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Problems on outbound SIP

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CptIPO

CptIPO

Technical User
May 4, 2012
95
0
0
ZA
Hi guys I am having a strange fault. If I make outbound calls using a specific provider we have in South Africa. I have the following:

I dial a number from my 1408
It shows number on screen until call is about to ring
Immediately on screen of phone it changes to Anonymous
On the SIP trace it show a bit further down trace I have included in red that they send back a message where contact is anonymous
How do I prevent system from changing the number on screen from using the "Contact: Anonymous <sip:196.43.231.50:5072>" to maybe just the number in the To header
Why does it change, the system should surely only show the number that is dialled?
Please can someone help, I have been struggling with this issue for a while and cant get it working
Problem is that on 3 other PBX's (Siemens, Alcatel and Samsung) plus I have testing using a Patton VoIP Router I don't have same issue. I still get message but the PBX or Patton dont change number that shows on phone.


137765772mS SIP Call Tx: 17
INVITE sip:0217888685@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 192.168.110.1:5060;rport;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
To: <sip:0217888685@196.43.231.50>
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Contact: "27217008100" <sip:27217008100@192.168.110.1:5060;transport=udp>
Authorization: Digest username="27217008100",realm="196.43.231.50",nonce="9456caecc06bfe602cc91e5462bc427c35b7",response="51835cfe009ae015858f27de286ad673",uri="sip:0217888685@196.43.231.50"
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 9.0.5.0 build 972
Content-Length: 228

v=0
o=UserA 898253156 3346606824 IN IP4 192.168.110.1
s=Session SDP
c=IN IP4 192.168.110.1
t=0 0
m=audio 49154 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
137765772mS SIP Tx: UDP 192.168.110.1:5060 -> 196.43.231.50:5060
INVITE sip:0217888685@196.43.231.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.110.1:5060;rport;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
To: <sip:0217888685@196.43.231.50>
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Contact: "27217008100" <sip:27217008100@192.168.110.1:5060;transport=udp>
Authorization: Digest username="27217008100",realm="196.43.231.50",nonce="9456caecc06bfe602cc91e5462bc427c35b7",response="51835cfe009ae015858f27de286ad673",uri="sip:0217888685@196.43.231.50"
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 9.0.5.0 build 972
Content-Length: 228

v=0
o=UserA 898253156 3346606824 IN IP4 192.168.110.1
s=Session SDP
c=IN IP4 192.168.110.1
t=0 0
m=audio 49154 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
137765844mS SIP Rx: UDP 196.43.231.50:5060 -> 192.168.110.1:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.110.1:5060;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617;received=105.236.105.179
To: <sip:0217888685@196.43.231.50>
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Content-Length: 0

137765846mS SIP Call Rx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.110.1:5060;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617;received=105.236.105.179
To: <sip:0217888685@196.43.231.50>
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Content-Length: 0

137768977mS SIP Rx: UDP 196.43.231.50:5060 -> 192.168.110.1:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.110.1:5060;received=105.236.105.179;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Record-Route: <sip:196.43.231.50:5060;transport=udp;lr>
Contact: Anonymous <sip:196.43.231.50:5072>
To: <sip:0217888685@196.43.231.50>;tag=zceblram34yircsq.i
From: 27217008100 <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Content-Type: application/sdp
Server: Sippy
Portasip-3264-action: preanswer 1
Content-Length: 219

v=0
o=Sippy 3393845204197527223 1 IN IP4 196.43.231.50
s=-
t=0 0
m=audio 54670 RTP/AVP 18 101
c=IN IP4 196.43.231.50
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
137768980mS SIP Call Rx: 17
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.110.1:5060;received=105.236.105.179;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Record-Route: <sip:196.43.231.50:5060;transport=udp;lr>
[highlight #CC0000]Contact: Anonymous <sip:196.43.231.50:5072>[/highlight]
To: <sip:0217888685@196.43.231.50>;tag=zceblram34yircsq.i
From: 27217008100 <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Content-Type: application/sdp
Server: Sippy
Portasip-3264-action: preanswer 1
Content-Length: 219

v=0
o=Sippy 3393845204197527223 1 IN IP4 196.43.231.50
s=-
t=0 0
m=audio 54670 RTP/AVP 18 101
c=IN IP4 196.43.231.50
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
137768981mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 196.43.231.50:5060 to 196.43.231.50:5060
137768982mS Sip: 17.3764.0 764 SIPTrunk Endpoint(f4971830) SetRfc2833TxPayload: use RFC2833 for dtmf
137768983mS Sip: 17.3764.0 764 SIPTrunk Endpoint(f4971830) SetRemoteRTPAddress to 196.43.231.50:54670
137768989mS Sip: 17.3764.0 764 SIPTrunk Endpoint(f4941df8) received CMFacility
137770131mS SIP Rx: UDP 196.43.231.50:5060 -> 192.168.110.1:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.110.1:5060;received=105.236.105.179;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Record-Route: <sip:196.43.231.50:5060;transport=udp;lr>
Contact: Anonymous <sip:196.43.231.50:5072>
To: <sip:0217888685@196.43.231.50>;tag=zceblram34yircsq.i
From: 27217008100 <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Server: Sippy
Portasip-3264-action: none 1
Content-Length: 0

137770133mS SIP Call Rx: 17
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.110.1:5060;received=105.236.105.179;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Record-Route: <sip:196.43.231.50:5060;transport=udp;lr>
Contact: Anonymous <sip:196.43.231.50:5072>
To: <sip:0217888685@196.43.231.50>;tag=zceblram34yircsq.i
From: 27217008100 <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Server: Sippy
Portasip-3264-action: none 1
Content-Length: 0

137770135mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 196.43.231.50:5060 to 196.43.231.50:5060
137771376mS PRN: CDR - TCPSend maxqueuesize=3000 framecount=3000 operational=0
137771380mS Sip: 17.3764.0 -1 SIPTrunk Endpoint(f4941df8) received CMReleaseComp
137771382mS SIP Call Tx: 17
CANCEL sip:0217888685@196.43.231.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.110.1:5060;rport;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Route: <sip:196.43.231.50:5060;transport=udp;lr>
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
To: <sip:0217888685@196.43.231.50>
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 CANCEL
Max-Forwards: 70
Authorization: Digest username="27217008100",realm="196.43.231.50",nonce="9456caecc06bfe602cc91e5462bc427c35b7",response="4cd6263cd7bdc3de28167d2e1062482f",uri="sip:0217888685@196.43.231.50"
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.5.0 build 972
Content-Length: 0

137771382mS SIP Tx: UDP 192.168.110.1:5060 -> 196.43.231.50:5060
CANCEL sip:0217888685@196.43.231.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.110.1:5060;rport;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Route: <sip:196.43.231.50:5060;transport=udp;lr>
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
To: <sip:0217888685@196.43.231.50>
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 CANCEL
Max-Forwards: 70
Authorization: Digest username="27217008100",realm="196.43.231.50",nonce="9456caecc06bfe602cc91e5462bc427c35b7",response="4cd6263cd7bdc3de28167d2e1062482f",uri="sip:0217888685@196.43.231.50"
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.5.0 build 972
Content-Length: 0

137771435mS SIP Rx: UDP 196.43.231.50:5060 -> 192.168.110.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.110.1:5060;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617;received=105.236.105.179
To: <sip:0217888685@196.43.231.50>
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 CANCEL
Content-Length: 0

137771437mS SIP Call Rx: 17
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.110.1:5060;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617;received=105.236.105.179
To: <sip:0217888685@196.43.231.50>
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 CANCEL
Content-Length: 0

137771453mS SIP Rx: UDP 196.43.231.50:5060 -> 192.168.110.1:5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.110.1:5060;received=105.236.105.179;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Record-Route: <sip:196.43.231.50:5060;transport=udp;lr>
To: <sip:0217888685@196.43.231.50>;tag=zceblram34yircsq.i
From: 27217008100 <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Server: Sippy
Content-Length: 0

137771455mS SIP Call Rx: 17
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.110.1:5060;received=105.236.105.179;rport=5060;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Record-Route: <sip:196.43.231.50:5060;transport=udp;lr>
To: <sip:0217888685@196.43.231.50>;tag=zceblram34yircsq.i
From: 27217008100 <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Server: Sippy
Content-Length: 0

137771458mS SIP Call Tx: 17
ACK sip:0217888685@196.43.231.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.110.1:5060;rport;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Route: <sip:196.43.231.50:5060;transport=udp;lr>
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
To: <sip:0217888685@196.43.231.50>;tag=zceblram34yircsq.i
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 ACK
Max-Forwards: 70
Authorization: Digest username="27217008100",realm="196.43.231.50",nonce="9456caecc06bfe602cc91e5462bc427c35b7",response="815e1547fa80c7d3248831a60684c4f7",uri="sip:0217888685@196.43.231.50"
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
User-Agent: IP Office 9.0.5.0 build 972
Content-Length: 0

137771458mS SIP Tx: UDP 192.168.110.1:5060 -> 196.43.231.50:5060
ACK sip:0217888685@196.43.231.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.110.1:5060;rport;branch=z9hG4bKfea6f6ba5bc3ae50e2bc8f65e49d2617
Route: <sip:196.43.231.50:5060;transport=udp;lr>
From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
To: <sip:0217888685@196.43.231.50>;tag=zceblram34yircsq.i
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 ACK
Max-Forwards: 70
Authorization: Digest username="27217008100",realm="196.43.231.50",nonce="9456caecc06bfe602cc91e5462bc427c35b7",response="815e1547fa80c7d3248831a60684c4f7",uri="sip:0217888685@196.43.231.50"
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
User-Agent: IP Office 9.0.5.0 build 972
Content-Length: 0


ACIS - Avaya Certified Implementation Specialist
ACSS - Avaya Certified Support Specialist
APSS - Avaya Professional Sales Specialist
Yes we have VoIP in Cape Town
 
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Look like your service provider changes it.

Is 27217008100 a valid DID number?
Your trying to call a number that begins with a 0 and send out CLI without zero.


From: "27217008100" <sip:27217008100@196.43.231.50>;tag=039aa8cb000d0b0f
To: <sip:0217888685@196.43.231.50>
Call-ID: c07f2d0e72c325483579df4d03d580f0
CSeq: 1700625903 INVITE
Contact: "27217008100" <sip:27217008100@192.168.110.1:5060;transport=udp>

"Trying is the first step to failure..." - Homer
 
Upvote 0 Downvote
27217008100 is the username and also DID. But the issue that the outbound number is showing anonymous on phone. I want to try force PBX to show the number the user dialled not what the SIP provider is sending back.

Thanks for reply

ACIS - Avaya Certified Implementation Specialist
ACSS - Avaya Certified Support Specialist
APSS - Avaya Professional Sales Specialist
Yes we have VoIP in Cape Town
 
Upvote 0 Downvote
I think that's correct, the service provider can change the number if it's for example routed to another destination.

I would ask the service provider why they are sending anonymous back.

"Trying is the first step to failure..." - Homer
 
Upvote 0 Downvote
I have but they say the can't and their reasoning is that it doesn't effect anyone else.

ACIS - Avaya Certified Implementation Specialist
ACSS - Avaya Certified Support Specialist
APSS - Avaya Professional Sales Specialist
Yes we have VoIP in Cape Town
 
Upvote 0 Downvote
Does it only behave that way when you call that specific number or any number?

"Trying is the first step to failure..." - Homer
 
Upvote 0 Downvote
Any number I dial out shows anonymous. If the number dialed is in the Directory than it will show the Name of the Directory entry.

ACIS - Avaya Certified Implementation Specialist
ACSS - Avaya Certified Support Specialist
APSS - Avaya Professional Sales Specialist
Yes we have VoIP in Cape Town
 
Upvote 0 Downvote
Why it shows the name if you have it in the directory is that you have "Favor Directory" set in the System or Trunk settings.

I think the service provider is wrong in changing the Contact header to anonymous on outgoing calls and IPO is right in changing the information based on the new contact information.

It's not uncommon that some smaller providers alter their SIP stack to make it fit their needs sometimes breaking the SIP RFC.


"Trying is the first step to failure..." - Homer
 
Upvote 0 Downvote
I have tried but this provider won't make change. They are one of the leading providers in SA.

I was really hoping there was a way to tell IP Office not to use the contact coming back from the Provider.

ACIS - Avaya Certified Implementation Specialist
ACSS - Avaya Certified Support Specialist
APSS - Avaya Professional Sales Specialist
Yes we have VoIP in Cape Town
 
Upvote 0 Downvote
You could solve it if you are using a SBC since it can rewrite SIP headers, also it might not send updated contact field.

Otherwise you would need to take this with Avaya.

"Trying is the first step to failure..." - Homer
 
Upvote 0 Downvote
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