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:
- For the tx stream destination IP is correct (IP phone registered in CUCM), but source IP missing
- For the rx stream destination IP is still the IP phone when should be the router, source IP missing
- 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
