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Creating a SIP trunk to CallWithUs service


cmrabet
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Hi

 

I am trying to set up a SIP register trunk for outgoing calls with the service "Call With Us". However I follow the instructions given by them and I get the following error from PBXnSIP service:

 

403 Forbidden (Bad auth) (Registration failed, retry after 60 seconds)

 

The info that my provider gave me is:

 

SIP/IAX client configuration

Different SIP clients (hardware SIP phones, ATAs and software ones) have different configuration screens, but all have a common set of configuration parameters. In the table below, username and password are your 9-digit long login name and 6-digit long password you received in the sign-up confirmation e-mail.

Configuration parameter Value

SIP server (or proxy, or domain) sip.callwithus.com or uk.callwithus.com if you are in Europe, Africa or Middle East.

SIP proxy (or "Outbound Proxy") leave blank

STUN server stun.callwithus.com, port 3478

Username (or User ID) username

Password password

Auth name (or Auth ID) username

Display Name (used for callerId information when you place a call) Your name

Register (or Send registration request) Yes

Register Expiry (or ReRegistration interval) 120 sec (2 minutes)

Silence suppression (or Voice Activity Detection) On

Use DNS SRV Yes

 

How should be this information set up in the "options" that PBXnSIP gives?:

 

Name:

Type:

Direction

Display Name:

Account:

Domain:

Username:

Password:

Password (repeat):

Outbound Proxy:

CO Lines:

Dialog Permissions:

Codec Preference:

Proposed Duration (s):

Keepalive Time:

Send email on status change: yesno

Strict RTP Routing: yesno

Avoid RFC4122 (UUID): yesno

Accept Redirect: yesno

Interpret SIP URI always as telephone number: yesno

 

Thanks.

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Hi

 

I am trying to set up a SIP register trunk for outgoing calls with the service "Call With Us". However I follow the instructions given by them and I get the following error from PBXnSIP service:

 

403 Forbidden (Bad auth) (Registration failed, retry after 60 seconds)

 

The info that my provider gave me is:

 

SIP/IAX client configuration

Different SIP clients (hardware SIP phones, ATAs and software ones) have different configuration screens, but all have a common set of configuration parameters. In the table below, username and password are your 9-digit long login name and 6-digit long password you received in the sign-up confirmation e-mail.

Configuration parameter Value

SIP server (or proxy, or domain) sip.callwithus.com or uk.callwithus.com if you are in Europe, Africa or Middle East.

SIP proxy (or "Outbound Proxy") leave blank

STUN server stun.callwithus.com, port 3478

Username (or User ID) username

Password password

Auth name (or Auth ID) username

Display Name (used for callerId information when you place a call) Your name

Register (or Send registration request) Yes

Register Expiry (or ReRegistration interval) 120 sec (2 minutes)

Silence suppression (or Voice Activity Detection) On

Use DNS SRV Yes

 

How should be this information set up in the "options" that PBXnSIP gives?:

 

Thanks.

 

 

Type: SIP Registration

Direction : inbound and outbound

Display Name: username

Account: username

Domain: sip.callwithus.com (depending on where you are)

Username: username

Password: password

Password (repeat): password

Outbound Proxy: sip.callwithus.com (depending on where you are)

 

Trunk ANI : caller id that you want the outsiders see

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Solved;

 

The trunk now is registered successfully. I created a very simple dialplan just to test the service:

PREF TRUNK PATTERN REPLACEMENT

 

100 Unassigned

100 CallWithUs 00*

 

 

But whenever I dial on my phone: 00<whatever> I just get busy tone. I tryied to look for some monitorin tool in the PBXnSIP web interface to see what is going on but I didn't find anything. With a soft SIP phone the calls are working.

 

Any idea?

 

By the way, thanks for your fast and effective support!

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But whenever I dial on my phone: 00<whatever> I just get busy tone. I tryied to look for some monitorin tool in the PBXnSIP web interface to see what is going on but I didn't find anything. With a soft SIP phone the calls are working.

 

What replacement are you using? Maybe also use 00* as replacement. Otherwise the PBX will cut off the first two zeros and then I can understand the carrier rejects the call.

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What replacement are you using? Maybe also use 00* as replacement. Otherwise the PBX will cut off the first two zeros and then I can understand the carrier rejects the call.

 

My SIP provider is expecting the following structure:

 

[CountryCode][phonenumber]

 

I made some changes so now the users should use the prefix '1' in order to place long distance calls.

 

So if somebody wants to call for instance to Spain (+ 34 956606060) must dial on the phone: 134956606060, but my SIP provider is only expecting 34956606060, so my dial plan will be:

 

Pref: 100

Trunk: CallWithUs

Pattern: 1*

Replacement: *

 

I tryed this but I am still getting the busy tone (if the problem was a bad number string compossition, I would get an "air" noise, as my provider says, so I am afraid that the Trunk is not even connecting to CallwithUs, but it is registered).

 

Thanks.

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My SIP provider is expecting the following structure:

 

[CountryCode][phonenumber]

 

I made some changes so now the users should use the prefix '1' in order to place long distance calls.

 

So if somebody wants to call for instance to Spain (+ 34 956606060) must dial on the phone: 134956606060, but my SIP provider is only expecting 34956606060, so my dial plan will be:

 

Pref: 100

Trunk: CallWithUs

Pattern: 1*

Replacement: *

 

I tryed this but I am still getting the busy tone (if the problem was a bad number string compossition, I would get an "air" noise, as my provider says, so I am afraid that the Trunk is not even connecting to CallwithUs, but it is registered).

 

That looks okay to me...

 

Can you turn SIP logging on and show the INVITE packet that is sent to the provider and the response which is sent back?

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That looks okay to me...

 

Can you turn SIP logging on and show the INVITE packet that is sent to the provider and the response which is sent back?

 

This is what I find after trying to place a call:

 

 

[9] 2008/11/27 12:06:19: Resolve 13158: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 12:06:19: Resolve 13158: a udp 192.168.1.101 5060

[9] 2008/11/27 12:06:19: Resolve 13158: udp 192.168.1.101 5060

[9] 2008/11/27 12:06:22: Resolve 13159: aaaa udp 192.168.1.102 5060

[9] 2008/11/27 12:06:22: Resolve 13159: a udp 192.168.1.102 5060

[9] 2008/11/27 12:06:22: Resolve 13159: udp 192.168.1.102 5060

[8] 2008/11/27 12:06:23: DNS: dns_naptr t2h.callwithus.com expired

[8] 2008/11/27 12:06:23: DNS: dns_srv _sips._tcp.t2h.callwithus.com expired

[8] 2008/11/27 12:06:23: DNS: dns_srv _sip._tcp.t2h.callwithus.com expired

[8] 2008/11/27 12:06:23: DNS: dns_srv _sip._udp.t2h.callwithus.com expired

[8] 2008/11/27 12:06:23: DNS: dns_aaaa t2h.callwithus.com expired

[9] 2008/11/27 12:06:24: Resolve 13160: url sip:sip.callwithus.com

[9] 2008/11/27 12:06:24: Resolve 13160: naptr sip.callwithus.com

[8] 2008/11/27 12:06:24: DNS: Add dns_naptr t2h.callwithus.com (ttl=60)

[9] 2008/11/27 12:06:24: Resolve 13160: naptr sip.callwithus.com

[9] 2008/11/27 12:06:24: Resolve 13160: srv tls _sips._tcp.t2h.callwithus.com

[8] 2008/11/27 12:06:24: DNS: Add dns_srv _sips._tcp.t2h.callwithus.com (ttl=60)

[9] 2008/11/27 12:06:24: Resolve 13160: srv tls _sips._tcp.t2h.callwithus.com

[9] 2008/11/27 12:06:24: Resolve 13160: srv tcp _sip._tcp.t2h.callwithus.com

[8] 2008/11/27 12:06:24: DNS: Add dns_srv _sip._tcp.t2h.callwithus.com (ttl=60)

[9] 2008/11/27 12:06:24: Resolve 13160: srv tcp _sip._tcp.t2h.callwithus.com

[9] 2008/11/27 12:06:24: Resolve 13160: srv udp _sip._udp.t2h.callwithus.com

[8] 2008/11/27 12:06:24: DNS: Add dns_srv _sip._udp.t2h.callwithus.com (ttl=60)

[9] 2008/11/27 12:06:24: Resolve 13160: srv udp _sip._udp.t2h.callwithus.com

[9] 2008/11/27 12:06:24: Resolve 13160: aaaa udp t2h.callwithus.com 5060

[8] 2008/11/27 12:06:25: DNS: Add dns_aaaa t2h.callwithus.com (ttl=60)

[9] 2008/11/27 12:06:25: Resolve 13160: aaaa udp t2h.callwithus.com 5060

[9] 2008/11/27 12:06:25: Resolve 13160: a udp t2h.callwithus.com 5060

[9] 2008/11/27 12:06:25: Resolve 13160: udp 38.99.70.46 5060

[8] 2008/11/27 12:06:25: Trunk 4 (CallWithUs) has outbound proxy udp:38.99.70.46:5060

[9] 2008/11/27 12:06:25: Resolve 13161: url sip:sip.callwithus.com

[9] 2008/11/27 12:06:25: Resolve 13161: naptr sip.callwithus.com

[9] 2008/11/27 12:06:25: Resolve 13161: srv tls _sips._tcp.t2h.callwithus.com

[9] 2008/11/27 12:06:25: Resolve 13161: srv tcp _sip._tcp.t2h.callwithus.com

[9] 2008/11/27 12:06:25: Resolve 13161: srv udp _sip._udp.t2h.callwithus.com

[9] 2008/11/27 12:06:25: Resolve 13161: aaaa udp t2h.callwithus.com 5060

[9] 2008/11/27 12:06:25: Resolve 13161: a udp t2h.callwithus.com 5060

[9] 2008/11/27 12:06:25: Resolve 13161: udp 38.99.70.46 5060

[8] 2008/11/27 12:06:25: Answer challenge with username 756165920

[9] 2008/11/27 12:06:25: Resolve 13162: udp 38.99.70.46 5060 udp:1

[9] 2008/11/27 12:06:25: Message repetition, packet dropped

[9] 2008/11/27 12:06:31: Resolve 13163: aaaa udp 192.168.1.100 5060

[9] 2008/11/27 12:06:31: Resolve 13163: a udp 192.168.1.100 5060

[9] 2008/11/27 12:06:31: Resolve 13163: udp 192.168.1.100 5060

[9] 2008/11/27 12:06:47: Resolve 13164: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 12:06:47: Resolve 13164: a udp 192.168.1.101 5060

[9] 2008/11/27 12:06:47: Resolve 13164: udp 192.168.1.101 5060

[9] 2008/11/27 12:06:51: Resolve 13165: aaaa udp 192.168.1.102 5060

[9] 2008/11/27 12:06:51: Resolve 13165: a udp 192.168.1.102 5060

[9] 2008/11/27 12:06:51: Resolve 13165: udp 192.168.1.102 5060

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I think we need to take a look at the SIP packets in detail, see http://wiki.pbxnsip.com/index.php/Log_Setup#SIP_Logging on how to turn that logging on. Maybe just use the watch list to see the traffic between the provider and the PBX (put 38.99.70.46 there).

 

I enabled the SIP login with as follows:

 

Log REGISTER: No

Log SUBSCRIBE/NOTIFY: No

Log OPTIONS: No

Log Other Messages (e.g. INVITE): Yes

Log Watch List (IP): 77.68.40.174

Log Watch List: 9

 

The IP now is 77.68.40.174 because I changed the server for the trunk to another one (uk.callwithus.com instead of sip.callwithus.com).

 

Looking at the log file after trying to call I get:

 

[9] 2008/11/27 13:26:26: Resolve 13998: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:26: Resolve 13998: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:26: Resolve 13998: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:26: Resolve 13999: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:26: Resolve 13999: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:26: Resolve 13999: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:32: Resolve 14000: aaaa udp 192.168.1.102 5060

[9] 2008/11/27 13:26:32: Resolve 14000: a udp 192.168.1.102 5060

[9] 2008/11/27 13:26:32: Resolve 14000: udp 192.168.1.102 5060

[9] 2008/11/27 13:26:34: SIP Rx udp:192.168.1.101:5060:

INVITE sip:134976100550@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK395425231300022963;rport

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 1 INVITE

Contact: <sip:101@192.168.1.101:5060>

Max-Forwards: 70

Supported: replaces

User-Agent: Voip Phone 1.0

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

Content-Type: application/sdp

Content-Length: 314

 

v=0

o=101 19402104 29892208 IN IP4 192.168.1.101

s=A conversation

c=IN IP4 192.168.1.101

t=0 0

m=audio 10130 RTP/AVP 8 4 18 0 9 101

a=rtpmap:8 PCMA/8000

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:9 G722/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

[9] 2008/11/27 13:26:34: UDP: Opening socket on port 51776

[9] 2008/11/27 13:26:34: UDP: Opening socket on port 51777

[9] 2008/11/27 13:26:34: UDPv6: Opening socket on port 51776

[9] 2008/11/27 13:26:34: UDPv6: Opening socket on port 51777

[8] 2008/11/27 13:26:34: Could not find a trunk (1 trunks)

[9] 2008/11/27 13:26:34: Resolve 14001: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14001: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14001: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK395425231300022963;rport=5060

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 1 INVITE

Content-Length: 0

 

[9] 2008/11/27 13:26:34: Resolve 14002: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14002: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14002: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK395425231300022963;rport=5060

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 1 INVITE

User-Agent: pbxnsip-PBX/3.0.1.3023

WWW-Authenticate: Digest realm="192.168.1.3",nonce="328ab85258b1439aaa23b9d20df141c7",domain="sip:134976100550@192.168.1.3",algorithm=MD5

Content-Length: 0

 

[9] 2008/11/27 13:26:34: SIP Rx udp:192.168.1.101:5060:

ACK sip:134976100550@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK395425231300022963;rport

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 1 ACK

Max-Forwards: 70

Content-Length: 0

 

[9] 2008/11/27 13:26:34: SIP Rx udp:192.168.1.101:5060:

INVITE sip:134976100550@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 2 INVITE

Contact: <sip:101@192.168.1.101:5060>

Authorization: Digest username="101", realm="192.168.1.3", nonce="328ab85258b1439aaa23b9d20df141c7", uri="sip:134976100550@192.168.1.3", response="37e39a61d64c4f7b64c170faafe5b4f0", algorithm=MD5

Max-Forwards: 70

Supported: replaces

User-Agent: Voip Phone 1.0

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

Content-Type: application/sdp

Content-Length: 314

 

v=0

o=101 19402104 29892208 IN IP4 192.168.1.101

s=A conversation

c=IN IP4 192.168.1.101

t=0 0

m=audio 10130 RTP/AVP 8 4 18 0 9 101

a=rtpmap:8 PCMA/8000

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:9 G722/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

[8] 2008/11/27 13:26:34: Tagging request with existing tag

[6] 2008/11/27 13:26:34: Sending RTP for 5592174505805-222652809710666@192.168.1.101#195fb6c832 to 192.168.1.101:10130

[9] 2008/11/27 13:26:34: Resolve 14003: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14003: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14003: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 2 INVITE

Content-Length: 0

 

[5] 2008/11/27 13:26:34: No dial plan for user 101 available

[9] 2008/11/27 13:26:34: Resolve 14004: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14004: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14004: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 2 INVITE

Contact: <sip:101@192.168.1.3:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

 

[9] 2008/11/27 13:26:34: Resolve 14005: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14005: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14005: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 2 INVITE

Contact: <sip:101@192.168.1.3:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

 

[9] 2008/11/27 13:26:34: SIP Rx udp:192.168.1.101:5060:

ACK sip:134976100550@192.168.1.3 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 2 ACK

Max-Forwards: 70

Content-Length: 0

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[9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 2 INVITE

Content-Length: 0

 

[5] 2008/11/27 13:26:34: No dial plan for user 101 available

[9] 2008/11/27 13:26:34: Resolve 14004: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14004: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14004: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 2 INVITE

Contact: <sip:101@192.168.1.3:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

 

[9] 2008/11/27 13:26:34: Resolve 14005: aaaa udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14005: a udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: Resolve 14005: udp 192.168.1.101 5060

[9] 2008/11/27 13:26:34: SIP Tx udp:192.168.1.101:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK24738223901826911228;rport=5060

From: Chakir Mrabet <sip:101@192.168.1.3>;tag=2369525718

To: 134976100550 <sip:134976100550@192.168.1.3>;tag=195fb6c832

Call-ID: 5592174505805-222652809710666@192.168.1.101

CSeq: 2 INVITE

Contact: <sip:101@192.168.1.3:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

 

Looks like '101' does not have any dial plan associated. You can do that either by going to domain setting or the extension setting to select the newly created dial plan

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Solved.

 

The user 101 didn't have a dial plan associated because it was set to “Domain default”.

 

I thought that all the users in a certain domain are by default associated to all the dial plans that are defined in the domain. I have misunderstood the concept of dial plan in PBXnSIP and finally I associated 101 to the dial plan that contents routes to the SIP provider trunk. Now it is working fine.

 

I have been working a long time with Asterisk and my error is to think in “Asterisk way” when I am playing around with PBXnSIP!

 

Thank you very much for your support, I am very amused by such fast answers; this really makes worthy to have decided to go with your product!

 

Now I am going to investigate if PBXnSIP is easy to setup a FAX, but this is another issue for another thread...

 

Regards!

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  • 6 months later...

But whenever I punch on my phone: 00 I just get active tone. I tryied to attending for some monitorin apparatus in the PBXnSIP web interface to see what is traveling on but I didn't acquisition anything. With a bendable SIP buzz the calls are working. So if somebody wants to alarm for instance to Spain (+ 34 956606060) have to punch on the phone: 134956606060, but my SIP provider is alone assured 34956606060, so my punch plan will be,

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