cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1666
Views
25
Helpful
9
Replies

Cisco 2911/k9 Voice PBX router , all outgoing calls are cranking but internal calls are fine

lovejit1313
Level 1
Level 1

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 

9 Replies 9

Vaijanath Sonvane
VIP Alumni
VIP Alumni
Please provide below details:
1. Are you using this router as CME?
2. Are the phones registered to CME or CUCM?
3. Are you using PRI, FXO or SIP Trunk lines?
4. Did you check with your Service Provider?
5. Have you tried restarting the router?



Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

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

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.

 

 

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

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,

 

 

"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
!

 

 

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

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

Please try upgrading the IOS to the latest version.
Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

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

bind control source-interface GigabitEthernet0/0/0; this command binds SIP signalling to the Gig0/0/0.
bind media source-interface GigabitEthernet0/0/0; this command binds RTP media (audio) to the Gig 0/0/0.
Please check this URL for more details:
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_fund/configuration/xe-3s/cube-fund-xe-3s-book/voi-sip-bind.html

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: