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Incoming sip breakout on PRI

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zoomatomic

zoomatomic

Technical User
Jun 2, 2009
82
IE
Hi all

I have a sip trunk setup to go to a third party SBC, calls to the SBC are fine and calls from the SBC are presenting to users on the IP Office.

My question is: Is it possible for calls to come into the IP Office from the SBC and break out on the local PRI?

IPO is V11.

Any help appreciated

Graham

ACSS - SME
ASPS - SME

 
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Thanks derfloh, my incoming route for the sip trunk is blank incoming number and . as destination.
The SBC provider has said they tried sending the number with a leading 9 as I have a short code 9N Dial N ARS 50(PRI) on my IPO.
How do get an incoming call to break out on my pri?

Thanks

ACSS - SME
ASPS - SME

 
Upvote 0 Downvote
Depending on the number format you get with the incoming call you have to adjust the number. If the SIP provider calls +49xxxxx you cannot call that number over PRI because you cannot send + over ISDN.

You have to watch the inbound SIP messages and what IPO sends to PRI.

Need some help with IP Office?
 
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Hi derfloh

See example below, customer dialled 90877438781 (Leading 9), I can see the call coming in on the sip trunk (line 30) but it doesn't match to my system short code 9N so doesn't send the call onto my PRI.

Do I need to add anything to my incoming call route?

Any help appreciated.

11:01:40 2736450070mS SIP Rx: TCP 52.236.59.18:33556 -> 192.168.101.1:5060
INVITE sip:90877438781@sbc.flogas.ie;user=phone SIP/2.0
Via: SIP/2.0/TCP 52.236.59.18:5068;alias;branch=z9hG4bKac1513219613
Max-Forwards: 69
From: "Evros Martin" <sip:+353412149767@sip.pstnhub.microsoft.com:5061;user=phone>;tag=1c1045321900
To: <sip:90877438781@sbc.flogas.ie;user=phone>
Call-ID: 67776425715201911139@52.236.59.18
CSeq: 1 INVITE
Contact: <sip:52.236.59.18:5068;transport=tcp;x-i=de031e26-4bbe-4d1f-9db9-f65de4e36a23;x-c=/v1/ngc/call/3c8fd756ee3d5ec6a00040e74aa10a79/d/8/aad62f5cb5344748b7536044470e6f4f>
Supported: timer,sdp-anat
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
Session-Expires: 3600
MIN-SE: 300
User-Agent: Mediant VE SBC/v.7.20A.204.222
P-Asserted-Identity: <sip:+353412149767@sip.pstnhub.microsoft.com>
Content-Type: application/sdp
Content-Length: 624

v=0
o=- 1519740359 1830168011 IN IP4 52.236.59.18
s=session
c=IN IP4 52.236.59.18
b=CT:10000000
t=0 0
m=audio 7265 RTP/AVP 104 117 9 103 111 18 0 8 97 118 101 13
c=IN IP4 52.236.59.18
a=rtcp:7266 IN IP4 52.236.59.18
a=label:main-audio
a=sendrecv
a=rtpmap:104 SILK/16000
a=rtpmap:117 G722/8000/2
a=rtpmap:9 G722/8000
a=rtpmap:103 SILK/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=ptime:20
11:01:40 2736450074mS Sip: TCP packet known trunk owner SIP Line (30)
11:01:40 2736450074mS SIP Call Rx: 30
INVITE sip:90877438781@sbc.flogas.ie;user=phone SIP/2.0
Via: SIP/2.0/TCP 52.236.59.18:5068;alias;branch=z9hG4bKac1513219613
Max-Forwards: 69
From: "Evros Martin" <sip:+353412149767@sip.pstnhub.microsoft.com:5061;user=phone>;tag=1c1045321900
To: <sip:90877438781@sbc.flogas.ie;user=phone>
Call-ID: 67776425715201911139@52.236.59.18
CSeq: 1 INVITE
Contact: <sip:52.236.59.18:5068;transport=tcp;x-i=de031e26-4bbe-4d1f-9db9-f65de4e36a23;x-c=/v1/ngc/call/3c8fd756ee3d5ec6a00040e74aa10a79/d/8/aad62f5cb5344748b7536044470e6f4f>
Supported: timer,sdp-anat
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
Session-Expires: 3600
MIN-SE: 300
User-Agent: Mediant VE SBC/v.7.20A.204.222
P-Asserted-Identity: <sip:+353412149767@sip.pstnhub.microsoft.com>
Content-Type: application/sdp
Content-Length: 624

v=0
o=- 1519740359 1830168011 IN IP4 52.236.59.18
s=session
c=IN IP4 52.236.59.18
b=CT:10000000
t=0 0
m=audio 7265 RTP/AVP 104 117 9 103 111 18 0 8 97 118 101 13
c=IN IP4 52.236.59.18
a=rtcp:7266 IN IP4 52.236.59.18
a=label:main-audio
a=sendrecv
a=rtpmap:104 SILK/16000
a=rtpmap:117 G722/8000/2
a=rtpmap:9 G722/8000
a=rtpmap:103 SILK/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=ptime:20
11:01:40 2736450075mS CMCallEvt: 0000000000000000 0.213296.0 -1 BaseEP: NEW CMEndpoint f4b64ba4 TOTAL NOW=19 CALL_LIST=9
11:01:40 2736450075mS Sip: SIP Line (30): sip_trunk_config_items 50020000, sip_trunk_config_items_2 00000000, voip.flags 00000948
11:01:40 2736450075mS Sip: SIPDialog f4b54358 created, dialogs 1 txn_keys 3 video 0 presentation 0 camera 0 unsupp audio 0
11:01:40 2736450075mS Sip: 0000000000000000 0.213296.0 -1 SIPTrunk Endpoint(f4b54358) SetUnIntTransactionCondition to UnInt_None
11:01:40 2736450076mS Sip: SIP Line (30) GetNetworkTopologySource Use Network Topology badly configured
11:01:40 2736450076mS Sip: SIP Line (30) GetNetworkTopologySource Use Network Topology badly configured
11:01:40 2736450076mS Sip: SipTCPUser 9386 has 1 dialog open (AttachDialogToSipTCPUser)
11:01:40 2736450076mS Sip: SIP Line (30): License, Valid 1, Available 5, Consumed 0
11:01:40 2736450077mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 52.236.59.18:5068 to 52.236.59.18:5068
11:01:40 2736450078mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) PreProcessMsg calling CheckMinSEField, MinSE value is set to 300 in the header
11:01:40 2736450078mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) PreProcessMsg calling CheckSessionExpiresField, session expires is 3600 refresher_is_ipo 0
11:01:40 2736450078mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) PreProcessMsg calling CheckMinSEField, MinSE value is set to 3600 in the header
11:01:40 2736450078mS Sip: FindContactParameters mFarDisplayString <> mFarDisplayNumber <+353412149767> restricted 0
11:01:40 2736450079mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) Cloned
11:01:40 2736450079mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent_by: 52.236.59.18:5068 trunk
11:01:40 2736450079mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent by transport: SIP/2.0/TCP trunk
11:01:40 2736450079mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 52.236.59.18:5068 to 52.236.59.18:5068
11:01:40 2736450080mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) SendSIPResponse: INVITE code 100 SENT TO 52.236.59.18 5068
11:01:40 2736450080mS SIP Call Tx: 30
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 52.236.59.18:5068;alias;branch=z9hG4bKac1513219613
From: "Evros Martin" <sip:+353412149767@sip.pstnhub.microsoft.com:5061;user=phone>;tag=1c1045321900
Call-ID: 67776425715201911139@52.236.59.18
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.0.4.0.0 build 74
Content-Length: 0
To: <sip:90877438781@sbc.flogas.ie;user=phone>;tag=3862dcb12ba509e5

11:01:40 2736450081mS SIP Tx: TCP 192.168.101.1:5060 -> 52.236.59.18:33556
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 52.236.59.18:5068;alias;branch=z9hG4bKac1513219613
From: "Evros Martin" <sip:+353412149767@sip.pstnhub.microsoft.com:5061;user=phone>;tag=1c1045321900
Call-ID: 67776425715201911139@52.236.59.18
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.0.4.0.0 build 74
Content-Length: 0
To: <sip:90877438781@sbc.flogas.ie;user=phone>;tag=3862dcb12ba509e5

11:01:40 2736450081mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) INVITE Received ep c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b64ba4), dialog f4b54358
11:01:40 2736450081mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_RCVD(9)
11:01:40 2736450082mS Sip: SIP Line (30): Incoming SIP Call Failed.
11:01:40 2736450083mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) Present Call, no match (90877438781) from URI in To header.
11:01:40 2736450083mS Sip: SIP Line (30): Incoming SIP Call Failed.
11:01:40 2736450084mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) Present Call, no match (90877438781) from URI in To header or (90877438781) from request URI
11:01:40 2736450084mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent_by: 52.236.59.18:5068 trunk
11:01:40 2736450084mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent by transport: SIP/2.0/TCP trunk
11:01:40 2736450084mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 52.236.59.18:5068 to 52.236.59.18:5068
11:01:40 2736450085mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) SendSIPResponse: INVITE code 404 SENT TO 52.236.59.18 5068
11:01:40 2736450085mS SIP Call Tx: 30
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 52.236.59.18:5068;alias;branch=z9hG4bKac1513219613
From: "Evros Martin" <sip:+353412149767@sip.pstnhub.microsoft.com:5061;user=phone>;tag=1c1045321900
Call-ID: 67776425715201911139@52.236.59.18
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.0.4.0.0 build 74
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
Content-Length: 0
To: <sip:90877438781@sbc.flogas.ie;user=phone>;tag=3862dcb12ba509e5

11:01:40 2736450086mS SIP Tx: TCP 192.168.101.1:5060 -> 52.236.59.18:33556
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 52.236.59.18:5068;alias;branch=z9hG4bKac1513219613
From: "Evros Martin" <sip:+353412149767@sip.pstnhub.microsoft.com:5061;user=phone>;tag=1c1045321900
Call-ID: 67776425715201911139@52.236.59.18
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.0.4.0.0 build 74
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
Content-Length: 0
To: <sip:90877438781@sbc.flogas.ie;user=phone>;tag=3862dcb12ba509e5

11:01:40 2736450087mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) UpdateSIPCallState SIPDialog::INVITE_RCVD(9) -> SIPDialog::FINAL(28)
11:01:40 2736450099mS SIP Rx: TCP 52.236.59.18:33556 -> 192.168.101.1:5060
ACK sip:90877438781@sbc.flogas.ie;user=phone SIP/2.0
Via: SIP/2.0/TCP 52.236.59.18:5068;alias;branch=z9hG4bKac1513219613
Max-Forwards: 70
From: "Evros Martin" <sip:+353412149767@sip.pstnhub.microsoft.com:5061;user=phone>;tag=1c1045321900
To: <sip:90877438781@sbc.flogas.ie;user=phone>;tag=3862dcb12ba509e5
Call-ID: 67776425715201911139@52.236.59.18
CSeq: 1 ACK
Contact: <sip:52.236.59.18:5068;transport=tcp>
User-Agent: Mediant VE SBC/v.7.20A.204.222
Content-Length: 0

11:01:40 2736450101mS Sip: TCP packet known trunk owner SIP Line (30)
11:01:40 2736450101mS SIP Call Rx: 30
ACK sip:90877438781@sbc.flogas.ie;user=phone SIP/2.0
Via: SIP/2.0/TCP 52.236.59.18:5068;alias;branch=z9hG4bKac1513219613
Max-Forwards: 70
From: "Evros Martin" <sip:+353412149767@sip.pstnhub.microsoft.com:5061;user=phone>;tag=1c1045321900
To: <sip:90877438781@sbc.flogas.ie;user=phone>;tag=3862dcb12ba509e5
Call-ID: 67776425715201911139@52.236.59.18
CSeq: 1 ACK
Contact: <sip:52.236.59.18:5068;transport=tcp>
User-Agent: Mediant VE SBC/v.7.20A.204.222
Content-Length: 0

11:01:40 2736450101mS Sip: Find End Point2 c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint (f4b64ba4) Sip CallId 67776425715201911139@52.236.59.18
11:01:40 2736450101mS Sip: c0a8650100034130 30.213296.1 -1 SIPTrunk Endpoint(f4b54358) Process SIP request dialog f4b54358, method ACK in state SIPDialog::FINAL(28)


ACSS - SME
ASPS - SME

 
Upvote 0 Downvote
Is the SIP URI set to Auto/Auto/Auto for incoming calls? If not IPO will search for a user or group with matching SIP entry.

Need some help with IP Office?
 
Upvote 0 Downvote
my incoming route for the sip trunk is blank incoming number and . as destination" does not work at all
Setting any route to a blank destination field, may cause the incoming number to be checked against system short codes for a match. This may lead to the call being rerouted off-switch.
So have a blank incoming call route without a destination using the SIP incoming line ID, SSA will show a error but ignore that.

A system short code to get your config going:
SC=9N
TN=N
Line ID = Pri outgoing line group
 
Upvote 0 Downvote
Hi derfloh

URI is set to "use internal data". Should I change it to auto? Will changing affect incoming calls for extensions?

Hi intrigrant

So I have to change the incoming destination to blank?

Thanks



ACSS - SME
ASPS - SME

 
Upvote 0 Downvote
If you change it to "auto" for incoming calls you have to configure Incoming Call Routing as you do with ISDN Trunks. But yes, you have to do that if you want to be able to adress incoming calls not only to objects that have the SIP tab filled (users, groups, voicemail).

Need some help with IP Office?
 
Upvote 0 Downvote
Yes, as per documentation....
Setting any route to a blank destination field, may cause the incoming number to be checked against system short codes for a match. This may lead to the call being rerouted off-switch.
 
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