06-19-2019 08:35 AM
Hello Experts,
We are using the Cisco 2911 router for VoIP solutions and it was working fine for years but yesterday suddenly all outgoing and incoming calls start getting cranky but internal calls are fine.
I already checked logs and sh processes and everything is normal and there was no change done by us.
Please advise what to troubleshoot now?
Thanks
06-19-2019 08:55 AM
06-19-2019 09:28 AM
Hello Vaijanath,
Thanks for replying,
we are using a router as a CME and phones are registered to CME. There is no CUCM. We are using a PSTN or FXO line connected to a router and already ticket going on with them.
Is there any way, I can see how many PSTN lines are connected to the router and see other statistics on that,
Why should I need to restart the router ?
Thanks
06-19-2019 10:28 AM - edited 06-19-2019 10:28 AM
You can use "show voice port summary" command but this will show you operational status of FXO ports. You wont be able to confirm how many POTS lines are connected until to physically check the router.
The reason I mentioned to restart the router because you mentioned in your post that issue was started suddenly and I am assuming that you did not perform any configuration change or network change. Sometimes restarting the router resolves the issue. If restart doesn't work then you could try upgrading the rotuer IOS as older IOS can cause issues due to software bugs.
06-21-2019 06:10 AM
Hello Vaijanath,
I rebooted the phone server but no improvement.
I have one question, when new Cisco 2911 router is connected with PRI or FXO line do we need any configuration on that port?
If no then how cisco 2911 route all calls to that line and if yes please share how to set up that?
Thanks,
06-21-2019 07:51 AM
"rebooted phone server" you mean Cisco 2911 CME Router. Am I right?
Yes, you need PRI or FXO line configuration on new router.
Below is the PRI line configuration using SIP Protocol:
(Please note that you will have to update port-numbers, IP addresses etc. from below configuration and you may require additional configuration such as translation rules, translation profiles etc. to complete your dial plan.)
! card type t1 0 0
!
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
! isdn switch-type primary-ni ! voice call send-alert voice rtp send-recv ! voice service voip ip address trusted list ipv4 0.0.0.0 0.0.0.0 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer supplementary-service media-renegotiate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip bind control source-interface GigabitEthernet0/0/0 bind media source-interface GigabitEthernet0/0/0 header-passing error-passthru registrar server expires max 600 min 60 midcall-signaling passthru privacy-policy passthru ! voice class codec 1 codec preference 1 g711ulaw ! voice-card 0/1
dsp services dspfarm
!
controller T1 0/0/0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24 voice-dsp
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn incoming-voice voice
!
dial-peer voice 4000 voip
description # Calls To and From CUCME #
destination-pattern 4...$
session protocol sipv2
session target ipv4:10.0.1.100
incoming called-number 9.T
voice-class codec 1
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 9000 pots
description # For Inbound Calls from PSTN #
incoming called-number .
direct-inward-dial
port 0/0/0:23
!
dial-peer voice 9001 pots
description # For Outbound 911 Emergency Calls #
destination-pattern 911
progress_ind alert enable 8
progress_ind progress enable 8
no digit-strip
port 0/0/0:23
!
dial-peer voice 9002 pots
description # For Outbound 7-Digit Local PSTN Calls #
destination-pattern 9[2-9]......
progress_ind alert enable 8
progress_ind progress enable 8
forward-digits 7
port 0/0/0:23
!
dial-peer voice 9003 pots
description # For Outbound 10-Digit Local PSTN Calls #
destination-pattern 9[2-9]..[2-9]......
progress_ind alert enable 8
progress_ind progress enable 8
forward-digits 10
port 0/0/0:23
!
dial-peer voice 9004 pots
description # For Outbound 11-Digit Local PSTN Calls #
destination-pattern 91[2-9]..[2-9]......
progress_ind alert enable 8
progress_ind progress enable 8
forward-digits 11
port 0/0/0:23
!
dial-peer voice 9005 pots
description # For Outbound International PSTN Calls #
destination-pattern 9011T
progress_ind alert enable 8
progress_ind progress enable 8
prefix 011
port 0/0/0:23
!
Below is the FXO lines configuration using SIP Protocol:
! trunk group PSTN-FXO hunt-scheme round-robin ! voice-card 0 dsp services dspfarm ! voice call send-alert voice rtp send-recv ! voice service voip ip address trusted list ipv4 0.0.0.0 0.0.0.0 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer supplementary-service media-renegotiate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip bind control source-interface GigabitEthernet0/0/0 bind media source-interface GigabitEthernet0/0/0 header-passing error-passthru registrar server expires max 600 min 60 midcall-signaling passthru privacy-policy passthru ! voice class codec 1 codec preference 1 g711ulaw ! voice-port 0/1/0 trunk-group PSTN-FXO supervisory disconnect dualtone mid-call timeouts call-disconnect 2 connection plar opx 4000 caller-id enable ! voice-port 0/1/1 trunk-group PSTN-FXO supervisory disconnect dualtone mid-call timeouts call-disconnect 2 connection plar opx 4000 caller-id enable ! voice-port 0/1/2 trunk-group PSTN-FXO supervisory disconnect dualtone mid-call timeouts call-disconnect 2 connection plar opx 4000 caller-id enable ! voice-port 0/1/3 trunk-group PSTN-FXO supervisory disconnect dualtone mid-call timeouts call-disconnect 2 connection plar opx 4000 caller-id enable ! dial-peer voice 4000 voip description # Calls To and From CUCME # destination-pattern 4...$ session protocol sipv2 session target ipv4:10.0.1.100 incoming called-number 9.T voice-class codec 1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad ! dial-peer voice 9000 pots description # For Inbound Calls from PSTN # incoming called-number . direct-inward-dial ! dial-peer voice 9001 pots description # For Outbound 911 Emergency Calls # destination-pattern 911 no digit-strip trunkgroup PSTN-FXO ! dial-peer voice 9002 pots description # For Outbound 7-Digit Local PSTN Calls # destination-pattern 9[2-9]...... forward-digits 7 trunkgroup PSTN-FXO ! dial-peer voice 9003 pots description # For Outbound 10-Digit Local PSTN Calls # destination-pattern 9[2-9]..[2-9]...... forward-digits 10 trunkgroup PSTN-FXO ! dial-peer voice 9004 pots description # For Outbound 11-Digit Local PSTN Calls # destination-pattern 91[2-9]..[2-9]...... forward-digits 11 trunkgroup PSTN-FXO ! dial-peer voice 9005 pots description # For Outbound International PSTN Calls # destination-pattern 9011T prefix 011 trunkgroup PSTN-FXO !
06-21-2019 12:07 PM
Hello Vaijanath,
Thanks for your help, I got an update from AT&T and they said its phone server. I did not change anything and it was working from the last two years. Please let me what else should I investigate?
One question, Auto attendant is also cracky it's not just calls.
Thanks
06-22-2019 07:32 PM
06-21-2019 12:58 PM
Hello Vaijanath,
what does these commands means,
bind control source-interface GigabitEthernet0/0/0 bind media source-interface GigabitEthernet0/0/0
voice register global
mode cme
source-address 10.6.4.2 port 5060
no outbound-proxy
max-dn 25
max-pool 25
load 7841 sip78xx.11-0-1-11
olsontimezone America/Chicago version 2013g
timezone 8
hold-alert
voicemail 399
tftp-path flash:
create profile sync 0012503726981739
ntp-server 132.146.11.238 mode unicast
Thanks
06-22-2019 07:36 PM
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide