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Calls Declined


pbxuser911
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i try making a call using a certain trunk, the call wont connect and i get a declined error on the phone

but when making the same call from a 2nd extension on the same domain and the same phone, it works

 

***Note all IP address, the proxy server Domain name, Phone numbers that i chose to keep private for the customer, has been replaced with, xx.xx.xxx.xxx companya.mydomain.com 9784256666***

the phone i tryed calling was 19785551212

 

3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:1025:

INVITE sip:19785551212@companya.mydomain.comSIP/2.0

Via: SIP/2.0/UDP 71.58.196.187:1025;branch=z9hG4bK-fe7f0737

From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com>

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 101 INVITE

Max-Forwards: 70

Contact: "107" <sip:107@xx.xx.xxx.xxx:1025>

Expires: 240

User-Agent: Linksys/SPA962-6.1.3(a)

Content-Length: 395

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: replaces

Content-Type: application/sdp

 

v=0

o=- 30452 30452 IN IP4 192.168.1.106

s=-

c=IN IP4 xx.xx.xxx.xxx

t=0 0

m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

 

[3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-fe7f0737

From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 101 INVITE

Content-Length: 0

 

 

[3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-fe7f0737

From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 101 INVITE

User-Agent: pbx/3.1.1.3110

WWW-Authenticate: Digest realm="companya.mydomain.com",nonce="bc93d1c2f115fdfa416d8bef1fd53b2e",domain="sip:19785551212@companya.mydomain.com",algorithm=MD5

Content-Length: 0

 

 

[3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:1025:

ACK sip:19785551212@companya.mydomain.comSIP/2.0

Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-fe7f0737

From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 101 ACK

Max-Forwards: 70

Contact: "107" <sip:107@xx.xx.xxx.xxx:1025>

User-Agent: Linksys/SPA962-6.1.3(a)

Content-Length: 0

 

 

[3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:1025:

INVITE sip:19785551212@companya.mydomain.com.com SIP/2.0

Via: SIP/2.0/UDP 71.58.196.187:1025;branch=z9hG4bK-10e069bb

From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com>

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 102 INVITE

Max-Forwards: 70

Authorization: Digest username="107",realm="companya.mydomain.com",nonce="bc93d1c2f115fdfa416d8bef1fd53b2e",uri="sip:19785551212@companya.mydomain.com",algorithm=MD5,response="5e3fe565d38eb9b0719748db7f5584a1"

Contact: "107" <sip:107@xx.xx.xxx.xxx:1025>

Expires: 240

User-Agent: Linksys/SPA962-6.1.3(a)

Content-Length: 395

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: replaces

Content-Type: application/sdp

 

v=0

o=- 30452 30452 IN IP4 192.168.1.106

s=-

c=IN IP4 xx.xx.xxx.xxx

t=0 0

m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

 

[3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-10e069bb

From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 102 INVITE

Content-Length: 0

 

 

[3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:5060:

INVITE sip:19785551212@xx.xx.xxx.xxx;user=phone SIP/2.0

Via: SIP/2.0/UDP xx.xx.xxx.xxx:5060;branch=z9hG4bK-0d16d66d82498c26250e21c8181d4eb3;rport

From: "W: Office 1" <sip:9784256666@companya.mydomain.com;user=phone>;tag=648644319

To: <sip:19785551212@xx.xx.xxx.xxx;user=phone>

Call-ID: 7b24ce66@pbx

CSeq: 6743 INVITE

Max-Forwards: 70

Contact: <sip:9784256666@xx.xx.xxx.xxx:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbx/3.1.1.3110

P-Asserted-Identity: "W: Office 1" <sip:9784256666@companya.mydomain.com ;user=phone>

Content-Type: application/sdp

Content-Length: 300

 

v=0

o=- 1344194633 1344194633 IN IP4 xx.xx.xxx.xxx

s=-

c=IN IP4 xx.xx.xxx.xxx

t=0 0

m=audio 41368 RTP/AVP 0 8 3 18 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[3] 2008/12/31 22:35:09: Could not open WAV file audio_moh/noise.wav

[3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-10e069bb

From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 102 INVITE

Contact: <sip:107@74.206.239.196:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbx/3.1.1.3110

Content-Type: application/sdp

Content-Length: 289

 

v=0

o=- 1165514095 1165514095 IN IP4 xx.xx.xxx.xxx

s=-

c=IN IP4 xx.xx.xxx.xxx

t=0 0

m=audio 19926 RTP/AVP 0 8 18 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:30

a=sendrecv

 

[3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:5060:

SIP/2.0 100 Giving a try

Via: SIP/2.0/UDP xx.xx.xxx.xxx:5060;branch=z9hG4bK-0d16d66d82498c26250e21c8181d4eb3;rport=5060

From: "W: Office 1" <sip:9784256666@companya.mydomain.com ;user=phone>;tag=648644319

To: <sip:19785551212@xx.xx.xxx.xxx;user=phone>

Call-ID: 7b24ce66@pbx

CSeq: 6743 INVITE

Server: OpenSIPS (1.4.2-notls (x86_64/linux))

Content-Length: 0

 

 

[3] 2008/12/31 22:35:09: Could not open WAV file audio_moh/noise.wav

[3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:5060:

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 74.206.239.196:5060;received=74.206.239.196;branch=z9hG4bK-0d16d66d82498c26250e21c8181d4eb3;rport=5060

From: "W: Office 1" <sip:9784256666@companya.mydomain.com ;user=phone>;tag=648644319

To: <sip:19785551212@xx.xx.xxx.xxx;user=phone>;tag=as6b435cf2

Call-ID: 7b24ce66@pbx

CSeq: 6743 INVITE

User-Agent: TSG_Global_GW

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:19785551212@xx.xx.xxx.xxx>

Content-Length: 0

 

 

[3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:5060:

ACK sip:19785551212@xx.xx.xxx.xxx;user=phone SIP/2.0

Via: SIP/2.0/UDP xx.xx.xxx.xxx:5060;branch=z9hG4bK-0d16d66d82498c26250e21c8181d4eb3;rport

From: "W: Office 1" <sip:9784256666@companya.mydomain.com;user=phone>;tag=648644319

To: <sip:19785551212@69.25.128.195;user=phone>;tag=as6b435cf2

Call-ID: 7b24ce66@pbx

CSeq: 6743 ACK

Max-Forwards: 70

Contact: <sip:7183840099@xx.xx.xxx.xxx:5060;transport=udp>

P-Asserted-Identity: "W: Office 1" <sip:9784256666@companya.mydomain.com;user=phone>

Content-Length: 0

 

 

[3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 71.58.196.187:1025;branch=z9hG4bK-10e069bb

From: "107" <sip:107@companya.mydomain.com >;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com >;tag=edf6f65045

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 102 INVITE

Contact: <sip:107@xx.xx.xxx.xxx:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbx/3.1.1.3110

Content-Length: 0

 

 

[3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:1025:

ACK sip:19785551212@companya.mydomain.com SIP/2.0

Via: SIP/2.0/UDP 71.58.196.187:1025;branch=z9hG4bK-10e069bb

From: "107" <sip:107@companya.mydomain.com >;tag=52f8d1546848e82fo2

To: <sip:19785551212@companya.mydomain.com >;tag=edf6f65045

Call-ID: 25c3fff8-6b34e9db@192.168.1.106

CSeq: 102 ACK

Max-Forwards: 70

Authorization: Digest username="107",realm="companya.mydomain.com ",nonce="bc93d1c2f115fdfa416d8bef1fd53b2e",uri="sip:19785551212@companya.mydomain.com ",algorithm=MD5,response="5e3fe565d38eb9b0719748db7f5584a1"

Contact: "107" <sip:107@xx.xx.xxx.xxx:1025>

User-Agent: Linksys/SPA962-6.1.3(a)

Content-Length: 0

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Ok here is what I figured out the problem was, but need helping finding a way around it

 

We have the name of the extensions W: Office 1

the termination provider has declined it due to the :, once I remove : everything works good..

 

Would anyone know how do i send the caller ID in quotations? supposedly that should work even there is a :

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Ok here is what I figured out the problem was, but need helping finding a way around it

 

We have the name of the extensions W: Office 1

the termination provider has declined it due to the :, once I remove : everything works good..

 

Would anyone know how do i send the caller ID in quotations? supposedly that should work even there is a :

 

Looks like the "TSG_Global_GW" has a little issue with RFC3261:

 

	  quoted-string  =  SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE
  qdtext		 =  LWS / %x21 / %x23-5B / %x5D-7E
					/ UTF8-NONASCII

 

Unfortunately, the PBX just follows the RFC rules; there is no way to perform a special escaping of characters. The best solution if the gateway gets a software update with a fix.

 

BTW is seems that the audio_moh files are not installed ("Could not open WAV file audio_moh/noise.wav").

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as far as the sound file missing, i renamed the NOISE (comfort tone) sound so it doesnt play

 

I would just reduce the volume of the noise. Having completely silent line is irritating as users believe the line is "dead". That was one of the findings when they introduced ISDN in Europe. People got confused when they did not hear anything (no kidding).

 

That is probably why the name is "comfort noise"...

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i told TSG what you recomended me telling them about the RFC,

Looks like the "TSG_Global_GW" has a little issue with RFC3261:

 

CODE

quoted-string = SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE

qdtext = LWS / %x21 / %x23-5B / %x5D-7E

/ UTF8-NONASCII

 

 

Unfortunately, the PBX just follows the RFC rules; there is no way to perform a special escaping of characters. The best solution if the gateway gets a software update with a fix.

 

 

and this is what he replyed back to me----

 

 

I will look into the call issue, however in terms of the RFC it states the following note the “ “ around the name, however when we get calls from your switch the “ “ are missing and thus when a charter other then a-z is used it causes the system to break.

 

Link for Ref http://www.ietf.org/rfc/rfc3261.txt

 

20.20 From

 

The From header field indicates the initiator of the request. This

may be different from the initiator of the dialog. Requests sent by

the callee to the caller use the callee's address in the From header

field.

 

The optional "display-name" is meant to be rendered by a human user

interface. A system SHOULD use the display name "Anonymous" if the

identity of the client is to remain hidden. Even if the "display-

name" is empty, the "name-addr" form MUST be used if the "addr-spec"

contains a comma, question mark, or semicolon. Syntax issues are

discussed in Section 7.3.1.

 

Two From header fields are equivalent if their URIs match, and their

parameters match. Extension parameters in one header field, not

present in the other are ignored for the purposes of comparison. This

means that the display name and presence or absence of angle brackets

do not affect matching.

 

See Section 20.10 for the rules for parsing a display name, URI and

URI parameters, and header field parameters.

 

The compact form of the From header field is f.

 

Examples:

 

From: "A. G. Bell" <sip:agb@bell-telephone.com> ;tag=a48s

From: sip:+12125551212@server.phone2net.com;tag=887s

f: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8

 

20.10 Contact

 

A Contact header field value provides a URI whose meaning depends on

the type of request or response it is in.

 

A Contact header field value can contain a display name, a URI with

URI parameters, and header parameters.

 

This document defines the Contact parameters "q" and "expires".

These parameters are only used when the Contact is present in a

REGISTER request or response, or in a 3xx response. Additional

parameters may be defined in other specifications.

 

When the header field value contains a display name, the URI

including all URI parameters is enclosed in "<" and ">". If no "<"

and ">" are present, all parameters after the URI are header

parameters, not URI parameters. The display name can be tokens, or a

quoted string, if a larger character set is desired.

 

Even if the "display-name" is empty, the "name-addr" form MUST be

used if the "addr-spec" contains a comma, semicolon, or question

mark. There may or may not be LWS between the display-name and the

"<".

 

These rules for parsing a display name, URI and URI parameters, and

header parameters also apply for the header fields To and From.

 

The Contact header field has a role similar to the Location header

field in HTTP. However, the HTTP header field only allows one

address, unquoted. Since URIs can contain commas and semicolons

as reserved characters, they can be mistaken for header or

parameter delimiters, respectively.

 

The compact form of the Contact header field is m (for "moved").

 

Examples:

 

Contact: "Mr. Watson" <sip:watson@worcester.bell-telephone.com>

;q=0.7; expires=3600,

"Mr. Watson" <mailto:watson@bell-telephone.com> ;q=0.1

m: <sips:bob@192.0.2.4>;expires=60

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I will look into the call issue, however in terms of the RFC it states the following note the " " around the name, however when we get calls from your switch the " " are missing and thus when a charter other then a-z is used it causes the system to break.

 

I don't understand... The From header that I see is From: "W: Office 1" <sip:9784256666@companya.mydomain.com;user=phone>;tag=648644319, with nice quotes around the "W: Office 1". Maybe they can just take a look at this forum post.

 

If the quotes are missing when it reaches their system, maybe there is a SIP-aware firewall in the middle that takes the quotes out?

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