pbxuser911 Posted January 1, 2009 Report Share Posted January 1, 2009 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 Quote Link to comment Share on other sites More sharing options...
pbxuser911 Posted January 1, 2009 Author Report Share Posted January 1, 2009 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 : Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 1, 2009 Report Share Posted January 1, 2009 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"). Quote Link to comment Share on other sites More sharing options...
pbxuser911 Posted January 4, 2009 Author Report Share Posted January 4, 2009 ill let TSG know about it as far as the sound file missing, i renamed the NOISE (comfort tone) sound so it doesnt play we find it not to work best in our set up thanks for the reply and bringing to my attention what you found besides the main issue i posted Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 4, 2009 Report Share Posted January 4, 2009 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"... Quote Link to comment Share on other sites More sharing options...
pbxuser911 Posted January 5, 2009 Author Report Share Posted January 5, 2009 is there a file that sounds better? that sound sounded wierd for our customers, they were asking us what is that wierd sound so we removed it Quote Link to comment Share on other sites More sharing options...
pbxuser911 Posted January 9, 2009 Author Report Share Posted January 9, 2009 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 10, 2009 Report Share Posted January 10, 2009 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? Quote Link to comment Share on other sites More sharing options...
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