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Issue with SIP trunk from CME to CUCM 11.5

carl_townshend
Spotlight
Spotlight

‎10-08-2018 05:12 AM - edited ‎03-17-2019 01:33 PM

Hi All

I have a SIP trunk between my CUCM v11.5 and CME, We can call the CME from the CUCM but not the other way around.

 

On the logs, I cannot see the UK subscriber as the destination

 

Setup is

 

dial-peer voice 8000 voip
 description *** CCM SUBSCRIBER NODE ***
 preference 2
 destination-pattern 8801....
 session protocol sipv2
 session target ipv4:172.30.1.1
 session transport tcp
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 codec g711ulaw
 fax-relay ecm disable
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 ip qos dscp cs3 signaling
 no vad

 

We get the below errors, any ideas?

 

The Call Setup Information is:
Call Control Block (CCB) : 0x0x7F7B219E3D30
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 8006
Called Number            : 88014853
Source IP Address (Sig  ): 172.29.x.x
Destn SIP Req Addr:Port  : :0
Destn SIP Resp Addr:Port : :0
Destination Name         :
109389: *Oct  8 14:21:07.646: //365784/15FD52D898EF/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 172.29.x.x
Source IP Port    (Media): 9280
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
109390: *Oct  8 14:21:07.646: //365784/15FD52D898EF/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 47
Disconnect Cause (SIP)   : 200
109391: *Oct  8 14:21:07.646: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_IWF
109392: *Oct  8 14:21:07.647: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
109393: *Oct  8 14:21:07.647: //365785/15FD52D898EF/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x7F7B219E3D30
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 8006
Called Number            : 88014853
Source IP Address (Sig  ): 172.29.x.x
Destn SIP Req Addr:Port  : :0
Destn SIP Resp Addr:Port : :0
Destination Name         :
109394: *Oct  8 14:21:07.647: //365785/15FD52D898EF/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 172.29.x.x
Source IP Port    (Media): 9282
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

 

 

 

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12 Replies 12

bichacko
Cisco Employee
Cisco Employee

‎10-08-2018 05:31 AM


Hi,

Ok, a few things here:

>> I see the dial-peer has preference 2. So which is set for preference 1?
>> Can you paste the output of:

# sh dial-peer voice summ
# sh run | sec dial-peer

>> Apart from that, it looks to be a CODEC failure and I believe it's because it's using some other dial-peer:

109390: *Oct 8 14:21:07.646: //365784/15FD52D898EF/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 47
Disconnect Cause (SIP) : 200
109391: *Oct 8 14:21:07.646: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_IWF
109392: *Oct 8 14:21:07.647: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
0 Helpful

carl_townshend
Spotlight
Spotlight
In response to bichacko

‎10-08-2018 05:54 AM

Hi, The other dial peer is exactly the same but points to the other subscriber.

 

There is also another dial peer which is the external SIP provider.

 

TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT    KEEPALIVE    VRF
201    voip  up   up             ^849[59].......$   0  syst sip-server                               NA
202    voip  up   up             ^89.........$      0  syst sip-server                               NA
203    voip  up   up             ^8[02-8]........-  0  syst sip-server                               NA
                                 .$
204    voip  up   up             810T               0  syst sip-server                               NA
205    voip  up   up             ^...$              0  syst sip-server                               NA
206    voip  up   up             ^0.$               0  syst sip-server                               NA
101    voip  up   up             8001               0  syst sip-server                               NA
102    voip  up   up             8002               0  syst sip-server                               NA
103    voip  up   up             8003               0  syst sip-server                               NA
104    voip  up   up             8004               0  syst sip-server                               NA
105    voip  up   up             8005               0  syst sip-server                               NA
106    voip  up   up             8006               0  syst sip-server                               NA
107    voip  up   up             8007               0  syst sip-server                               NA
108    voip  up   up             8008               0  syst sip-server                               NA
20001  pots  up   up             A0001$             0                           50/0/1               NA
20002  pots  up   up             A0001$             0                           50/0/2               NA
20003  pots  up   up             A0003$             0                           50/0/3               NA
8000   voip  up   up             8801....           2  syst ipv4:172.30.208.17                       NA
8001   voip  up   up             8801....           3  syst ipv4:172.30.208.2                        NA
40005  voip  up   up             8005$              0  syst ipv4:172.29.74.17:50                     NA
109    voip  up   up             8009               0  syst sip-server                               NA
40003  voip  up   up             8008$              0  syst ipv4:172.29.74.11:50                     NA
110    voip  up   up             8000               0  syst sip-server                               NA
40004  voip  up   up             8006$              0  syst ipv4:172.29.74.18:50                     NA
40006  voip  up   up             8004$              0  syst ipv4:172.29.74.16:50                     NA
40002  voip  up   up             8007$              0  syst ipv4:172.29.74.14:50                     NA
40008  voip  up   up             8000$              0  syst ipv4:172.29.74.13:50                     NA
40001  voip  up   up             8002$              0  syst ipv4:172.29.74.210:5                     NA
8002   voip  up   up             8801....           1  syst ipv4:172.30.208.1                        NA

 

 

show run | section dial-peer
dial-peer voice 201 voip
 description Local Calls
 translation-profile outgoing 2
 destination-pattern ^849[59].......$
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 no vad
dial-peer voice 202 voip
 description Cellular Calls
 translation-profile outgoing 2
 destination-pattern ^89.........$
 session protocol sipv2
 session target sip-server
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
dial-peer voice 203 voip
 description Long Distance Calls
 translation-profile outgoing 2
 destination-pattern ^8[02-8].........$
 session protocol sipv2
 session target sip-server
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
dial-peer voice 204 voip
 description International Calls
 translation-profile outgoing 2
 destination-pattern 810T
 session protocol sipv2
 session target sip-server
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
dial-peer voice 205 voip
 description Emergency Calls
 translation-profile outgoing 2
 destination-pattern ^...$
 session protocol sipv2
 session target sip-server
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
dial-peer voice 206 voip
 description Emergency Calls
 translation-profile outgoing 2
 destination-pattern ^0.$
 session protocol sipv2
 session target sip-server
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
dial-peer voice 101 voip
 description -= Incoming to 8001 =-
 translation-profile incoming 1
 destination-pattern 8001
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138001
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 102 voip
 description -= Incoming to 8002 =-
 translation-profile incoming 1
 destination-pattern 8002
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138002
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 103 voip
 description -= Incoming to 8003 =-
 translation-profile incoming 1
 destination-pattern 8003
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138003
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 104 voip
 description -= Incoming to 8004 =-
 translation-profile incoming 1
 destination-pattern 8004
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138004
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 105 voip
 description -= Incoming to 8005 =-
 translation-profile incoming 1
 destination-pattern 8005
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138005
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 106 voip
 description -= Incoming to 8006 =-
 translation-profile incoming 1
 destination-pattern 8006
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138006
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 107 voip
 description -= Incoming to 8007 =-
 translation-profile incoming 1
 destination-pattern 8007
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138007
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 108 voip
 description -= Incoming to 8008 =-
 translation-profile incoming 1
 destination-pattern 8008
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138008
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 8000 voip
 description *** UK CCM ROC SUBSCRIBER NODE ***
 preference 2
 destination-pattern 8801....
 session protocol sipv2
 session target ipv4:172.30.208.17
 session transport tcp
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 codec g711ulaw
 fax-relay ecm disable
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 8001 voip
 description *** UK CCM UTT SUBSCRIBER NODE ***
 preference 3
 destination-pattern 8801....
 session protocol sipv2
 session target ipv4:172.30.208.2
 session transport tcp
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 codec g711ulaw
 fax-relay ecm disable
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 109 voip
 description -= Incoming to 8009 =-
 translation-profile incoming 1
 destination-pattern 8009
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138009
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 110 voip
 description -= Incoming to 8000 =-
 translation-profile incoming 1
 destination-pattern 8000
 session protocol sipv2
 session target sip-server
 incoming called-number 9637138000
 dtmf-relay rtp-nte sip-notify
 codec g711alaw
 no vad
dial-peer voice 8002 voip
 description *** UK CCM UTT PUBLISHER NODE ***
 preference 1
 destination-pattern 8801....
 session protocol sipv2
 session target ipv4:172.30.208.1
 session transport tcp
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 codec g711ulaw
 fax-relay ecm disable
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 ip qos dscp cs3 signaling
 no vad

0 Helpful

bichacko
Cisco Employee
Cisco Employee
In response to carl_townshend

‎10-08-2018 06:18 AM

Hi,

It looks like one of those should be ideally selected.

>> Can you confirm the region settings on CUCM?
>> Can you gather debugs from the gateway:

# deb ccsip messages
# deb voice ccapi inout


Thanks,
Bijo
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carl_townshend
Spotlight
Spotlight
In response to bichacko

‎10-08-2018 06:22 AM

The Region on the CUCM is system default using G722/G711

 

Is it possible I can dial from the Gateway to the CUCM, I don't think any users are on site to call at the min

 

cheers

 

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carl_townshend
Spotlight
Spotlight
In response to bichacko

‎10-08-2018 07:15 AM

Here you are

 

It does say transcoder not configured, not sure why it would say this as its G711 both ends, could this be the issue?

 

how would we fix ?

 

 

SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.29.74.18:5060;branch=z9hG4bK6e547037
From: "8006" <sip:8006@172.29.74.252>;tag=00727827955801b22ae145cd-4f848b0a
To: <sip:88014853@172.29.74.252>;tag=F7CC926D-1BFD
Date: Mon, 08 Oct 2018 14:09:45 GMT
Call-ID: 00727827-95580010-086a34de-414d2163@172.29.74.18
CSeq: 101 INVITE
Allow-Events: telephone-event
Warning: 399 172.29.74.252 "Transcoder Not Configured"
Server: Cisco-SIPGateway/IOS-16.7.1
Reason: Q.850;cause=47
Session-ID: 00000000000000000000000000000000;remote=15206b5e00105000a000007278279558
Content-Length: 0

110296: *Oct  8 17:09:45.151: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:88014853@172.29.74.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.29.74.18:5060;branch=z9hG4bK6e547037
From: "8006" <sip:8006@172.29.74.252>;tag=00727827955801b22ae145cd-4f848b0a
To: <sip:88014853@172.29.74.252>;tag=F7CC926D-1BFD
Call-ID: 00727827-95580010-086a34de-414d2163@172.29.74.18
Session-ID: 15206b5e00105000a000007278279558;remote=00000000000000000000000000000000
Max-Forwards: 70
Date: Mon, 08 Oct 2018 14:09:38 GMT
CSeq: 101 ACK
Content-Length: 0

110297: *Oct  8 17:09:45.152: //366783/A47909EA997D/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x7F7B197423A0, Tag=0x0, Call Id=366783,
   Call Entry(Disconnect Cause=47, Voice Class Cause Code=0, Retry Count=0)
110298: *Oct  8 17:09:45.152: //366783/A47909EA997D/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
110299: *Oct  8 17:09:45.152: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
110300: *Oct  8 17:09:45.152: :cc_free_feature_vsa freeing 7F7B1DF68940
110301: *Oct  8 17:09:45.152: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
110302: *Oct  8 17:09:45.152:  vsacount in free is 0
110303: *Oct  8 17:10:28.902: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.74.252:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.72.129:5060;branch=z9hG4bK407258912-104363779
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
From: <sip:172.26.72.129>;tag=0_407258912-104363780
To: <sip:172.29.74.252>
Call-ID: 407258912-104363778
CSeq: 1 OPTIONS
Contact: <sip:172.26.72.129:5060;transport=udp>
User-Agent: Mitel-3300-ICP 14.0.1.29
Content-Length: 0
 
0 Helpful

bichacko
Cisco Employee
Cisco Employee
In response to carl_townshend

‎10-08-2018 08:28 AM


Is it only with SIP phone or both SIP and SCCP?
What's the codec assigned to the phone on CME?

Can you try configuring "voice-class codec" with g711ulaw in preference 1 and g729r8 in preference 2, then it assign to the
voice register pool of the sip phone (if not already done)?

Bijo
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R0g22
Cisco Employee
Cisco Employee
In response to carl_townshend

‎10-08-2018 09:15 AM

CME and CUBE co-located are NOT supported. I won't encourage the design as well even though with some tweaking and troubleshooting you might be able to get it working. But that does not mean that it should be recommended.

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carl_townshend
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Spotlight
In response to R0g22

‎10-09-2018 04:11 AM

Hi

So are you saying that you cannot have an external SIP trunk directly on the same CME system? are you saying you would need to buy a another voice router just for the CUBE functionality ?

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R0g22
Cisco Employee
Cisco Employee
In response to carl_townshend

‎10-09-2018 08:13 AM

Correct. The same router CANNOT be used for CME and as a CUBE simultaneously. If you need to run CME, you need a different box. You can use the current one for CUBE only.

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carl_townshend
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Spotlight
In response to R0g22

‎10-10-2018 04:48 AM

Hi

I would like to know why this is?, as the whole point of a CME is office in a box style functionality, having another router just for the SIP trunk is a massive overkill in my eyes, we can use acl's / zbfw etc to protect the CME.

We have actually now got it working, it was a codec issue

 

cheers

 

0 Helpful

livi9003
Level 1
Level 1
In response to carl_townshend

‎01-24-2019 08:17 AM

How did you figure out it was a codec issue? 

Thanks,

Kevin

@carl_townshend wrote:

Hi

I would like to know why this is?, as the whole point of a CME is office in a box style functionality, having another router just for the SIP trunk is a massive overkill in my eyes, we can use acl's / zbfw etc to protect the CME.

We have actually now got it working, it was a codec issue

 

cheers

 

 

0 Helpful

carl_townshend
Spotlight
Spotlight
In response to livi9003

‎01-28-2019 06:43 AM

I think we ended up switching to a voice class for the codec rather than nailing it down on the dial peer

 

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