IPO Server Edition SIP Trunk to ISP---PROBLEM 3

faqeel · Dec 9, 2013

Hi All,

Im installing a SIP trunk to a service provider here in KSA.

Service provider has provided with 2 ip addresses.
Near end XX.xx.xx.xx
Provider end xx.xx.xx.xx
Subnet mask: xx.xx.xx.xx
and a sIP Server Address ip: xx.xx.xx.xx

im able to ping the provider end, i have connected the SIP trunk to WAN port.
im even able to ping the SIP Server ip address, using the router as Provider end, as SIP server is on some other subnet.

Im not able to dial outside, even though my SIP trunk shows in service and Idle.

the system tries to make a call for 4 sec silence and disconnects with nothing.

nnaarrnn · Dec 9, 2013

We have had similar issues with our SIP provider. Oddly enough, after getting the One-X mobile app with VOIP working, and I can dial out fine via the app, but still can't via the deskphones...

hairlessupportmonkey · Dec 9, 2013

What firewall are you uing? is port 5060 port forwarded? Perhaps you need to use a STUN as well?

there isnt a lot of meat on the bone here.....

ACSS - SME
General Geek

faqeel · Dec 9, 2013

nnaarn,

did you open a case with avaya for this issue? seems strange.

hairless....

STC provider in KSA has provided this sip trunk.
its a direct box from STC.

from that box i have a cable directly connected to my WAN port on the server.

service provider is not giving any more info other than those 3 ip address...provider end, client end and SIP Server.

Gunnaro · Dec 9, 2013

I don't even see a bone [smile]

Am I correct assuming that there is no firewall between you and the provider?

Routing must be set up properly on the IPO: <SIP-IP>/MASK/GATEWAY/LAN2
Any authentication credentials from the provider that you forgot to put in?

If you're going to ping, use the SSA. Then you know if the IPO can reach the SIP provider.

Kind regards

Gunnar
__________________________________________________________________
Hippos have bad eyesight, but considering their weight, it’s hardly their problem

faqeel · Dec 9, 2013

Wakeup please!
Im able to ping the provider using SSA.

Hi All,

Im installing a SIP trunk to a service provider here in KSA.

Service provider has provided with 2 ip addresses.
Near end XX.xx.xx.xx
Provider end xx.xx.xx.xx
Subnet mask: xx.xx.xx.xx
and a sIP Server Address ip: xx.xx.xx.xx

im able to ping the provider end, i have connected the SIP trunk to WAN port.
im even able to ping the SIP Server ip address, using the router as Provider end, as SIP server is on some other subnet.

Im not able to dial outside, even though my SIP trunk shows in service and Idle.

the system tries to make a call for 4 sec silence and disconnects with nothing.

Service provider has not provided me any authentication details.

Gunnaro · Dec 9, 2013

Oh, I'm quite awake, but you seriously expect that we know you used SSA out of thin air? (Even in BOLD it doesn't show)

So, your system tries for 4 sec, then gives up. No line available, the line is not activated, short code for dialling is pointing at the wrong LINE, trunk needs credentials, etc... I could go on.

A trace would be nice.

I see that STC wants you to activate the service online, have you done that at this portal:

http://www.stc.com.sa/wps/wcm/conne...Office/voicePlansAndSolutions/alHatifBusiness

Your provider should have some documentation, probably available on the same site.

Kind regards

Gunnar
__________________________________________________________________
Hippos have bad eyesight, but considering their weight, it’s hardly their problem

faqeel · Dec 9, 2013

thanks for the info,

complaint has been logged with STC and the line is active, i can upload the traces tomorrow morning.

thanks for the help!

Bas1234 · Dec 9, 2013

In your ARS is there still an ? this needs to be N;

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

faqeel · Dec 9, 2013

is it necessary to use an ARS?

im using user rights short code and directly selecting the trunk group.. i think im at fault.

kindly advise how should i program this as this is my first time for a ISP sip trunk.

waiting for a comment...

Bas1234 · Dec 9, 2013

Yes otherwise SIP won't work.

ARS:

Code: N;
Feat: Dial
TelN: .
LnID: Outgoing group ID of the SIP URI.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

faqeel · Dec 9, 2013

thanks,

i will give this a try tomorrow morning, and will update it here!!! thanks a lot.

Gunnaro · Dec 9, 2013

You will find all you need to know about the setup in the KB if you like to read.

Like outbound dialling:

http://marketingtools.avaya.com/kno...s/manager/index.htm?outgoing_call_routing.htm

For outgoing call routing, a combination of system short codes and ARS entries are used. The default operation is listed below. The default configuration is not sufficient to complete call routing, additional configuration is required to route the calls from the IP Office Server Edition Primary Server to the external trunks hosted by the network.

• Summary
The default settings send all potential external calls to the IP Office Server Edition Primary Server where it is assumed those
calls will be routed to SIP trunks hosted by the primary. Additional configuration is necessary to complete the routing from the
IP Office Server Edition Primary Server. The configuration to do that will vary depending on which system is hosting the external
trunk or trunks. Examples of typical configuration changes required are given below.

• Detail
The above default operation is achieved by the following defaults:

1. IP Office Server Edition Primary Server
The following external call routing is configured by default on the IP Office Server Edition Primary Server:

a. Default System Short Code
The server has a default system short code that is used as a match for any dialling that does not match an extension number or
any other short code. This default system short code is also used for matching to digits received on calls from other systems in the network.
The default system short code used depends on the server's companding (A-Law or U-Law) setting:

• A-Law
On A-Law systems, a default ? short code is used to route any external dialing to ARS record 50:Main.
This will include matching any digits received on calls from other servers in the network that don't match extension numbers.

• U-Law
On U-Law systems, it is assumed that external calls are indicated by a 9 prefix.
A default short code 9N is used to route the digits N to ARS record 50:Main.

b. Default ARS 50:Main
A first ? short code in the ARS form routes calls to the H.323 line that goes to the IP Office Server Edition
Primary Server by using the Line Group ID of 0.

2. All Other Servers
On all other server types, the system and ARS defaults are set to route all potential external calls to the
IP Office Server Edition Primary Server.

a. Default System Short Code
A default system ? short code is present in the configuration.
This routes any dialing that has no other match to the ARS record 50:Main in the configuration of the system where the dialing occurred.

b. Default ARS 50:Main
A default ? short code in the ARS record is used to route all calls to the IP Office Server Edition Primary Server.
On expansion systems, an additional ? short code is used to route calls to the IP Office Server Edition Secondary Server
if the IP Office Server Edition Primary Server is not available for some reason.

Kind regards

Gunnar
__________________________________________________________________
Hippos have bad eyesight, but considering their weight, it’s hardly their problem

faqeel · Dec 9, 2013

Hi Guys,

thanks for all the help you are giving, atleast from BAS and Gunnaro instruction of ARS and going through the documentation im seeing myself on track.

still im not able to dial outside but now im seeing some traces that would mean something.

some times the traces shows forbidden. and sometimes the PHONE shows its waiting for line.

traces showing forbidden is as following, please let me know if you can understand this,
UNICODE-UTF8
ara

********** Warning: Logging to Screen Started **********

********** SysMonitor v9.0.0.0 build 829 [connected to 192.168.230.12 ] **********
85150mS SIP Rx: UDP 10.205.20.50:5060 -> 10.66.200.174:5060
OPTIONS sip:10.66.200.174:5060 SIP/2.0
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bK75wf137k4k453ksc21sc6s5faT30080
Call-ID: isbc5s3281kkskwf547hs6hf8pc1cphwssa4@SoftX3000
From: <sip:10.66.200.174:5060>;tag=sbc08023kakf165
To: <sip:10.66.200.174>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

85150mS Sip: Association found trunk: SIP Line (11)
85150mS SIP Reg/Opt Rx: 11
OPTIONS sip:10.66.200.174:5060 SIP/2.0
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bK75wf137k4k453ksc21sc6s5faT30080
Call-ID: isbc5s3281kkskwf547hs6hf8pc1cphwssa4@SoftX3000
From: <sip:10.66.200.174:5060>;tag=sbc08023kakf165
To: <sip:10.66.200.174>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

85150mS Sip: (e1cf8030) ExtractContactFromMessage: cannot get Contact Header 2012
85150mS Sip: SIPDialog::ExtractPortFromViaHeader remote: 10.205.20.50:5060 trunk
85150mS Sip: (e1cf8030) SendSIPResponse: rport not found - 0
85150mS Sip: (e1cf8030) SendSIPResponse: OPTIONS code 200 SENT TO 10.205.20.50 5060
85150mS SIP Reg/Opt Tx: 11
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bK75wf137k4k453ksc21sc6s5faT30080
From: <sip:10.66.200.174:5060>;tag=sbc08023kakf165
To: <sip:10.66.200.174>;tag=c89b878b485122cd
Call-ID: isbc5s3281kkskwf547hs6hf8pc1cphwssa4@SoftX3000
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (43)
Content-Type: application/sdp
Content-Length: 160

v=0
o=UserA 1839554844 143811337 IN IP4 0.0.0.0
s=Session SDP
c=IN IP4 0.0.0.0
t=0 0
m=audio 8000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
85150mS SIP Tx: UDP 10.66.200.174:5060 -> 10.205.20.50:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bK75wf137k4k453ksc21sc6s5faT30080
From: <sip:10.66.200.174:5060>;tag=sbc08023kakf165
To: <sip:10.66.200.174>;tag=c89b878b485122cd
Call-ID: isbc5s3281kkskwf547hs6hf8pc1cphwssa4@SoftX3000
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (43)
Content-Type: application/sdp
Content-Length: 160

v=0
o=UserA 1839554844 143811337 IN IP4 0.0.0.0
s=Session SDP
c=IN IP4 0.0.0.0
t=0 0
m=audio 8000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
85150mS Sip: (e1cf8030) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::FINAL(26)
86682mS Sip: SIPTrunks: Make Target voip, line group id is 22 and ip 192.168.1.14
86682mS Sip: SIP Line (10) cannot find a suitable SIP URI to dial out
86682mS Sip: SIPTrunks: Make Target voip, line group id is 22 and ip 10.205.20.50
86682mS Sip: License, Valid 1, Available 5, Consumed 0
86682mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e951aab8) received CMSetup
86682mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) CMSetup received, owner e951c0d4, dialog e950d3e8, dialling 0549496425
86682mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) Using Responding Party Number: user Extn1900
86682mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) AddPktTime cannot update ptime media attribute
86682mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) SetLocalRTPAddress to 10.66.200.174:49160 (index 0)
86683mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) SdpClone
86683mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) INVITE (method) SENT TO 10.205.20.50:5060 (reg required 0 registered 0)
86683mS SIP Call Tx: 11
INVITE sip:0549496425@10.205.20.50 SIP/2.0
Via: SIP/2.0/UDP 10.66.200.174:5060;rport;branch=z9hG4bK2c2c9b8ec55cce288cee1220a7edc344
From: "Extn1900" <sip:1900@10.205.20.50>;tag=07f9789aebfd1a9d
To: <sip:0549496425@10.205.20.50>
Call-ID: 88024ac00776370bb1fb56a8a25dc8e7
CSeq: 736191018 INVITE
Contact: "Extn1900" <sip:1900@10.66.200.174:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 8.1 (43)
Content-Length: 241

v=0
o=UserA 1363773285 237969832 IN IP4 10.66.200.174
s=Session SDP
c=IN IP4 10.66.200.174
t=0 0
m=audio 49160 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
86683mS SIP Tx: UDP 10.66.200.174:5060 -> 10.205.20.50:5060
INVITE sip:0549496425@10.205.20.50 SIP/2.0
Via: SIP/2.0/UDP 10.66.200.174:5060;rport;branch=z9hG4bK2c2c9b8ec55cce288cee1220a7edc344
From: "Extn1900" <sip:1900@10.205.20.50>;tag=07f9789aebfd1a9d
To: <sip:0549496425@10.205.20.50>
Call-ID: 88024ac00776370bb1fb56a8a25dc8e7
CSeq: 736191018 INVITE
Contact: "Extn1900" <sip:1900@10.66.200.174:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 8.1 (43)
Content-Length: 241

v=0
o=UserA 1363773285 237969832 IN IP4 10.66.200.174
s=Session SDP
c=IN IP4 10.66.200.174
t=0 0
m=audio 49160 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
86683mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_SENT(1)
86683mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_SENT(1)
86694mS SIP Rx: UDP 10.205.20.50:5060 -> 10.66.200.174:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.66.200.174:5060;branch=z9hG4bK2c2c9b8ec55cce288cee1220a7edc344;rport
Call-ID: 88024ac00776370bb1fb56a8a25dc8e7
From: "Extn1900"<sip:1900@10.205.20.50>;tag=07f9789aebfd1a9d
To: <sip:0549496425@10.205.20.50>
CSeq: 736191018 INVITE
Content-Length: 0

86694mS Sip: Find End Point 11.1009.0 2 SIPTrunk Endpoint (e951aab8) Sip CallId 88024ac00776370bb1fb56a8a25dc8e7
86694mS SIP Call Rx: 11
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.66.200.174:5060;branch=z9hG4bK2c2c9b8ec55cce288cee1220a7edc344;rport
Call-ID: 88024ac00776370bb1fb56a8a25dc8e7
From: "Extn1900"<sip:1900@10.205.20.50>;tag=07f9789aebfd1a9d
To: <sip:0549496425@10.205.20.50>
CSeq: 736191018 INVITE
Content-Length: 0

86694mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) Process SIP response dialog e950d3e8, method INVITE, CodeNum 100 in state SIPDialog::INVITE_SENT(1)
86694mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) UpdateSIPCallState SIPDialog::INVITE_SENT(1) -> SIPDialog::INV_PROV_RESP_RCVD(5)
86744mS SIP Rx: UDP 10.205.20.50:5060 -> 10.66.200.174:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.66.200.174:5060;branch=z9hG4bK2c2c9b8ec55cce288cee1220a7edc344;rport=5060
Record-Route: <sip:10.205.20.50:5060;transport=udp;lr>
Call-ID: 88024ac00776370bb1fb56a8a25dc8e7
From: "Extn1900"<sip:1900@10.205.20.50>;tag=07f9789aebfd1a9d
To: <sip:0549496425@10.205.20.50>;tag=sbc0806s3f7sh67
CSeq: 736191018 INVITE
Reason: Q.850;cause=21;text="call rejected"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

86744mS Sip: Find End Point 11.1009.0 2 SIPTrunk Endpoint (e951aab8) Sip CallId 88024ac00776370bb1fb56a8a25dc8e7
86744mS SIP Call Rx: 11
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.66.200.174:5060;branch=z9hG4bK2c2c9b8ec55cce288cee1220a7edc344;rport=5060
Record-Route: <sip:10.205.20.50:5060;transport=udp;lr>
Call-ID: 88024ac00776370bb1fb56a8a25dc8e7
From: "Extn1900"<sip:1900@10.205.20.50>;tag=07f9789aebfd1a9d
To: <sip:0549496425@10.205.20.50>;tag=sbc0806s3f7sh67
CSeq: 736191018 INVITE
Reason: Q.850;cause=21;text="call rejected"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

86744mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) Process SIP response dialog e950d3e8, method INVITE, CodeNum 403 in state SIPDialog::INV_PROV_RESP_RCVD(5)
86744mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) LR is On and route is Route: <sip:10.205.20.50:5060;transport=udp;lr>
86744mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) ExtractContactFromMessage: cannot get Contact Header 2012
86744mS Sip: 11.1009.0 2 SIPTrunk Endpoint(e950d3e8) UpdateSIPCallState SIPDialog::INV_PROV_RESP_RCVD(5) -> SIPDialog::INV_FINAL_RESP_RCVD(16)

Logs showing waiting for line is as following

ACK sip:0549496425@10.205.20.50 SIP/2.0
Via: SIP/2.0/UDP 10.66.200.174:5060;rport;branch=z9hG4bK3c323f8dfb1acc85512cea17235aff03
Route: <sip:10.205.20.50:5060;transport=udp;lr>
From: "Extn1900" <sip:1900@10.205.20.50>;tag=6ddaaf58b1fe764c
To: <sip:0549496425@10.205.20.50>;tag=sbc08021p8f3k43
Call-ID: 82a1b718828d3a3e1880b5ed8659aeba
CSeq: 231309774 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
User-Agent: IP Office 8.1 (43)
Content-Length: 0

237025mS SIP Tx: UDP 10.66.200.174:5060 -> 10.205.20.50:5060
ACK sip:0549496425@10.205.20.50 SIP/2.0
Via: SIP/2.0/UDP 10.66.200.174:5060;rport;branch=z9hG4bK3c323f8dfb1acc85512cea17235aff03
Route: <sip:10.205.20.50:5060;transport=udp;lr>
From: "Extn1900" <sip:1900@10.205.20.50>;tag=6ddaaf58b1fe764c
To: <sip:0549496425@10.205.20.50>;tag=sbc08021p8f3k43
Call-ID: 82a1b718828d3a3e1880b5ed8659aeba
CSeq: 231309774 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
User-Agent: IP Office 8.1 (43)
Content-Length: 0

237025mS Sip: 11.1011.0 3 SIPTrunk Endpoint(e7a1a5a0) UpdateSIPCallState SIPDialog::INV_FINAL_RESP_RCVD(16) -> SIPDialog::FINAL(26)
237027mS Sip: 11.1011.0 -1 SIPTrunk Endpoint(e7a1a5a0) KeepDlgOnCmCallLost SIPDialog::FINAL
237027mS Sip: ~SipTrunkEndpoint 11.1011.0 -1 SIPTrunk Endpoint
240025mS Sip: sip_indicateTimeOut Timer 10
240025mS Sip: Timer 10 callback found dialog e561f070 isbck65fksf45w3f13w7hc517cak44k358w3@SoftX3000 SIPDialog::FINAL
240025mS Sip: Completed e561f070 ... removing Dialog of CallId isbck65fksf45w3f13w7hc517cak44k358w3@SoftX3000 and State: SIPDialog::FINAL(26)
242025mS Sip: sip_indicateTimeOut Timer 4
242025mS Sip: Timer 4 callback didn't find dialog, method ACK
255058mS SIP Rx: UDP 10.205.20.50:5060 -> 10.66.200.174:5060
OPTIONS sip:10.66.200.174:5060 SIP/2.0
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bK8wcfpaa24s1fhs15c7ff3c13cT21822
Call-ID: isbc1fw383h74a3a44wwh2k2scsahw4k377h@SoftX3000
From: <sip:10.66.200.174:5060>;tag=sbc08066k832f61
To: <sip:10.66.200.174>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

255058mS Sip: Association found trunk: SIP Line (11)
255058mS SIP Reg/Opt Rx: 11
OPTIONS sip:10.66.200.174:5060 SIP/2.0
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bK8wcfpaa24s1fhs15c7ff3c13cT21822
Call-ID: isbc1fw383h74a3a44wwh2k2scsahw4k377h@SoftX3000
From: <sip:10.66.200.174:5060>;tag=sbc08066k832f61
To: <sip:10.66.200.174>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

255058mS Sip: (e561ee00) ExtractContactFromMessage: cannot get Contact Header 2012
255059mS Sip: SIPDialog::ExtractPortFromViaHeader remote: 10.205.20.50:5060 trunk
255059mS Sip: (e561ee00) SendSIPResponse: rport not found - 0
255059mS Sip: (e561ee00) SendSIPResponse: OPTIONS code 200 SENT TO 10.205.20.50 5060
255059mS SIP Reg/Opt Tx: 11
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bK8wcfpaa24s1fhs15c7ff3c13cT21822
From: <sip:10.66.200.174:5060>;tag=sbc08066k832f61
To: <sip:10.66.200.174>;tag=82e6c455b8b69784
Call-ID: isbc1fw383h74a3a44wwh2k2scsahw4k377h@SoftX3000
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (43)
Content-Type: application/sdp
Content-Length: 160

v=0
o=UserA 2071124963 943304884 IN IP4 0.0.0.0
s=Session SDP
c=IN IP4 0.0.0.0
t=0 0
m=audio 8000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
255059mS SIP Tx: UDP 10.66.200.174:5060 -> 10.205.20.50:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bK8wcfpaa24s1fhs15c7ff3c13cT21822
From: <sip:10.66.200.174:5060>;tag=sbc08066k832f61
To: <sip:10.66.200.174>;tag=82e6c455b8b69784
Call-ID: isbc1fw383h74a3a44wwh2k2scsahw4k377h@SoftX3000
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (43)
Content-Type: application/sdp
Content-Length: 160

v=0
o=UserA 2071124963 943304884 IN IP4 0.0.0.0
s=Session SDP
c=IN IP4 0.0.0.0
t=0 0
m=audio 8000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
255059mS Sip: (e561ee00) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::FINAL(26)

thanks,

chrisag · Dec 9, 2013

Some providers want to sent them a valid number in the from field in uri.
You are sending extension number.
Sip uri make all internal data and on user sip tap put one of your numbers.

faqeel · Dec 9, 2013

we have 100 DID's how will we use those?

faqeel · Dec 9, 2013

which field are you exactly talking about?

Local URI: <over here the number will come?>
Contact: Use internal Data
Display Name: Use Internal Data
PAI: Use Internal Data

I have tried that as well still im getting Forbidden, and phone shows rejected. here are the traces.

91318mS SIP Rx: UDP 10.205.20.50:5060 -> 10.66.200.174:5060
OPTIONS sip:10.66.200.174:5060 SIP/2.0
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKsc2h1686sp782314s43531177T08243
Call-ID: isbca1sk886637aa5w8s477swpkcaf885pw5@SoftX3000
From: <sip:10.66.200.174:5060>;tag=sbc0804a35p7w18
To: <sip:10.66.200.174>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

91318mS Sip: Association found trunk: SIP Line (11)
91318mS SIP Reg/Opt Rx: 11
OPTIONS sip:10.66.200.174:5060 SIP/2.0
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKsc2h1686sp782314s43531177T08243
Call-ID: isbca1sk886637aa5w8s477swpkcaf885pw5@SoftX3000
From: <sip:10.66.200.174:5060>;tag=sbc0804a35p7w18
To: <sip:10.66.200.174>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

91318mS Sip: (e14fe690) ExtractContactFromMessage: cannot get Contact Header 2012
91318mS Sip: SIPDialog::ExtractPortFromViaHeader remote: 10.205.20.50:5060 trunk
91318mS Sip: (e14fe690) SendSIPResponse: rport not found - 0
91318mS Sip: (e14fe690) SendSIPResponse: OPTIONS code 200 SENT TO 10.205.20.50 5060
91318mS SIP Reg/Opt Tx: 11
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKsc2h1686sp782314s43531177T08243
From: <sip:10.66.200.174:5060>;tag=sbc0804a35p7w18
To: <sip:10.66.200.174>;tag=bee7e9952652558b
Call-ID: isbca1sk886637aa5w8s477swpkcaf885pw5@SoftX3000
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (43)
Content-Type: application/sdp
Content-Length: 160

v=0
o=UserA 1970455819 392912442 IN IP4 0.0.0.0
s=Session SDP
c=IN IP4 0.0.0.0
t=0 0
m=audio 8000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
91318mS SIP Tx: UDP 10.66.200.174:5060 -> 10.205.20.50:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKsc2h1686sp782314s43531177T08243
From: <sip:10.66.200.174:5060>;tag=sbc0804a35p7w18
To: <sip:10.66.200.174>;tag=bee7e9952652558b
Call-ID: isbca1sk886637aa5w8s477swpkcaf885pw5@SoftX3000
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (43)
Content-Type: application/sdp
Content-Length: 160

v=0
o=UserA 1970455819 392912442 IN IP4 0.0.0.0
s=Session SDP
c=IN IP4 0.0.0.0
t=0 0
m=audio 8000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
91318mS Sip: (e14fe690) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::FINAL(26)
93928mS Sip: SIPTrunks: Make Target voip, line group id is 22 and ip 10.205.20.50
93928mS Sip: License, Valid 1, Available 5, Consumed 0
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e731ab28) received CMSetup
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) CMSetup received, owner e731c144, dialog e730f9f0, dialling 0549496425
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) Using Responding Party Number: user Extn1900
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) AddPktTime cannot update ptime media attribute
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) SetLocalRTPAddress to 10.66.200.174:49154 (index 0)
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) SdpClone
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) INVITE (method) SENT TO 10.205.20.50:5060 (reg required 0 registered 0)
93928mS SIP Call Tx: 11
INVITE sip:0549496425@10.205.20.50 SIP/2.0
Via: SIP/2.0/UDP 10.66.200.174:5060;rport;branch=z9hG4bK8a10616f575a7508c6f2fb6ffe0e7470
From: "Extn1900" <sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>
Call-ID: 86eac52201693044e37416c97c1dbf2b
CSeq: 1706326714 INVITE
Contact: "Extn1900" <sip:1900@10.66.200.174:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 8.1 (43)
Content-Length: 217

v=0
o=UserA 1313787625 783743142 IN IP4 10.66.200.174
s=Session SDP
c=IN IP4 10.66.200.174
t=0 0
m=audio 49154 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
93928mS SIP Tx: UDP 10.66.200.174:5060 -> 10.205.20.50:5060
INVITE sip:0549496425@10.205.20.50 SIP/2.0
Via: SIP/2.0/UDP 10.66.200.174:5060;rport;branch=z9hG4bK8a10616f575a7508c6f2fb6ffe0e7470
From: "Extn1900" <sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>
Call-ID: 86eac52201693044e37416c97c1dbf2b
CSeq: 1706326714 INVITE
Contact: "Extn1900" <sip:1900@10.66.200.174:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 8.1 (43)
Content-Length: 217

v=0
o=UserA 1313787625 783743142 IN IP4 10.66.200.174
s=Session SDP
c=IN IP4 10.66.200.174
t=0 0
m=audio 49154 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_SENT(1)
93928mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_SENT(1)
93943mS SIP Rx: UDP 10.205.20.50:5060 -> 10.66.200.174:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.66.200.174:5060;branch=z9hG4bK8a10616f575a7508c6f2fb6ffe0e7470;rport
Call-ID: 86eac52201693044e37416c97c1dbf2b
From: "Extn1900"<sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>
CSeq: 1706326714 INVITE
Content-Length: 0

93943mS Sip: Find End Point 11.1006.0 1 SIPTrunk Endpoint (e731ab28) Sip CallId 86eac52201693044e37416c97c1dbf2b
93943mS SIP Call Rx: 11
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.66.200.174:5060;branch=z9hG4bK8a10616f575a7508c6f2fb6ffe0e7470;rport
Call-ID: 86eac52201693044e37416c97c1dbf2b
From: "Extn1900"<sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>
CSeq: 1706326714 INVITE
Content-Length: 0

93943mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) Process SIP response dialog e730f9f0, method INVITE, CodeNum 100 in state SIPDialog::INVITE_SENT(1)
93943mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) UpdateSIPCallState SIPDialog::INVITE_SENT(1) -> SIPDialog::INV_PROV_RESP_RCVD(5)
94004mS SIP Rx: UDP 10.205.20.50:5060 -> 10.66.200.174:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.66.200.174:5060;branch=z9hG4bK8a10616f575a7508c6f2fb6ffe0e7470;rport=5060
Record-Route: <sip:10.205.20.50:5060;transport=udp;lr>
Call-ID: 86eac52201693044e37416c97c1dbf2b
From: "Extn1900"<sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>;tag=sbc08076563aak6
CSeq: 1706326714 INVITE
Reason: Q.850;cause=21;text="call rejected"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

94004mS Sip: Find End Point 11.1006.0 1 SIPTrunk Endpoint (e731ab28) Sip CallId 86eac52201693044e37416c97c1dbf2b
94004mS SIP Call Rx: 11
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.66.200.174:5060;branch=z9hG4bK8a10616f575a7508c6f2fb6ffe0e7470;rport=5060
Record-Route: <sip:10.205.20.50:5060;transport=udp;lr>
Call-ID: 86eac52201693044e37416c97c1dbf2b
From: "Extn1900"<sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>;tag=sbc08076563aak6
CSeq: 1706326714 INVITE
Reason: Q.850;cause=21;text="call rejected"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

94004mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) Process SIP response dialog e730f9f0, method INVITE, CodeNum 403 in state SIPDialog::INV_PROV_RESP_RCVD(5)
94004mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) LR is On and route is Route: <sip:10.205.20.50:5060;transport=udp;lr>
94004mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) ExtractContactFromMessage: cannot get Contact Header 2012
94004mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) UpdateSIPCallState SIPDialog::INV_PROV_RESP_RCVD(5) -> SIPDialog::INV_FINAL_RESP_RCVD(16)
94004mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) SendSIPRequest: ACK SENT TO 10.205.20.50 5060
94004mS SIP Call Tx: 11
ACK sip:0549496425@10.205.20.50 SIP/2.0
Via: SIP/2.0/UDP 10.66.200.174:5060;rport;branch=z9hG4bK8a10616f575a7508c6f2fb6ffe0e7470
Route: <sip:10.205.20.50:5060;transport=udp;lr>
From: "Extn1900" <sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>;tag=sbc08076563aak6
Call-ID: 86eac52201693044e37416c97c1dbf2b
CSeq: 1706326714 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
User-Agent: IP Office 8.1 (43)
Content-Length: 0

94004mS SIP Tx: UDP 10.66.200.174:5060 -> 10.205.20.50:5060
ACK sip:0549496425@10.205.20.50 SIP/2.0
Via: SIP/2.0/UDP 10.66.200.174:5060;rport;branch=z9hG4bK8a10616f575a7508c6f2fb6ffe0e7470
Route: <sip:10.205.20.50:5060;transport=udp;lr>
From: "Extn1900" <sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>;tag=sbc08076563aak6
Call-ID: 86eac52201693044e37416c97c1dbf2b
CSeq: 1706326714 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
User-Agent: IP Office 8.1 (43)
Content-Length: 0

94004mS Sip: 11.1006.0 1 SIPTrunk Endpoint(e730f9f0) UpdateSIPCallState SIPDialog::INV_FINAL_RESP_RCVD(16) -> SIPDialog::FINAL(26)
94006mS Sip: 11.1006.0 -1 SIPTrunk Endpoint(e730f9f0) KeepDlgOnCmCallLost SIPDialog::FINAL
94006mS Sip: ~SipTrunkEndpoint 11.1006.0 -1 SIPTrunk Endpoint
96319mS Sip: sip_indicateTimeOut Timer 10
96319mS Sip: Timer 10 callback found dialog e14fe690 isbca1sk886637aa5w8s477swpkcaf885pw5@SoftX3000 SIPDialog::FINAL
96319mS Sip: Completed e14fe690 ... removing Dialog of CallId isbca1sk886637aa5w8s477swpkcaf885pw5@SoftX3000 and State: SIPDialog::FINAL(26)
99004mS Sip: sip_indicateTimeOut Timer 4
99004mS Sip: Timer 4 callback didn't find dialog, method ACK

hairlessupportmonkey · Dec 9, 2013

This forum really isn't a place for you to train.

statements like -

is it necessary to use an ARS?

I'm using user rights short code and directly selecting the trunk group.. i think im at fault.

kindly advise how should i program this as this is my first time for a ISP sip trunk

really doesn't fill anyone with any confidence, and least of all your customer.

You need to learn to read the trace.

Code:

From: "Extn1900"<sip:966138041900@10.205.20.50>;tag=6269a65ec8019722
To: <sip:0549496425@10.205.20.50>;tag=sbc08076563aak6
CSeq: 1706326714 INVITE
Reason: Q.850;cause=21;text="call rejected"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

ACSS - SME
General Geek

Gunnaro · Dec 9, 2013

So, STC use Softx3000, bitchy Huawei.

I have one customer connected to the exact same thing. If you don't have a service that allows any number sent from A-end, it has to be your local number.

It looks like you have added country code for Saudi Arabia (966138041900).
Change that to 0138041900 (or just 138041900 if you don't need the 0 for domestic calls, I think you do)

Kind regards

Gunnar
__________________________________________________________________
Hippos have bad eyesight, but considering their weight, it’s hardly their problem

faqeel · Dec 10, 2013

Thanks,
STC is looking on his part.

Gunnaro thanks for the help!!! im really getting a lot from you guys!

Incoming and outgoing started working.
the Local uri was the 7 digit number only!

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