Voice gate way configuration

[k.elnokrashy1]

Dear s
i have a problem with voice gateway 
I can not make outgoing or incoming call 

for the incoming i have the following error 

*Apr 22 12:15:38.423: //11433/3AEFBAADAD74/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK26F60
From: <sip:559853245@10.10.0.6>;tag=AB738CC-2115
To: <sip:1100@10.10.90.70>;tag=793691639
Date: Sat, 22 Apr 2017 11:52:01 GMT
Call-ID: 3AF056D5-268C11E7-AD7AA190-4B0C4798@10.10.0.6
CSeq: 101 INVITE
Allow-Events: presence
Warning: 399 PTA-CUCM-B "Unable to find a device handler for the request received on port 64512 from 10.10.0.6"
Content-Length: 0

*Apr 22 12:15:38.427: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:1100@10.10.90.70:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK26F60
From: <sip:559853245@10.10.0.6>;tag=AB738CC-2115
To: <sip:1100@10.10.90.70>;tag=793691639
Date: Sat, 22 Apr 2017 12:15:38 GMT
Call-ID: 3AF056D5-268C11E7-AD7AA190-4B0C4798@10.10.0.6
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

*Apr 22 12:15:38.427: //11432/3AEFBAADAD74/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKcphkaoc7ekdkhppsoacphkc44T18808
From: <sip:559853245@10.68.8.134;user=phone>;tag=sbc0802s7bsbcph-CC-39
To: <sip:0118261100@10.68.8.134;user=phone>;tag=AB738D4-E52
Date: Sat, 22 Apr 2017 12:15:38 GMT
Call-ID: isbcaetubu2fbuhkepkf7aa2ccdeshkfkbp2@SoftX3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=63
Content-Length: 0

*Apr 22 12:15:38.439: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0118261100@10.68.8.134;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKcphkaoc7ekdkhppsoacphkc44T18808
Call-ID: isbcaetubu2fbuhkepkf7aa2ccdeshkfkbp2@SoftX3000
From: <sip:559853245@10.68.8.134;user=phone>;tag=sbc0802s7bsbcph-CC-39
To: <sip:0118261100@10.68.8.134;user=phone>;tag=AB738D4-E52
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

I attached the configuration file 

i appreciate your help 

[singhkulbir29881]

This occurs when the source IP address that is sending the SIP INVITE does not exist as a SIP trunk in CUCM. You may have never created it, or the IP address may not be the one that is in use. Check bind commands for gateways, SIP trunk settings in CUCM etc. 

As the SIP request to 10.10.90.70 having sending by source IP 10.10.0.6. Verify that same is set in SIP trunk in CUCM.

[k.elnokrashy1]

thanks for your help 
it was one issue 
i was point with sip trunck to loopback address I  modefied it to the port address 
but I still have problems 

it seam that the sip between the cucm and Voicce gateway have some thing wrong

[Alok Jaiswal · Accepted solution]

I don't think the 503 service unavailable is related to call issue.

Look at the message below.

Received:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK26F60
From: <sip:559853245@10.10.0.6>;tag=AB738CC-2115
To: <sip:1100@10.10.90.70>;tag=793691639
Date: Sat, 22 Apr 2017 11:52:01 GMT
Call-ID: 3AF056D5-268C11E7-AD7AA190-4B0C4798@10.10.0.6
CSeq: 101 INVITE
Allow-Events: presence
Warning: 399 PTA-CUCM-B "Unable to find a device handler for the request received on port 64512 from 10.10.0.6"
Content-Length: 0

It doesn't knows how to handle this presence event request coming in. 

Can you run debug ccsip messages and debug voice ccapi inout and post it here ??

Regards,

Alok

[k.elnokrashy1]

i have changed the conf
and this is the new debug 

for the outgoing call the debug ccapi is show nothing at all 

[singhkulbir29881]

Hi k.elnokrashy1,

Indebug it says Codec negotiated is g729r8 but is voice gateway config you set it to g711ulaw or g711alaw. Have a look at both.

In debug output,

Negotiated Codec : g729r8
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.10.0.6
Source IP Port (Media): 0
Destn IP Address (Media): 10.10.90.70
Destn IP Port (Media): 25870
Orig Destn IP Address:Port (Media): [ - ]:0

In cofig,

voice class codec 3
codec preference 1 g711ulaw
codec preference 2 g711alaw

Can you verify the CUCM codec setting. 

[k.elnokrashy1]

I checked the CM 
the codec is g711ulaw 

I notice that it fail with codec 47 which mean law resource memory or TCP port 
I think it is the TCP port but i could no know why 

[singhkulbir29881 · Accepted solution]

If you look at the whole debug output there are few things to be noticed. Have a look

PTA-VG2#debug ccsip ca
PTA-VG2#debug ccsip calls
SIP Call statistics tracing is enabled
PTA-VG2#
*Apr 23 14:29:00.829: //17778/E17955800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x27F9348
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 1100
Called Number : 90559853245
Source IP Address (Sig ): 10.10.0.6
Destn SIP Req Addr:Port : 10.10.90.70:0
Destn SIP Resp Addr:Port : 10.10.90.70:46277
Destination Name : 10.10.90.70

*Apr 23 14:29:00.829: //17778/E17955800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729r8
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.10.0.6
Source IP Port (Media): 0
Destn IP Address (Media): 10.10.90.70
Destn IP Port (Media): 25870
Orig Destn IP Address:Port (Media): [ - ]:0

*Apr 23 14:to set 29:00.829: //17778/E17955800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 488

Disconnect cause 65 means media negotiations failed. And TCP SOCKET is in use. 

What is IOS version running on at gateway and can you try to set g729r8 at gateway to test what will happen?

[k.elnokrashy1]

thanks alot 

now i can make incoming call

for the outgoing call I can reach mobile and international call only 

for the local and national call I still have problem 

i submit debug of succeeded mobile  call and non succeeded national call 

 I appreciate  if you could help me in that also

[Ayodeji Okanlawon]

please send a "debug ccsip messages"

include calling and called number and attach using a text editor not w=ms word

[k.elnokrashy1]

this the error message from ccsip message 

Apr 24 17:19:10.233: //5234/CDF1EB800000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 10.68.8.134:5060;branch=z9hG4bK7BEA2D

Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>

Call-ID: F696F84A-284811E7-A7578FC9-EDE507BE@10.68.8.134

From: "Saeed Alqahtani"<sip:8261101@10.200.7.157>;tag=21FA878-1C1A

To: <sip:90148455215@10.200.7.157>;tag=sbc0802dba44dda

CSeq: 101 INVITE

Reason: Q.850;cause=1;text="Unallocated number"

Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"

Content-Length: 0


the calling number is 8261101
and the called number  148455215

[Alok Jaiswal]

Hi,

Is the number to be sent with leading "90" ??

To: <sip:90148455215@10.200.7.157>;tag=sbc0802dba44dda

From the 404 not found by softx, it is complaining about not able to find the number. May be its your translation profile if the prefix number "90" needs to be discarded.

Also the ccsip debug doesn't contains this call, i don;t see this number. 

Regards,

Alok

[k.elnokrashy1]

thanks for your help 
it was wrong translation in the outgoing 
it should be outgoing and  I but it translated incoming 

[Ayodeji Okanlawon]

There are two legs to a call. The inbound leg/dial-peer and outbound leg/dial-peer.

Your inbound dial-peer is 200 and this dial-peer is using default G729r8 codec.

dial-peer voice 200 voip

description *********** dial-peer incoming( CM & WAN ) ***********

translation-profile incoming pta-incoming

session protocol sipv2

incoming called-number .%

dtmf-relay rtp-nte

But your outbound dial-peer is configured to use only G711 codec

dial-peer voice 100 voip

description *****outgoing dial-peer to the CM *******

destination-pattern ....

session protocol sipv2

session target ipv4:10.10.90.70

voice-class codec 3

voice-class sip bind control source-interface GigabitEthernet0/0.111

voice-class sip bind media source-interface GigabitEthernet0/0.111

dtmf-relay rtp-nte

Please apply voice-class codec 3 to your inbound dial-peer too and ensure the region setting between cucm and sip trunk is set to 64Kbps

dial-peer voice 200 voip

voice-class codec 3