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not able to dial 800 toll free numbers

Vijay Anand
Level 1
Level 1

Hi All, 

I came across an issue where users are not able to dial 800 toll free numbers. 

They are using CME and sip trunk to provider. when they are dialling 0800XXXXXXX getting busy signal not connecting. 

for test, we tried calling 8001248888 from extension 5370. 

logs attached.

in the config i saw that toll free dial peer is missing so i added below config. 

voice translation-rule 800

rule 1 /^0/ //

 

voice translation-profile 800

 translate called 800

 

 

dial-peer voice 800 voip

 corlist outgoing INTERNATIONAL

 description **Outgoing Call to  800**

 translation-profile outgoing 800

 destination-pattern .

 session protocol sipv2

 session target sip-server

 session transport udp

 voice-class sip dtmf-relay force rtp-nte

 dtmf-relay rtp-nte

 codec g711ulaw

 no vad

still calls not working, 

regards

Vijay

4 Replies 4

Vivek Batra
VIP Alumni
VIP Alumni

Hi,

You are sending only 0 to SIP provider. I didn't check full config to get the reason. Configure more specific destination pattern like 0800.......$ or 800.......$ depends on what you are going to dial and see if it works for you.

- Vivek

Thanks Vivek,

I follow as per your suggestion, i guess service provider is not allowing the 800 calls. as per log below.

I checked the configuration for CME, there was missing dial peer for dialling toll free 800 numbers, after adding the dial peer, I made test calls by dialling 8001248888 toll free number but we are getting busy signal. From the logs I see that SIP service provider is sending no routes found for these numbers. Can we ask the service provide to check on this issue and adjust their end configuration for 800 numbers.

 

Below is the dial peer i currently added in CME

 

dial-peer voice 800 voip

corlist outgoing INTERNATIONAL

description **Outgoing Call to  800**

destination-pattern 800.......$

session protocol sipv2

session target sip-server

session transport udp

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

codec g711ulaw

no vad

 

 

now below is the log in which we are getting 480 not found error. I tried sending 08001248888 and 8001248888 to service provider but in both case we are getting 480 error from service provider.

 

 

INVITE sip:8001248888@94.77.211.70:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.6.2:5060;branch=z9hG4bK945B16EA

From: " <sip:5280@192.168.6.2>;tag=205C4D00-24DB

To: <sip:8001248888@94.77.211.70>

Date: Thu, 19 May 2016 13:17:21 GMT

Call-ID: DA56DB54-1CFA11E6-9679F7C4-48FC87FB@192.168.6.2

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 3632085194-0486150630-2524248004-1224509435

User-Agent: Cisco-SIPGateway/IOS-15.3.3.M2

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

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1463663841

Contact: <sip:5280@192.168.6.2:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 293

v=0

o=CiscoSystemsSIP-GW-UserAgent 1865 1986 IN IP4 192.168.6.2

s=SIP Call

c=IN IP4 192.168.6.2

t=0 0

m=audio 18008 RTP/AVP 0 100 101

c=IN IP4 192.168.6.2

a=rtpmap:0 PCMU/8000

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.6.2:5060;branch=z9hG4bK945B16EA

From: <sip:5280@192.168.6.2>;tag=205C4D00-24DB

To: <sip:8001248888@94.77.211.70>

Call-ID: DA56DB54-1CFA11E6-9679F7C4-48FC87FB@192.168.6.2

CSeq: 101 INVITE

Timestamp: 1463663841

 

Received:

SIP/2.0 480 No Routes Found   Service Provider is sending this error

Via: SIP/2.0/UDP 192.168.6.2:5060;branch=z9hG4bK945B16EA

From: <sip:5280@192.168.6.2>;tag=205C4D00-24DB

To: <sip:8001248888@94.77.211.70>;tag=aprqngfrt-u61ast30000a6

Call-ID: DA56DB54-1CFA11E6-9679F7C4-48FC87FB@192.168.6.2

CSeq: 101 INVITE

Timestamp: 1463663841

 

ACK sip:8001248888@94.77.211.70:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.6.2:5060;branch=z9hG4bK945B16EA

From: <sip:5280@192.168.6.2>;tag=205C4D00-24DB

To: <sip:8001248888@94.77.211.70>;tag=aprqngfrt-u61ast30000a6

Date: Thu, 19 May 2016 13:17:21 GMT

Call-ID: DA56DB54-1CFA11E6-9679F7C4-48FC87FB@192.168.6.2

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Hi Vijay,

Now we can seen number being sent correctly in Request-URI but still service provider is returning 480 error response. You can cross verify if the number in Request-URI is correct, else should report it to service provider.

- Vivek

Hi Vijay,

In initial INVITE, you are sending only 4 digits in calling party number. Does your provider allow it? 

Best Regards,

Sudheer Shenoy