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yamikani2g2
Beginner

SIP phone Register fail - Phones on a different subnet

Good day experts.

 

I have two sites and would like to share my CME now turned to CUBE. I have the 2 Voice and Data Networks in HQ. 192.168.1.X and 192.168.10.X. The 192.168.10.X is voice and the TFTP server is 192.168.10.1.

Phones on my HQ are working they can register can call in and out the struggle was real but alas all worked.

 

I want some workers from a remote branch B to have their phones register they have DHCP there and have the following IP address Data(192.168.2.X) and  Voice ( 192.168.3.X). The PCs get ip addresses and work the phones get IP address and i can reach them from 192.168.10.X network and vice versa. Basically there is end to end IP reacheability.

 

Issue is the remote phones can not get TFTP 192.168.10.1 its not loading through option 150. Below is the debug message I get remember these phones loaded and worked and HQ take them to Branch B we get this. ( 4040 is a registered extension....

 

Nov 21 06:53:39.437: VOICE_REG_POOL: No entry for (4044) found in contact table
*Nov 21 06:53:39.449: VOICE_REG_POOL: No entry for (4044) found in contact table
*Nov 21 06:53:39.485: VOICE_REG_POOL: No entry for (4044) found in contact table
*Nov 21 06:53:39.485: voice_reg_get_reg_expires_timer: no voice register pool fo
und
*Nov 21 06:53:39.485: VOICE_REG_POOL: Register request for (4044) from (192.168.
3.100)
*Nov 21 06:53:39.945: VOICE_REG_POOL: No entry for (4044) found in contact table
*Nov 21 06:53:39.957: VOICE_REG_POOL: No entry for (4044) found in contact table
*Nov 21 06:53:39.985: auth absent

*Nov 21 06:53:39.985: VOICE_REG_POOL: key(B852E755003E51EB) added to nonce table
*Nov 21 06:53:39.985: VOICE_REG_POOL: Contact doesn't match any pools
*Nov 21 06:53:39.985: VOICE_REG_POOL: No entry for (4044) found in contact table

   

1 ACCEPTED SOLUTION

Accepted Solutions

Okay there are a few things to consider here to resolve these issues.

1. The bind statements

2. Source Address of the call manager

Solutions:

a) So i had to put the bind statements on the external interfaces the ones facing the ITSP, 

voice service voip
sip
bind control source-interface GigabitEthernet0/2
bind media source-interface GigabitEthernet0/2

b) Source address i had to marry it with the IP address that was on GigabitEthernet0/2 (10.x.x.x) 

voice register global
mode cme
source-address 10.x.x.x port 5060 ------------------- IP address on Gig0/2

c) I also put the Bind on the outgoing dail peer

dial-peer voice 4001 voip
description SIP TRUNK TO ITSP
translation-profile outgoing OUTSIDECALLING
destination-pattern 0T
session protocol sipv2
session target ipv4:10.x.x.x
voice-class codec 1
voice-class sip options-keepalive up-interval 20 down-interval 40 retry 6
voice-class sip bind control source-interface GigabitEthernet0/2.549
voice-class sip bind media source-interface GigabitEthernet0/2.549
dtmf-relay rtp-nte
no vad

With the above configured

1. I had the phones register to the CUBE..

2. I was able to call external and internal

Remember this is a solution where i split two sites registering to one CUBE and are on two subnets....

 

Note that you may run into authentication errors when you do this please use the following solution

use  a user name and password my extension here were 40XX below is how i resolved the authentication issue

voice register dn 45
number 4045

voice register pool 45
username 4045 password 4045
The above resolves the error that you get in the debugs below..

Nov 21 06:53:39.957: VOICE_REG_POOL: No entry for (4045) found in contact table
*Nov 21 06:53:39.985: auth absent

 

 

View solution in original post

9 REPLIES 9
TONY SMITH
VIP Rising star

You seem to have just one phone defined there, extension 4000 mac address 2C73.A003.A1AF.   Is that phone registered correctly?   I can't see any configuration to allow any other phones to register.  

I am not sure why you have raised multiple threads with slightly different questions about the same issue, that might cause some confusion.  For example in one thread you say that the phones aren't getting their TFTP server settings, but in another thread you have debugs showing them trying to register.

Phone 4000 is working and is at head office. I cut out the config showing other phones the MAC address is there.

The configs are too long.... I have the basics right...

 

voice register dn 44
number 4044
name XXXXX
label XXXXXX

*******************************

!
voice register pool 44
busy-trigger-per-button 2
id mac 4CBC.489F.5E61
type 7821
number 1 dn 44
dtmf-relay sip-notify
codec g711alaw
no vad
!

Will put the full configs then.... 

 

Hi Tony

I have MPLS configured i don't seem to see tftp traffic going back to my phone's .I did a packet capture. I can see one way tftp requests leaving my remote site to HQ. I don't see the return traffic anything am missing here? Check my debugs any pointers?

It doesn't seem likely that the MPLS is blocking TFTP if other IP is OK over the link.  Can you capture at the CME end, see whether it's receiving TFTP requests and if so whether it's sending replies?

Looking at your earlier debugs there are a lot of entries that appear to show it trying for files that don't exist.  Are you sure you have all the correct files on the CME and correctly published on TFTP?

Ok from site B I can see tftp traffic leaving going to head office I don't see it hitting the router at head office. I don't see traffic returning.

In the show voice register tftp binding all these phones and Mac addresses are they cause they once registered.am sure the config files are here to

What I must say is after they loaded first time around I shipped them but continued tweaking the system here. So the final config file could differ a bit? I did a load of no create profile and create profile.

I have been to a site where all phones in remote sites register to the call manager at head office how is this failing?

"Ok from site B I can see tftp traffic leaving going to head office I don't see it hitting the router at head office. I don't see traffic returning."

If that's literally true, TFTP packets are being sent but not being received at HQ then you have a network problem.  Are you completely sure they're not arriving, as opposed to arriving but not having the desired effect?  How are you capturing at HQ?  I'm just labouring this point because your earlier debugs showed what appeared to be traffic relating to this phone being processed at the CME.  For example these excerpts seem to show it being successful at TFTP but then not finding it's registration entry.

*Nov 21 07:29:15.285: TFTP: Looking for SEP4CBC489F5E61.cnf.xml
*Nov 21 07:29:15.289: TFTP: Opened flash:/its/SEP4CBC489F5E61.cnf.xml, fd 10, si
ze 4805 for process 164
*Nov 21 07:29:15.441: TFTP: Finished flash:/its/SEP4CBC489F5E61.cnf.xml, time 00
:00:00 for process 164

Assuming those refer to the phone at issue, then clearly at that time it's TFTP and SIP traffic was reaching the CME.

Tony

I posted using a small screen ( Phone) I meant i see TFTP traffic hitting my router but i don't see the return traffic. As you stated " Traffic arrives but it does not have desired effect" 

If i can have the router push the files to the phones or rather if the phones can pull the files then i would be home and dry. I have seen many a time in CUCM deployments you see the server on different subnet and some phones even over an MPLS but still register to the server farm at the head office.

 

You're quite correct there are countless CUCM installs with phones at remote locations over every sort of link.  Strictly that's not a Cisco supported design for CME, but there's no technical reason why it won't work.  We have customers who use this model.

Back to your issue,  I am still unclear exactly what works and what doesn't, and I think things are being changed which doesn't help.  Earlier you posted debugs showing TFTP being successful, but the phone wasn't registering at as some of the SIP configuration was missing.

One course of action would be to review what has changed since the 21st to stop TFTP working.

However I think I favour a higher level approach since there are so many unknowns, and we are not being shown the whole picture at any given time. 

(1) Take this problem phone ext 4044 back to HQ and confirm that it truly does register there.  If it doesn't then get that resolved before moving on.

(2) Back to the new site, confirm that DHCP is all in place and correct, confirm that the phone is receiving all the correct DHCP info including TFTP server

(3) If these two preceding steps don't find an resolve the issue then it's back to fundamental IP networking

Okay there are a few things to consider here to resolve these issues.

1. The bind statements

2. Source Address of the call manager

Solutions:

a) So i had to put the bind statements on the external interfaces the ones facing the ITSP, 

voice service voip
sip
bind control source-interface GigabitEthernet0/2
bind media source-interface GigabitEthernet0/2

b) Source address i had to marry it with the IP address that was on GigabitEthernet0/2 (10.x.x.x) 

voice register global
mode cme
source-address 10.x.x.x port 5060 ------------------- IP address on Gig0/2

c) I also put the Bind on the outgoing dail peer

dial-peer voice 4001 voip
description SIP TRUNK TO ITSP
translation-profile outgoing OUTSIDECALLING
destination-pattern 0T
session protocol sipv2
session target ipv4:10.x.x.x
voice-class codec 1
voice-class sip options-keepalive up-interval 20 down-interval 40 retry 6
voice-class sip bind control source-interface GigabitEthernet0/2.549
voice-class sip bind media source-interface GigabitEthernet0/2.549
dtmf-relay rtp-nte
no vad

With the above configured

1. I had the phones register to the CUBE..

2. I was able to call external and internal

Remember this is a solution where i split two sites registering to one CUBE and are on two subnets....

 

Note that you may run into authentication errors when you do this please use the following solution

use  a user name and password my extension here were 40XX below is how i resolved the authentication issue

voice register dn 45
number 4045

voice register pool 45
username 4045 password 4045
The above resolves the error that you get in the debugs below..

Nov 21 06:53:39.957: VOICE_REG_POOL: No entry for (4045) found in contact table
*Nov 21 06:53:39.985: auth absent

 

 

View solution in original post

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