person, but the inside (C796X or CSF/Jabber) cannot hear the outside person.
i noticed RTP timers expiring in debug logs (attached), though it seems by looking at firewalls on either side, both CUBE-to-PSTN and CUBE-to-inside phone shows tx and rx bytes. Also, I notice the call goes in a Call State=3 (Connected) but never goes Call State=4 (active). I don't believe it's a routing issue since outbound calls work fine. (H323 GW on the inside, and IPT end-points are registered to CUCM).

I have a 2nd CUBE to the same provider (both new) and calls work correctly in both directions. I've stared/compared SBC, H323 and firewall but can't find any difference.

Thanks for your help!!! Please see config snippet below:

voice service voip
ip address trusted list
ipv4 10.0.0.0 255.0.0.0
ipv4 12.194.41.30
ipv4 12.194.107.150
ipv4 12.253.1.102
ipv4 12.194.107.142
rtp-port range 16384 32766
address-hiding
mode border-element
media disable-detailed-stats
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip handle-replaces
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem relay nse codec g711ulaw gw-controlled
h323
h225 timeout t302 15
no h225 timeout keepalive
h225 timeout ntf 200
h225 signal overlap
h225 display-ie ccm-compatible
sip
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
header-passing
error-passthru
asserted-id pai
no update-callerid
early-offer forced
midcall-signaling passthru
privacy-policy passthru
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8 bytes 30
!
voice class codec 2
codec preference 1 g729r8 bytes 30
codec preference 2 g711ulaw
!
voice class codec 3
codec preference 1 g711ulaw
!
voice class h323 1
h225 timeout tcp establish 2
h225 timeout setup 2
call preserve
!
!
voice class sip-profiles 1
response ANY sip-header Allow-Header modify "UPDATE," ""
request INVITE sip-header Diversion modify "<sip:2(....)@" "<sip:845892\1@"
request INVITE sip-header Diversion modify "<sip:4(....)@" "<sip:845894\1@"
request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
request INVITE sdp-header Audio-Attribute add "a=ptime:30"
!
!
!
voice iec syslog
!
!
voice translation-rule 1
rule 1 /\+184589/ //
!
voice translation-rule 2
rule 1 /^[2]\(....\)/ /1845892\1/
rule 2 /^[4]\(....\)/ /1845894\1/
rule 3 /^9/ //
rule 4 /69302/ /18455926006/
!
!
voice translation-profile pstn-in
translate calling 1
translate called 1
!
voice translation-profile pstn-out
translate calling 2
translate called 2
!
!
!
!
voice-card 0/1
dsp services dspfarm
no watchdog
!
cts logging verbose
license udi pid ISR4431/K9 sn FOC24384KEV
no license smart enable
diagnostic bootup level minimal
!
spanning-tree extend system-id
!
!
!
!
redundancy
mode none
!
!
!
!
!
!
!
!
interface Loopback1
ip address 10.45.0.127 255.255.255.255
!
interface GigabitEthernet0/0/0
description NEFK300MDFEDGDW01-Eth1/34 Inside
ip address 10.45.4.97 255.255.255.248
negotiation auto
h323-gateway voip interface
!
interface GigabitEthernet0/0/1
description NEFK300MDFEDGDS01-Eth1/33 Outside
ip address 10.45.4.89 255.255.255.248
negotiation auto
!
interface GigabitEthernet0/0/2
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet0/0/3
no ip address
shutdown
negotiation auto
!
interface Service-Engine0/1/0
!
interface GigabitEthernet0/2/0
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet0/2/1
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
ip address 10.47.157.232 255.255.252.0
negotiation auto
!
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip tftp source-interface GigabitEthernet0
ip rtcp report interval 2000
ip route 10.0.0.0 255.0.0.0 10.45.4.102
ip route 12.0.0.0 255.0.0.0 10.45.4.49
ip route vrf Mgmt-intf 0.0.0.0 0.0.0.0 10.47.159.254
ip tacacs source-interface GigabitEthernet0

!
!
!
control-plane
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dspfarm profile 1 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g729r8
codec g729br8
maximum sessions 8
associate application CUBE
!
dspfarm profile 2 mtp
codec g711ulaw
maximum sessions hardware 8
associate application CUBE
!
dial-peer voice 101 voip
description Incoming from AT&T - AT&T Call Leg
huntstop
no modem passthrough
session protocol sipv2
incoming called-number 84589[24]....
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 151 voip
description Incoming from AT&T - VoIP/PBX Leg
preference 2
shutdown
destination-pattern [24]....
session target ipv4:10.45.0.128
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 152 voip
description Incoming from AT&T - VoIP/PBX Leg
preference 3
shutdown
destination-pattern [24]....
session target ipv4:10.45.0.129
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 200 voip
description Outgoing to AT&T -VoIP and PBX Call Leg
translation-profile outgoing pstn-out
no modem passthrough
session protocol sipv2
incoming called-number .
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax-relay ecm disable
fax-relay sg3-to-g3
fax rate disable
fax nsf 000000
fax protocol pass-through g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 201 voip
description Outgoing to AT&T -AT&T Call Leg Primary IPBE
translation-profile outgoing pstn-out
preference 1
destination-pattern 9T
no modem passthrough
session protocol sipv2
session target ipv4:12.253.1.102
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 202 voip
translation-profile outgoing pstn-out
preference 2
destination-pattern 9T
no modem passthrough
session protocol sipv2
session target ipv4:12.194.107.142
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 160 voip
description Test Phone B322
destination-pattern 26008
session target ipv4:10.10.52.2
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 150 voip
description Test DID
preference 1
destination-pattern 26006
session target ipv4:10.10.52.2
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 149 voip
description Test DID to PBX
preference 1
destination-pattern 26007
session target ipv4:10.45.0.128
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 147 voip
description Test DID to PBX
preference 1
destination-pattern 26009
session target ipv4:10.45.0.128
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 1200
!

Roger Kallberg · ‎04-05-2021

Any specific reason for why you use H.323 for the call leg to/from CM? It would be much cleaner if you would use SIP for this.

Roger Kallberg · ‎04-05-2021

You don’t have any bind statements on the dial peers to/from CM, so they would use the global configuration and there you are binding it to the external interface. Change that and see if it would solve your problem.

voice service voip
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0

Michael Mertens · ‎04-06-2021

Roger,

Thank you for the replies...I'm using H323 on the inside as that is what all are other gateways are- they all are ISDN-PRI on the PSTN side still...this is our first SIP PSTN connection(s). So if I bind the inside interfaces to SIP, there's no call ringing or connection at all- just reorder tone. I backed out those commands and binded my G0/0/0 (inside H323) interface to it's own IP address (which I had earlier) and I have the same results. Oddly, I have another new CUBE with another new SIP trunk (exact configurations with obviously different IP addresses) and that works fine.

Thanks for your input and time...

Roger Kallberg · ‎04-06-2021

Instead of having a global bind the optional and in my view preferred way to do this would be to specify the bind on each dial peer for the interface it should use. This applies to both inbound and outbound dial peers.

So for the DPs that is used for the connection to the ITSP use these lines.
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1

And for the DPs that is used for the connection for CM use these lines.
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0

About using SIP for the inside call path, it's a quite small thing to switch from H.323 to SIP. It's more or less a matter of doing these steps.

Create a SIP trunk in CM for the connection to the gateway. This replaces the H.323 gateway in CM. Configuration is quite similar for the key elements, aka call routing and so on.
Put the new trunk into the RG used for the H.323 GW, so that you don't need to update outbound call routing.
On the dial peers that points to/from CM set the protocol to SIP with this command session protocol sipv2 and set the dtmf relay to dtmf-relay rtp-nte sip-kpml.

That should pretty much be it.

Scott Leport · ‎04-07-2021

In addition to my message yesterday Michael, from looking at the statistics of the BYE message, it looks like there are no packets sent or received for the call. I know it's not an end-to-end SIP call, but I would have expected to have seen some statistics at the end of the call if there was RTP being transmitted:

008174: *Apr 5 18:06:15.105: //14320/61BB683BB856/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:12.253.1.102:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.45.4.89:5060;branch=z9hG4bK262121
From: <sip:8455926008@10.45.4.89>;tag=EE30A41-25C6
To: "WIRELESS CALLER" <sip:5189254753@12.253.1.102:5060>;tag=14848802724159682_c3b05.1.1.1616489021152.0_1327156_2643026
Date: Mon, 05 Apr 2021 18:05:47 GMT
Call-ID: 16380666200145255@c3b05_1_1
User-Agent: Cisco-SIPGateway/IOS-16.9.5
Max-Forwards: 70
Timestamp: 1617645975
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=19
Session-ID: 00000000000000000000000000000000;remote=2f175dd0524355d58766a854f7da5846
Content-Length: 0

As already mentioned, it's best to go fully SIP here. If that's not the way you want to go at the moment then compare your CUCM H.323 config settings between this site and your working site.

Also worth running packet captures from your CUBE / SBC / Gateway and Firewall when running a test call to see what comes of it.

Scott Leport · ‎04-06-2021

Hi Michael,

Its better to go fully SIP in this circumstance. You'll have much less issues going forward. Assuming you didn't want to necessarily blow away your H.323 configuration in CUCM right away, you could create a new SIP trunk in parallel, but targeting a different interface on your gateway, physical or loopback and then reconfigure CUCM Route Group / Route List / Route Patterns to suit.

As for the issue, I see the Q.850 code in the SIP debugs is 86. Only similar issue I have seen to this is on this thread. Looks like the exact same issue, but the topology is different. You may also want to compare the CUCM H.323 settings between your working and non-working site.

https://community.cisco.com/t5/ip-telephony-and-phones/dropped-incoming-calls-after-20-seconds-reason-q-850-cause-86/td-p/2466679

Michael Mertens · ‎04-09-2021

Scott,
Very interesting observation on the PS/PR...I'm pretty new to SIP trunking so didn't realize those counters were there. The odd thing is that both sides here conversation....you can talk but only for 19 seconds....I've opened up a TAC ticket. This CUBE has the same code level as the other, working CUBE. I've even tried a reboot on this one...the dreaded Hail Mary.

Lastly, I went H323 on the inside because on the other side of the network is am H323-to-PBX integration VGW, though I guess I could do SIP right to it also.

Anyway, I thank you and Roger for the input (I also tried the bind command to no effect). I have opened a case with Cisco TAC. I'll let you know how I make out- when I get back from vacation.

Thanks!