some numbers are not routed

[engreda22]

In can't call any local number starts with 46

below the call information and attached the logging

Call Control Block (CCB) : 0x0x30E10E8
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 4552
Called Number            : 94646677
Source IP Address (Sig  ): 172.20.201.18
Destn SIP Req Addr:Port  : 10.0.32.220:5060
Destn SIP Resp Addr:Port : 10.0.32.220:54271
Destination Name         : 10.0.32.220

000643: Jul 27 16:37:55.320: //155969/AE4141000000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 172.20.201.18
Source IP Port    (Media): 30136
Destn  IP Address (Media): 10.0.32.220
Destn  IP Port    (Media): 26106
Orig Destn IP Address:Port (Media): [ - ]:0

000644: Jul 27 16:37:55.320: //155969/AE4141000000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 57
Disconnect Cause (SIP)   : 403

any advice please

[Chris Deren]

This error is what you need to chase:

SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.68.12.194:5060;branch=z9hG4bK1B7BA04
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: 852F5C25-339B11E5-9868D278-86C302A3@10.68.12.194
From: "reda tvtc"<sip:2504552@10.68.12.194>;tag=B2522D20-598
To: <sip:4646677@10.200.7.157>;tag=sbc0802o4fokucf
CSeq: 101 INVITE
Reason: Q.850;cause=57;text="bearer capability not authorized"

What country is this in?

What codec is expected to be used?

[engreda22]

the service located in KSA

all other calls are working fine even local and national

attached the config file

[Wilson Samuel]

Thanks for information. It looks like somehow the Service Provider does not like the format of the number you sent or the Provider has mistakenly barred your Local Call Services (it is doubtful but possible).

Could you please try sending the complete E.164 Format for the Local calls and see if it works, e.g. Presuming you are in Riyadh, please try using the following format +966-11-4646677

More over, just wondering I would want to see, if the 9 is really be stripped before the digits are being sent to the Carrier

HTH

[engreda22]

this is the update from the provider ..

PCMA as first priority

PCMA as second priority

INVITE sip:4646677@10.200.0.7:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.194.62:5060;branch=z9hG4bK1E5CF05T00459

Record-Route: <sip:10.141.194.62:5060;transport=udp;lr>

Call-ID: isbcE9DEA50F-353111E5-84F3D278-86C302A3@10.68.12.194

From: "reda tvtc"<sip:2504552@10.200.0.7>;tag=sbc0804BCB97EC8-D8A

To: <sip:4646677@10.200.0.7>

CSeq: 101 INVITE

Date: Wed, 29 Jul 2015 14:06:59 GMT

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6a

Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY,INFO,REGISTER

Contact: <sip:2504552@10.141.194.62:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires: 1800

Remote-Party-ID: "reda tvtc" <sip:2504552@10.68.12.194>;party=calling;screen=yes;privacy=off

Cisco-Guid: 0311822592-0000065536-0000002170-3693084682

Content-Length: 244

Content-Type: application/sdp

Content-Disposition: session;handling=required

v=0

o=- 5881 1407 IN IP4 10.201.20.45

s=SBC call

c=IN IP4 10.201.20.45

t=0 0

m=audio 38004 RTP/AVP 0 101 19

c=IN IP4 10.201.20.45

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:19 CN/8000

a=ptime:20

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 10.141.194.62:5060;branch=z9hG4bK1E5CF05T00459

Call-ID: isbcE9DEA50F-353111E5-84F3D278-86C302A3@10.68.12.194

From: "reda tvtc"<sip:2504552@10.200.0.7>;tag=sbc0804BCB97EC8-D8A

To: <sip:4646677@10.200.0.7>;tag=tskakp4u

CSeq: 101 INVITE

Reason: Q.850;cause=57;text="bearer capability not authorized"

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

Content-Length: 0

any advise  please

[Tagir Temirgaliyev]

this is the update from the provider ..

PCMA as first priority

PCMA as second priority

=====================

but u send invite with

m=audio 38004 RTP/AVP 0 101 19

c=IN IP4 10.201.20.45

a=rtpmap:0 PCMU/8000

so do

conf t

dial-peer voice 140 voip

 codec  g711alaw

[Mohammed al Baqari]

On your dialpeers facing the ITSP, can you make sure the you configure G711alaw statically and take off any voice-class.

[engreda22]

thanks  Mohamed for your reply

if I have an issue , other calls should stopped as well .am I right ?

local and national calls are working fine

[Mohammed al Baqari]

Not necessary. It depends how your ITSP is matching incoming calls. If itsp is matching all calls on single dialpeer then your thinking is valid. But if they have multiple dialpeers which might be the case here then you can have different behaviour for different calls.

In anyway your itsp is advising to make g711alaw as primary codec so we need to follow them

[engreda22]

Dear Mohamed ,

based on the attached file

the call matches dial-peer 140 and 200

both dial peer 200 and 140 are using voice codec 1

voice codec 1 is using the required codec format

any advice how to clear this ?

thanks

[Tagir Temirgaliyev · Accepted solution]

The CUCM is sending an Early Offer INVITE now. That means you have enabled MTP on the SIP trunk. The codec being sent in the INVITE is G711ulaw. In order to change this, go to the SIP trunk and look for "MTP Preferred Originating Codec" and change it from default G711ulaw to G711alaw. 

[engreda22]

Thanks a lot Tagir .. solved  :)

now my question is  why only this pattern was getting dropped ?

Thanks again for all for time and appreciate your kind support

[Tagir Temirgaliyev]

I think call goes from cube to itsp border switch a then to the next switch 46..... witch only supports 711a

[Mohammed al Baqari]

Hi,

Looking at the config, all looks good. The message indicate there is a codec problem. Try to contact your ITSP and see what are they matching for calls starting with 46. Sounds to be different codec that rest of number within same dialpeer.

[Tagir Temirgaliyev]

voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8
 codec preference 4 g729br8
 codec preference 5 g728

dial-peer voice 140 voip
 description *** Outgoing Local calls,HLatta ***
 translation-profile outgoing OUT-SIP
 destination-pattern 9[1245678]...... 
 session protocol sipv2
 session target ipv4:10.200.7.157
 voice-class codec 1  

 dtmf-relay rtp-nte

but in sip invite only one codec  g711ulaw

000606: Jul 27 16:37:55.208: //155970/000000000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:4646677@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.68.12.194:5060;branch=z9hG4bK1B7BA04
Remote-Party-ID: "reda tvtc" <sip:2504552@10.68.12.194>;party=calling;screen=yes;privacy=off
From: "reda tvtc" <sip:2504552@10.68.12.194>;tag=B2522D20-598
To: <sip:4646677@10.200.7.157>

.....

v=0
          
.....
m=audio 30138 RTP/AVP 0 101 19

only one codec  g711ulaw in invite from cucm to cube

make another good call and share log

and ask provider what codecs supported