SIP-to-SIP through CUBE between CM 6.1 and a SIP provider

[valdes.argm]

I’m trying to set up a CUBE between our CM 6.1 and a SIP provider for PSTN access. So far I can make a phone outside the PSTN to ring and even pick up the call, but the Cisco phone on the CM side keeps in ring out state and obviously there is no audio.

I set up the transcode on the same 2801 that host CUBE, but I think I still have a signaling problem.

Do I miss something at the router's config? Thanks.

Building configuration...

Current configuration : 4842 bytes
!
! Last configuration change at 15:27:29 AR Tue Jun 5 2012 by servicio
! NVRAM config last updated at 15:33:39 AR Tue Jun 5 2012 by servicio
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service sequence-numbers
!
hostname CUBE01
!
boot-start-marker
boot system flash c2801-ipvoice_ivs-mz.124-24.T6.bin
boot-end-marker
!
logging count
logging message-counter syslog
logging buffered 200000
logging rate-limit 10000
no logging console
enable secret 5 *************************
!
no aaa new-model
clock timezone AR -3
ip source-route
!
ip cef
no ip domain lookup
ip domain name avvid
no ipv6 cef
multilink bundle-name authenticated
!
voice service voip
address-hiding
allow-connections sip to sip
fax protocol cisco
sip
  midcall-signaling passthru
!
voice class codec 20
codec preference 1 g729r8
codec preference 2 g729br8
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
voice translation-rule 10
rule 1 /^************\.*/ /\1/
!
voice translation-rule 20
rule 1 /^*********[*,*]$/ /****/
!
voice translation-profile Strip
translate called 10
!
voice-card 0
dsp services dspfarm
!
archive
log config
  hidekeys
!
interface FastEthernet0/0
ip address 10.0.0.9 255.255.255.0
duplex auto
speed auto
no cdp enable
no mop enabled
!
interface FastEthernet0/1
ip address [public ip] 255.255.255.240
duplex auto
speed auto
no cdp enable
!
ip forward-protocol nd
ip route *.*.*.* 0.0.0.0 *.*.*.* name DG
ip route 10.0.0.0 255.255.252.0 10.0.0.1 name LAN
!
ip http server
ip http authentication local
!
control-plane
!
sccp local FastEthernet0/0
sccp ccm 10.0.0.9 identifier 1 version 5.0.1
sccp
!
sccp ccm group 1
bind interface FastEthernet0/0
associate ccm 1 priority 1
associate profile 2 register MTP-CUBE01
associate profile 1 register TX-CUBE01
keepalive retries 5
switchover method immediate
switchback method immediate
switchback interval 15
!
dspfarm profile 1 transcode 
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
!
dspfarm profile 2 mtp 
codec g711ulaw
maximum sessions software 100
associate application SCCP
!
dial-peer voice 10 voip
translation-profile outgoing Strip
destination-pattern 0170.+
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 20
session protocol sipv2
session target ipv4:[SIP server]
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip
answer-address 0170
voice-class codec 1
session protocol sipv2
dtmf-relay rtp-nte digit-drop
!
dial-peer voice 20 voip
voice-class codec 20
session target ipv4:10.0.0.10 /// Call Manager's IP
incoming called-number **********[*,*]
dtmf-relay rtp-nte
!
gatekeeper
shutdown
!
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 4
sdspfarm tag 1 TX-CUBE01
max-ephones 1
max-dn 1
ip source-address 10.0.0.9 port 2000
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Jun 05 2012 10:26:09
!
line con 0
exec-timeout 60 0
timeout login response 60
login local
line aux 0
line vty 0 4
exec-timeout 60 0
timeout login response 60
login local
!
scheduler allocate 20000 1000
no process cpu extended
no process cpu autoprofile hog
ntp update-calendar
end
!

[Ayodeji Okanlawon]

Hi,

This sounds like when the call is about to connect cube needs a xcoder and it cant find one..

Can you please confirm what the region between the phone and CUBE is?

can you send the ff: debugs

1. debug ccsip all

2. debug voip ccapi inout

3. Can you pls put a brief description of your device ips, so i can know the call flow better

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"I am complete in God, God completes me"

[valdes.argm]

The phones and CUBE are on the same LAN (that’s what you mean with region?), and CUBE has the other ethernet port of the 2801 direct connected to internet with a public IP, at least for now during testing.

I have a file with the log of a test call with the debug options enabled, but they are very long, do I paste here anyway?

And what do mean with “brief description of your device ips”? Sorry, maybe because of my poor english or technical skills I don’t understand what you ask for.

[Ayodeji Okanlawon]

No, dont post them here..put them in a text file and attach them here.

By region I mean what region in cucm is the sip trunk and the phone set to use. Ie what codec is defined between these two devices. Its either g711 or g729 and this is usually configured in the region settings

By Description, I mean what ip address is for your cube, your cucm etc

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"I am complete in God, God completes me"

[valdes.argm]

I found the attach option in the reply, sorry. The log is attached now.

The SIP trunk is configured to use g711ulaw codec in CUCM and the SIP provider needs g729r8, this is one of the needs for CUBE. The phones use g711ulaw too. And because of the transcoding needed the 2801 has DSPs on it.

The CUCM and CUBE are on the same subnet, in fact they are at IP .10 (CUMM) and IP .9 (CUBE). And the phones are at the same subnet too.

[Ayodeji Okanlawon]

+++Ok here is what we see+++Calleg to ITSP uses g729

008775: Jun  6 12:04:51.835: //160/A1B9EEDA83C3/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729br8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): [public IP]

Source IP Port    (Media): 0

Destn  IP Address (Media): [SIP provider media IP]

Destn  IP Port    (Media): 19224

Orig Destn IP Address:Port (Media): [ - ]:0

+++Call from cucm uses G711+++++ (on this calleg no codec was negotiated)

008733: Jun  6 12:04:51.815: //159/A1B9EEDA83C3/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): 10.0.0.9

Source IP Port    (Media): 17052

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0

+++Here we see CUBE saying the call needs a xcoder and it actually sent its stream to a xcoder so cube found a xcoder++

need transcoding for codec mismatch

008568: Jun  6 12:04:45.460: //159/A1B9EEDA83C3/SIP/Info/sipSPIDtmfTranscoder: local DTMF 6, peer DTMF 6

008569: Jun  6 12:04:45.460: //159/A1B9EEDA83C3/SIP/Info/sipSPIDtmfTranscoder: local codec 5, peer codec 12

008570: Jun  6 12:04:45.460: //159/xxxxxxxxxxxx/CCAPI/cc_api_set_xcode_stream:

008571: Jun  6 12:04:45.460: cc_api_set_xcode_stream Line: 4556

008572: Jun  6 12:04:45.460: //159/A1B9EEDA83C3/SIP/Info/ccsip_do_xcode_stream_ind:

So what it looks like is that the CUBE can talk to the xcoder (beacuse its local to it) but the phone cant...

Can you go to the phone and check the device pool on the phone. Can you also check the Region assigned to the device pool

Can you do the same for the cube sip trunk and then ensure that the region setting between the region if they are different is g711.

can you also send a sh dspfarm all from cube..

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"I am complete in God, God completes me"

[valdes.argm]

The SIP trunk was on different device pool and different region, I move it to the same device pools of the phones now.

I make another test call and still see the same codec mismatch error that thank to you I now about it now, I attach a new log.

Here is the sh dspfarm all output:

ARGMCUBE01# sh dspfarm all
Dspfarm Profile Configuration

Profile ID = 1, Service = TRANSCODING, Resource ID = 1 
Profile Description : 
Profile Service Mode : Non Secure
Profile Admin State : UP
Profile Operation State : ACTIVE
Application : SCCP   Status : ASSOCIATED
Resource Provider : FLEX_DSPRM   Status : UP
Number of Resource Configured : 4
Number of Resource Available : 4
Codec Configuration
Codec : g711ulaw, Maximum Packetization Period : 30
Codec : g711alaw, Maximum Packetization Period : 30
Codec : g729ar8, Maximum Packetization Period : 60
Codec : g729abr8, Maximum Packetization Period : 60
Codec : g729r8, Maximum Packetization Period : 60
Codec : g729br8, Maximum Packetization Period : 60
Dspfarm Profile Configuration

Profile ID = 2, Service = MTP, Resource ID = 2 
Profile Description : 
Profile Service Mode : Non Secure
Profile Admin State : UP
Profile Operation State : ACTIVE
Application : SCCP   Status : ASSOCIATED
Resource Provider : NONE   Status : NONE
Number of Resource Configured : 100
Number of Resource Available : 100
Hardware Configured Resources : 0
Hardware Available Resources : 0
Software Resources : 100
Codec Configuration
Codec : g711ulaw, Maximum Packetization Period : 30

SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0    1   24.3.6   UP     N/A  FREE  xcode  1      -         -         -       
0    1   24.3.6   UP     N/A  FREE  xcode  1      -         -         -       
0    1   24.3.6   UP     N/A  FREE  xcode  1      -         -         -       
0    1   24.3.6   UP     N/A  FREE  xcode  1      -         -         -       

Total number of DSPFARM DSP channel(s) 4

CUBE01#

[Ayodeji Okanlawon]

Did you reset the sip trunk?

Do you know how to get cucm traces? if so can you send cucm traces..that will really help

when you do a test call can you send the ff: from the cube

sh voip rtp connection

sh dspfarm sessions

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"I am complete in God, God completes me"

[Mohammed al Baqari]

In fact CUBE is able to invoke XCODER. But in your debugs I can't see 200OK response from SIP Provider. This means that remote phone didn't pickup the call. SIP Provider is sending to CUBE 183 messages only then CANCLE message from provider. If the remote phone pickup the phone, ideally, SIP Provider should provide you with 200OK message.

Are you sure that remote phone is answered? If yes, then you need to escelate this to your ITSP since they aren't forwarding 200OK message to you.

[Ayodeji Okanlawon]

I also observe that the provider is not sending 200 ok. But I observed that he said he could pickup the call. That suggest that he is calling the phone himself and he is picking it up.

So Yes Can you clarify that you are picking up the call..as we do not see a 200 ofter the session progress

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"I am complete in God, God completes me"

[valdes.argm]

You are correct, I’m calling to my cell phone and I pick up the call, but the phone on my desk still ringing out. If I hang up on the cell phone the desk phone still ringing out. If I hang up on the desk phone, then the cell phone yes disconnect the call.

The no 200OK message and still getting the 183 problem could be a bug at the CUCM as mentioned here:

http://comments.gmane.org/gmane.comp.voip.cisco/33626

Because my CUCM is version 6.1.2.1000-13, I enable the option to require MTP for the SIP trunk at the CUCM, and also create a DSP profile for MTP on the CUBE, but this part I’m not sure if is working properly.

I will try with the CUCM traces to see if there is a problem with this leg. It’s possible that the provider keeps at 183 messages because there is no RTP in the other leg between the phone and CUBE?

[valdes.argm]

I forgot to mention that I reset the SIP trunk after changing the device pool, and after the first call test and the same error I reset the CUBE, just in case…

[Mohammed al Baqari]

Its not a bug in your CUCM. Your provider isn't sending 200OK as we can see on CUBE. This is still before reaching CUCM.

This needs to be fixed first then we can go further.

Also, I don't see bind source-interface command in your SIP service configuration which is mandatory. Please add this as well.

[valdes.argm]

Ok, and this is something I need to troubleshoot with the SIP provider right?

[Ayodeji Okanlawon]

Yes speak to them and we can take it from there

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"I am complete in God, God completes me"