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Voice call flow working just one direction in VOIP CUCM to POTS CME

gil95
Beginner
Beginner

Greetings everyone,

For weeks now that I’m struggling with the problem above. We have, as an institution a CUCM cluster in Headquarters connected through SIP trunks to our branches CME IOS Gateways.

 This specific branch we have a more complex schema as we had hardware restrictions:

  • Branch Phones are registered in CUCM
  • On-net calls work fine
  • CUCM has a SIP trunk to the branch Router 1751 via VPN
  • Branch IOS router serves as Gateway to PTSN with a FXO access

Problem description:

  • Either trying to call PTSN -> On-Premises or the other way around, audio doesn’t come in but can go out, i.e, we can’t listen to outsiders.

Configurations for the R1751 dial-peer and voice-port:

voice-port 0/0

 no battery-reversal

 disc_pi_off

 cptone PT

 timeouts interdigit 2

 timeouts call-disconnect 3

 timeouts wait-release 3

 timing hookflash-out 50

 timing guard-out 1000

 connection plar 4XXXXX //Number registred in CUCM that serves that Branch reception

 impedance complex2

!

voice-port 0/1

 impedance complex2

!

dial-peer voice 110 voip

 tone ringback alert-no-PI

destination-pattern [2-6].....

 progress_ind setup enable 3

 progress_ind progress enable 2

 progress_ind connect enable 2

 voice-class codec 1

 session protocol sipv2

 session target ipv4:10.c.c.c //CUCM Subscriber

!

dial-peer voice x25365514 pots

incoming called-number x25365514

 destination-pattern [2,9]........

 progress_ind setup enable 3

 progress_ind alert enable 8

 progress_ind progress enable 8

 progress_ind connect enable 8

 port 0/0

 forward-digits 9

!

 

This is some rtp debug for an On-Premisses to PTSN that I think indicates the problem since the  RTP streams IP doesn´t make sense. Traces are collected from the R1751 and two aspects must be noticed:

  1. For the tx stream destination IP is correct (IP phone registered in CUCM), but source IP missing
  2. For the rx stream destination IP is still the IP phone when should be the router, source IP missing
  3. rtp session is created as I believe correctly, 10.b.b.b (CME 1751 router) + 10.a.a.a (CUCM phone that makes the call)

 

Jan 13 07:17:28.291: voip_rtp_create_session: callID=11, dstCallID=12 laddr=10.b.b.b, lport=17052,raddr=10.a.a.a, rport=22032, type=3, sig_tos=3, ip_tos=5

Jan 13 07:17:28.291: voip_rtcp_get_cname: cname=0.0.0@10.b.b.b

Jan 13 07:17:28.291:  voip_rtp_update_xmit_info

Jan 13 07:17:28.291:  voip_rtp_update_xmit_info, dstvdbptr: 825F83FC, dstCallID 12, gccb: 81532334, xmitFunc 8027E1D8,context 824530F4

Jan 13 07:17:28.291:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 825F83FC, xmit_info->dstCallID 12, xmit_info->xmitFunc 8027E1D8, xmit_info->context 824530F4

Jan 13 07:17:28.295:  voip xmit info count: 1

Jan 13 07:17:28.295: voip_rtp_set_non_rtp_call: Non-RTP call end

Jan 13 07:17:28.295: voip_rtp_exchange_context_info

Jan 13 07:17:28.295:  voip_rtp_update_xmit_info

Jan 13 07:17:28.295:  voip_rtp_update_xmit_info, dstvdbptr: 825F83FC, dstCallID 12, gccb: 81532334, xmitFunc 8027E1D8,context 824530F4

Jan 13 07:17:28.295:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 825F83FC, xmit_info->dstCallID 12, xmit_info->xmitFunc 8027E1D8, xmit_info->context 824530F4

Jan 13 07:17:28.295:  voip xmit info count: 1

Jan 13 07:17:28.295: voip_rtp_exchange_context_info

Jan 13 07:17:28.295:  voip_rtp_update_xmit_info

Jan 13 07:17:28.295:  voip_rtp_update_xmit_info, dstvdbptr: 825F83FC, dstCallID 12, gccb: 81532334, xmitFunc 8027E1D8,context 824530F4

Jan 13 07:17:28.295:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 825F83FC, xmit_info->dstCallID 12, xmit_info->xmitFunc 8027E1D8, xmit_info->context 824530F4

Jan 13 07:17:28.299:  voip xmit info count: 1

Jan 13 07:17:28.299: voip_rtp_exchange_context_info

Jan 13 07:17:28.299:  voip_rtp_update_xmit_info

Jan 13 07:17:28.299:  voip_rtp_update_xmit_info, dstvdbptr: 825F83FC, dstCallID 12, gccb: 81532334, xmitFunc 8027E1D8,context 824530F4

Jan 13 07:17:28.299:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 825F83FC, xmit_info->dstCallID 12, xmit_info->xmitFunc 8027E1D8, xmit_info->context 824530F4

Jan 13 07:17:28.299:  voip xmit info count: 1

Jan 13 07:17:28.303: voip_rtcp_start_session:

Jan 13 07:17:28.303: voip_rtcp_start_session: start session

Jan 13 07:17:28.315: RTP(101): fs tx d=10.a.a.a(22032), pt=8, ts=3F20, ssrc=24D204FD

Jan 13 07:17:28.335: RTP(102): fs tx d=10.a.a.a(22032), pt=8, ts=3FC0, ssrc=24D204FD

Jan 13 07:17:28.355: RTP(103): fs tx d=10.a.a.a(22032), pt=8, ts=4060, ssrc=24D204FD

 

----------(….)-------------

Jan 13 07:17:37.167: RTP(64628): ps rx d=10.a.a.a(22032), pt=8, ts=41562816, ssrc=F0E34107

Jan 13 07:17:37.175: RTP(255): fs tx d=10.a.a.a(22032), pt=8, ts=15400, ssrc=24D204FD

--------------------

MMP-R1751-VGW-FXO#sh voice call status
CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers
0x3C 1 0x825F83FC 0/0 0/1 *933333333 g711alaw 110/225365514
1 active call found

MMP-R1751-VGW-FXO#sh voice dsp

DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT
==== === == ======== ======= ===== ======= === == ========= == ===== ============
C549 000 00 g711alaw 4.1.41 Busy Idle 0 0 0/0 0 0 842135/30073

C549 000 01 g711ulaw 4.1.41 Idle Idle 0 0 0/1 NA 0 10/6

Active Voice Call details

C549 000 00 g711alaw 4.1.41 Busy Idle 0 0 0/0 0

Current total analog signalling channels: 2
Current max allowed digital timeslot for voice: 30
Current number of DSP group: 1
Group 0:
Current allocated analog signalling channels: 2
Current free analog signalling channels: 0
Current allocated digital signalling channels: 0
Current free digital signalling channels: 30
Port(s) served: 0/0 0/1
Current Available MIPS: 550
SPMM DSPRM State Image D-sig D-sig A-sig A-sig Mips Voice/Xcode
Dsp Dsp allocate free allocate free Free Chan
0/0 0 UP FIXHC 0 0 2 0 50 1
0/1 1 UP FLEX6 0 6 0 0 100 0
0/2 2 UP FLEX6 0 6 0 0 100 0
1/0 3 UP FLEX6 0 6 0 0 100 0
1/1 4 UP FLEX6 0 6 0 0 100 0
1/2 5 UP FLEX6 0 6 0 0 100 0

 

What did I went through:

  • Telephone web GUI show sent rtp packets but receives none
  • Messing with MRGL to restrict the resources used to the subscriber used in the trunk
  • Changing dial-peers config

 

I'm in a point without ideas, hopping the experts can help me, or tell me that it ain't possible because of router and card used for the FXO access (WIC/VIC-2FXO-EU)

[Edit]

R1751 config and requested debug for a call test attached. CUCM traces give Normal call clearing.

10.a.a.a - CUCM calling Phone IP
10.b.b.b - GW R1751
10.c.c.c - Subscriber
10.d.d.d - Publisher
411111 - CUCM Calling Number
93333333 - Called Number