Tek-Tips is the largest IT community on the Internet today!

Members share and learn making Tek-Tips Forums the best source of peer-reviewed technical information on the Internet!

Register Log in
  • Congratulations bkrike on being selected by the Tek-Tips community for having the most helpful posts in the forums last week. Way to Go!

OUTGOING SIP "NUMBER BUSY"

Status
Not open for further replies.
Hazmat77

Hazmat77

Technical User
Jun 27, 2002
15
I am trying to configure a SIP trunk on an IP500.
The Carrier is Xchange Telecom, they work with IP Authentication.
I am able to get Incoming Calls, but when i try to place an outgoing call i get a message "NUMBER BUSY"

Any ideas ?
 
Sort by date Sort by votes
>>>> Any ideas ?

NO!


Works fine here, you might get more result if you give us the right info about the version, setup ars/sip trunk.
Also a trace helps a lot.

What did your BP say?

Avaya_Red.gif

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

Dain Bramaged
___________________________________________
 
Upvote 0 Downvote
Please explain Version ?
I have ougoing setup as follows
S/C: 7N;
Feature: Dial
Tel #: N"@xxx.xxx.xxx.xxx"
Line group: 21 (Which was setup in Lin>SipUri)
 
Upvote 0 Downvote
Is your ip route (gateway) correct?

Goto;

Line > SIP Line > SIP Line tab > Use Network Topology Info:NONE (if none is selected then is uses the ip route)

use sip.sipmedia.com or 69.1.236.33 as ip address in ITSP and ARS.

Avaya_Red.gif

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

Dain Bramaged
___________________________________________
 
Upvote 0 Downvote
How does the IP Route need to be Setup ?

Do I need to use LAN 2 as well ?

I have "Use Network Topology Info" Set to "Lan 1"
(When i have it set to "None" I get only one-way Audio)

Sipmedia IP's will not work I am using a direct Xchange product not a Sipmedia
{sip13.xchangetele.com - 66.128.2.136}\


Thanks

 
Upvote 0 Downvote
In the avaya manager add an IP Route; 0.0.0.0/0.0.0.0/xxx.xxx.xxx.xxx
(xxx.xxx.xxx.xxx is you gateway address)

I think xchangetele.com uses the SIP server from SipMedia ( first try it with the one you have.

No need for LAN2 that all depends on how you network/vlan is setup.

Try to post a SysMon trace.

Avaya_Red.gif

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

Dain Bramaged
___________________________________________
 
Upvote 0 Downvote
I have the route setup that way already,

Xchange has their own seperate sip server for a hosted solution they are currently selling under the name central office (
{the begining is an outgoing call which didnt go out "Number Busy", and the end is an incoming call i placed which came thru OK}

46517325mS Sip: License, Valid 1, Available 2, Consumed 0
46517328mS Sip: 20.3803.0 26 SIPTrunk Endpoint(f55d4614) received CMSetup
46517330mS SIP Call Tx: 20
INVITE sip:19175676509@66.128.66.136 SIP/2.0
Via: SIP/2.0/UDP 69.112.200.97:5060;rport;branch=z9hG4bK528d6127ee5893ff7c2492663792311a
From: <Tel:+6464482024>;tag=516f1cfb01e3d94f
To: <sip:19175676509@66.128.66.136>
Call-ID: d4dcd10818b17f1093513c34ad4b86c3@69.112.200.97
CSeq: 795529660 INVITE
Contact: <Tel:+6464482024>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 300

v=0
o=UserA 629733186 2932402325 IN IP4 69.112.200.97
s=Session SDP
c=IN IP4 69.112.200.97
t=0 0
m=audio 49152 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
46517356mS SIP Call Rx: 20
SIP/2.0 400 Invalid Contact URI Scheme
Via: SIP/2.0/UDP 69.112.200.97:5060;received=69.112.200.97;branch=z9hG4bK528d6127ee5893ff7c2492663792311a;rport=15060
From: <Tel:+6464482024>;tag=516f1cfb01e3d94f
To: <sip:19175676509@66.128.66.136>;tag=aprqngfrt-gerds6s5abkpb
Call-ID: d4dcd10818b17f1093513c34ad4b86c3@69.112.200.97
CSeq: 795529660 INVITE

46517356mS Sip: Find End Point 20.3803.0 26 SIPTrunk Endpoint (f55d4614) Sip CallId d4dcd10818b17f1093513c34ad4b86c3@69.112.200.97
46517358mS SIP Call Tx: 20
ACK sip:19175676509@66.128.66.136 SIP/2.0
Via: SIP/2.0/UDP 69.112.200.97:5060;rport;branch=z9hG4bK528d6127ee5893ff7c2492663792311a
From: <Tel:+6464482024>;tag=516f1cfb01e3d94f
To: <sip:19175676509@66.128.66.136>;tag=aprqngfrt-gerds6s5abkpb
Call-ID: d4dcd10818b17f1093513c34ad4b86c3@69.112.200.97
CSeq: 795529660 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0

46517362mS Sip: ~SipTrunkEndpoint 20.3803.0 -1 SIPTrunk Endpoint
46520806mS SIP Reg/Opt Rx: 20
OPTIONS sip:metaswitch@69.112.200.97:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bKnh1a1p30c8fhbisbo1k0.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info
Max-Forwards: 69
Call-ID: 61384BD2@sip10.xchangetele.com
From: <sip:metaswitch@66.128.2.136:5060>;tag=sip10.xchangetele.com+1+0+e29342a8
CSeq: 702948779 OPTIONS
Organization: MetaSwitch
Supported: 100rel
Content-Length: 0
Contact: <sip:metaswitch@66.128.2.136:5060;transport=udp>
To: <sip:metaswitch@69.112.200.97>

46520807mS Sip: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
46520808mS Sip: 20.3804.1 -1 SIPTrunk Endpoint(f55d3540) SendSIPResponse: OPTIONS code 200 SENT TO 66.128.2.136 5060
46520809mS SIP Reg/Opt Tx: 20
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bKnh1a1p30c8fhbisbo1k0.1
From: <sip:metaswitch@66.128.2.136:5060>;tag=sip10.xchangetele.com+1+0+e29342a8
To: <sip:metaswitch@69.112.200.97>;tag=4a3bf00ddd34c92e
Call-ID: 61384BD2@sip10.xchangetele.com
CSeq: 702948779 OPTIONS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 301

v=0
o=UserA 3664763522 3814459829 IN IP4 69.112.200.97
s=Session SDP
c=IN IP4 69.112.200.97
t=0 0
m=audio 8000 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
46525809mS Sip: Completed f55d3540 ... removing Dialog of CallId 61384BD2@sip10.xchangetele.com and State: SIPDialog::FINAL(26)
46525809mS Sip: ~SipTrunkEndpoint 20.3804.1 -1 SIPTrunk Endpoint
46526672mS SIP Reg/Opt Rx: phone
REGISTER sip:192.168.2.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.102:2051;branch=z9hG4bK-3vlls9db3ljd;rport
From: "IP Phone" <sip:201@192.168.2.250>;tag=gnaoaicd41
To: "IP Phone" <sip:201@192.168.2.250>
Call-ID: 3c267032216c-fw38yoyl980j
CSeq: 6395 REGISTER
Max-Forwards: 70
Contact: <sip:201@192.168.2.102:2051;line=xppy1gff>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:95b6e875-730b-49ad-8147-ebb9206c735e>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INV User-Agent: snom300/7.3.14
User-Agent: snom300/7.3.14
Supported: gruu
Allow-Events: dialog
X-Real-IP: 192.168.2.102
Expires: 3600
Content-Length: 0

46526672mS Sip: (f55d6c4c) SendSIPResponse: REGISTER code 200 SENT TO 192.168.2.102 2051
46526673mS SIP Reg/Opt Tx: phone
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.102:2051;branch=z9hG4bK-3vlls9db3ljd;rport
From: "IP Phone" <sip:201@192.168.2.250>;tag=gnaoaicd41
To: "IP Phone" <sip:201@192.168.2.250>;tag=eb0ef1c24ba50d99
Call-ID: 3c267032216c-fw38yoyl980j
CSeq: 6395 REGISTER
User-Agent: IP Office 5.0 (8)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH
Contact: <sip:201@192.168.2.102:2051;line=xppy1gff>
Date: Wed, 02 Sep 2009 09:22:50 GMT
Expires: 180
Content-Length: 0

46528218mS Sip: SIP Line (20): Option timer expired
46528218mS Sip: Creating a EndPoint to refresh binding of 0
46528219mS SIP Reg/Opt Tx: 20
OPTIONS sip:Unknown@66.128.2.136 SIP/2.0
Via: SIP/2.0/UDP 69.112.200.97:5060;rport;branch=z9hG4bK018c5a75121dafae33a28f1b8d75a50e
From: <Tel:+Unknown>;tag=64580fa9efa11bdd
To: <sip:Unknown@66.128.2.136>
Call-ID: a22a9a86d092aaf98bf9130f2c209d2b@69.112.200.97
CSeq: 1032945775 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0

46528240mS SIP Reg/Opt Rx: 20
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 69.112.200.97:5060;received=69.112.200.97;branch=z9hG4bK018c5a75121dafae33a28f1b8d75a50e;rport=15060
From: <Tel:+Unknown>;tag=64580fa9efa11bdd
To: <sip:Unknown@66.128.2.136>;tag=aprqngfrt-kpqi76l7h40v6
Call-ID: a22a9a86d092aaf98bf9130f2c209d2b@69.112.200.97
CSeq: 1032945775 OPTIONS

46528241mS Sip: Find End Point 0.3805.0 -1 SIPTrunk Endpoint (f55d4614) Sip CallId a22a9a86d092aaf98bf9130f2c209d2b@69.112.200.97
46528241mS Sip: ~SipTrunkEndpoint 0.3805.0 -1 SIPTrunk Endpoint
46531641mS SIP Call Rx: 20
INVITE sip:6464482024@69.112.200.97:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bKc41rms30a8a0picml6s1.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info
Max-Forwards: 69
Call-ID: A779714D@sip10.xchangetele.com
From: NEW YORK <sip:7182656922@66.128.2.136:5060>;tag=sip10.xchangetele.com+1+3c4c0+4fe4aa9d;isup-oli=00
To: <sip:6464482024@69.112.200.97>
CSeq: 868716687 INVITE
Expires: 180
Organization: MetaSwitch
Supported: 100rel
Content-Length: 166
Content-Type: application/sdp
Contact: NEW YORK <sip:7182656922@66.128.2.136:5060;transport=udp>;isup-oli=00
P-Asserted-Identity: NEW YORK <sip:7182656922@66.128.2.136:5060>

v=0
o=- 2062292348 2062292348 IN IP4 66.128.2.136
s=-
c=IN IP4 66.128.2.136
t=0 0
m=audio 51784 RTP/AVP 18 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20
46531642mS Sip: License, Valid 1, Available 2, Consumed 0
46531642mS Sip: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
46531643mS Sip: 20.3806.1 -1 SIPTrunk Endpoint(f55d3540) SendSIPResponse: INVITE code 100 SENT TO 66.128.2.136 5060
46531644mS SIP Call Tx: 20
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bKc41rms30a8a0picml6s1.1
From: NEW YORK <sip:7182656922@66.128.2.136:5060>;tag=sip10.xchangetele.com+1+3c4c0+4fe4aa9d;isup-oli=00
To: <sip:6464482024@69.112.200.97>;tag=e8959bdb52d16ae5
Call-ID: A779714D@sip10.xchangetele.com
CSeq: 868716687 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0

46531650mS Sip: 20.3806.1 27 SIPTrunk Endpoint(f55d4614) received CMProceeding
46531652mS Sip: 20.3806.1 27 SIPTrunk Endpoint(f55d4614) received CMAlerting
46531653mS Sip: 20.3806.1 27 SIPTrunk Endpoint(f55d3540) SendSIPResponse: INVITE code 180 SENT TO 66.128.2.136 5060
46531654mS SIP Call Tx: 20
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bKc41rms30a8a0picml6s1.1
From: NEW YORK <sip:7182656922@66.128.2.136:5060>;tag=sip10.xchangetele.com+1+3c4c0+4fe4aa9d;isup-oli=00
To: <sip:6464482024@69.112.200.97>;tag=e8959bdb52d16ae5
Call-ID: A779714D@sip10.xchangetele.com
CSeq: 868716687 INVITE
Contact: "6464482024" <sip:6464482024@69.112.200.97:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0

46536872mS Sip: 20.3806.1 27 SIPTrunk Endpoint(f55d4614) received CMConnect
46536873mS Sip: 20.3806.1 27 SIPTrunk Endpoint(f55d3540) SendSIPResponse: INVITE code 200 SENT TO 66.128.2.136 5060
46536874mS SIP Call Tx: 20
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bKc41rms30a8a0picml6s1.1
From: NEW YORK <sip:7182656922@66.128.2.136:5060>;tag=sip10.xchangetele.com+1+3c4c0+4fe4aa9d;isup-oli=00
To: <sip:6464482024@69.112.200.97>;tag=e8959bdb52d16ae5
Call-ID: A779714D@sip10.xchangetele.com
CSeq: 868716687 INVITE
Contact: "6464482024" <sip:6464482024@69.112.200.97:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 229

v=0
o=UserA 2328384158 1257165114 IN IP4 69.112.200.97
s=Session SDP
c=IN IP4 69.112.200.97
t=0 0
m=audio 49152 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
46536903mS SIP Call Rx: 20
ACK sip:6464482024@69.112.200.97:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bK9lq2af10e0dgbg0un700.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info
Max-Forwards: 69
Call-ID: A779714D@sip10.xchangetele.com
From: NEW YORK <sip:7182656922@66.128.2.136:5060>;tag=sip10.xchangetele.com+1+3c4c0+4fe4aa9d;isup-oli=00
To: <sip:6464482024@69.112.200.97>;tag=e8959bdb52d16ae5
CSeq: 868716687 ACK
Contact: NEW YORK <sip:7182656922@66.128.2.136:5060;transport=udp>;isup-oli=00
Organization: MetaSwitch
Content-Length: 0

46536903mS Sip: Find End Point 20.3806.1 27 SIPTrunk Endpoint (f55d4614) Sip CallId A779714D@sip10.xchangetele.com
46538297mS SIP Call Tx: 20
BYE sip:7182656922@66.128.2.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 69.112.200.97:5060;rport;branch=z9hG4bK6500fdfbaf96fa0925e94f9e1b0c2316
From: "6464482024" <sip:6464482024@69.112.200.97>;tag=e8959bdb52d16ae5
To: "NEW YORK" <sip:7182656922@66.128.2.136>;tag=sip10.xchangetele.com+1+3c4c0+4fe4aa9d
Call-ID: A779714D@sip10.xchangetele.com
CSeq: 868716688 BYE
Contact: "6464482024" <sip:6464482024@69.112.200.97:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO
Content-Length: 0

46538322mS SIP Call Rx: 20
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.112.200.97:5060;received=69.112.200.97;branch=z9hG4bK6500fdfbaf96fa0925e94f9e1b0c2316;rport=15060
From: "6464482024" <sip:6464482024@69.112.200.97>;tag=e8959bdb52d16ae5
To: "NEW YORK" <sip:7182656922@66.128.2.136>;tag=sip10.xchangetele.com+1+3c4c0+4fe4aa9d
Call-ID: A779714D@sip10.xchangetele.com
CSeq: 868716688 BYE
Server: DC-SIP/2.0
Organization: MetaSwitch
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info
Supported: 100rel
Content-Length: 0

46538322mS Sip: Find End Point 20.3806.1 -1 SIPTrunk Endpoint (f55d4614) Sip CallId A779714D@sip10.xchangetele.com
 
Upvote 0 Downvote
From: <Tel:+Unknown>;tag=64580fa9efa11bdd

You have ticked "Use tel URI" untick that option.

Avaya_Red.gif

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

Dain Bramaged
___________________________________________
 
Upvote 0 Downvote
Tried that already, Same thing

what i also tried is in "SIP URI"

Setting everything to "Use User Data" Then i get "Call Rejected"

When everything is set to "Use Authentication Name" I get "Number Busy
 
Upvote 0 Downvote
Leave it on "Use Authentication Name" make sure you have filled in the right username and password, sometimes the provider want's a PhoneNumber as a Username verify that with your provider.
Also make sure the the sip10.xchangetele.com is able to handle "SIP trunk" instead of "SIP extension".
One of the other posts you pasted sip13.xchangetele.com is this a typo?!

Avaya_Red.gif

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

Dain Bramaged
___________________________________________
 
Upvote 0 Downvote
That is the way i have it setup right now.
sip13.xchangetele.com is the Proxy given to me by Xchange

Are you using a Sipmedia SIP Trunk?
 
Upvote 0 Downvote
No but it can't be hard.

Most providers can see what going wrong on their system.
Try to set it up using X-lite or another sipphone to see if that works.
Is the sip line a prepaid line? We have some providers who work with prepaid.

Avaya_Red.gif

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

Dain Bramaged
___________________________________________
 
Upvote 0 Downvote
Status
Not open for further replies.

Similar threads

dsm600rr
  • Locked
  • Question
Avaya J100 over Site-to-Site VPN, &quot;SIP Proxy List is Empty&quot; 2
Replies
1
Views
489
dsm600rr
dsm600rr
rsv23
  • Locked
  • Question
SIP Trunk Routing
Replies
3
Views
419
rsv23
rsv23
R
  • Question
cannot find a suitable SIP URI to dial out | SIP Error
Replies
6
Views
462
MikeeOK
MikeeOK
OmegaStar
  • Locked
  • Question
Avaya SBC TCP Port outgoing Invite
Replies
0
Views
226
OmegaStar
OmegaStar
phoneDTC
  • Locked
  • Question
SIP Trunk Question
Replies
5
Views
573
phoneDTC
phoneDTC

Part and Inventory Search

Sponsor

Back
Top