kelvin

Hamlet, Noted. thanks

Hamlet, Thanks for responding, any idea does the evaluation license come with prepay feature license? would like to try out before acquired. regards, Kelvin

Hi Support, The version 4 prepay feature sound good. May i know how to enable this feature? does it require additional license? Regard the credit control does it apply to domain level instead of account level. It mean entire domain sharing same credit amount. Thanks Regards, Kelvin

hi support, we assigned in account number field by putting multiple value as below. i have emailed the log file. thanks +60327215890 60327215890 60327215891 60327215892 60327215893 60327215894 60327215895 60327215896 60327215897 60327215898 60327215899

Support, the DID number unchanged, but pbxnsip sent multiple invite even we just called 1 DID number on. please looked below pbx log. we called DID +60327215891 but pbx sent all DID invite. we configure multiple DID in 1 extension. 2009/09/02 16:03:49: Trunk IN-Cisco-94 (global) sends call to 60327215891 in domain donotdelete.com [9] 2009/09/02 16:03:49: Resolve 653334: url sip:+60327215890@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653334: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653334: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653335: url sip:+60327215899@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653335: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653335: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653336: url sip:+60327215891@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653336: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653336: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653337: url sip:+60327215892@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653337: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653337: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653338: url sip:+60327215893@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653338: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653338: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653339: url sip:+60327215894@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653339: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653339: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653340: url sip:+60327215895@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653340: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653340: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653341: url sip:+60327215896@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653341: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653341: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653342: url sip:+60327215897@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653342: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653342: tcp 202.168.66.202 5060 [9] 2009/09/02 16:03:49: Resolve 653343: url sip:+60327215898@mycybhpv02.my.tria.de;transport=tcp [9] 2009/09/02 16:03:49: Resolve 653343: a tcp mycybhpv02.my.tria.de 5060 [9] 2009/09/02 16:03:49: Resolve 653343: tcp 202.168.66.202 5060 [5] 2009/09/02 16:03:49: Identify trunk (IP address match) 11

hi support, we have map multiple DDIs to a single extension and this particular extension we added manual registration for relay to OCS2007. routing path as below. DDIs --> SIP --> Pbxnsip --> SIP --> OCS 2007 R2 When we call particular DDI, let said call DDI - 1111 but DDI - 2222, DDI - 3333 and so on... all DDI numbers will relay to OCS 2007 R2. OCS 2007 R2 receiving multiple SIP invite (we map 10DDIS, Pbxnsip will sent 10 SIP invite to OCS) please advise how can we configure to sent called DDI sip invite only. example: we call DDI - 2222 and pbxnsip only sent DDI - 2222 SIP invite only. regards, Kelvin

noted. problem resolved. thanks

hi guys, i am try to integrate pbxnsip with OCS 2007 R2. i have no problem making outgoing calls from OCS to pbxnsip but i receiving "unknow RTP version 0" return error in RTP packet from OCS when receiving incoming calls from pbxnsip. the phone will ring for few ring then drop call. i am using pbxnsip version 3.4.0.3201(win32) and OCS 2007 R2 below is the wireshark packet captured. 3 2009-08-25 20:07:35.838839 202.168.66.202 202.79.201.86 RTP Unknown RTP version 0 Real-Time Transport Protocol 00.. .... = Version: Old VAT Version (0) Any advice on this is truly appreciated. Thank you! Regards, Kelvin

hi support, what else we can do to resolve the caller id display issue. thx regards, kelvi

hi Support, The upgrade does not solve the problem, i still receiving caller id = Haniza () when incoming with not caller id and i notice the tel:xxx changed to +xxx after upgrade. what i suppose to do with +xxx? manual change to tel:xxx? thx

hi support, yes, please provide me the 3.1.1 version. thx regards, kelvin

hi Support, our pbxnsip is version 3.0.1.3023 on windows 2000 server. thx regards, kelvin

hi Support, any update on this request? thx regards, kelvin

In version 2.. caller id will display as private no. thx

hi Support, In pbxnsip v3, when incoming call with no caller no, the caller id will change to weird name either domain name or extension name. please refer to follow log. thx INVITE sip:65014582@202.79.201.86:5080 SIP/2.0 Via: SIP/2.0/UDP 202.79.201.83:5060 From: <sip:202.79.201.83>;tag=170CFF74-1CC7 To: <sip:65014582@202.79.201.86> Date: Wed, 19 Nov 2008 05:02:35 gmt Call-ID: 1D0F310D-B52E11DD-AFADC2C9-4499683E@202.79.201.83 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 487454909-3039695325-2947203785-1150904382 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:202.79.201.83>;party=calling;screen=no;privacy=off Timestamp: 1227070955 Contact: <sip:202.79.201.83:5060> change to: INVITE sip:215@192.168.1.246 SIP/2.0 Via: SIP/2.0/UDP 202.79.201.86:5080;branch=z9hG4bK-596d1b86ce9eb2f2eea733ce72ec8892;rport From: "Haniza" <sip:pbx.pgcomms.com;user=phone>;tag=4607 To: "Kelvin Tee" <sip:215@pbx.pgcomms.com> Call-ID: 894b27a1@pbx CSeq: 27463 INVITE Max-Forwards: 70 Contact: <sip:215@202.79.201.86:5080;transport=udp> my ip phone will display "haniza" as caller.

by default the ANI = caller extension no right? i want to display Caller extension so that we can identify the calls. by the way what is this message mean? "[5] 2008/09/05 14:06:22: Received loopback request without tag"

hi support, i using above method, and my snom return Authentication required with busy tone when try call interbranch extension. below is the log. thx SIP/2.0 183 Ringing Via: SIP/2.0/TLS 192.168.1.176:3101;branch=z9hG4bK-2v9nief449ot;rport=3101 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=dkm152o0i5 To: <sip:365215@pbx.pgcomms.com.my;user=phone>;tag=539287bafe Call-ID: 3c3a1fd5d921-1d8euqo6pq07 CSeq: 1 INVITE Contact: <sip:102@192.168.1.10:5081;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pgcomms-PBX/3.0.0.2998 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 433 v=0 o=- 18443 18443 IN IP4 192.168.1.10 s=- c=IN IP4 192.168.1.10 t=0 0 m=audio 54216 RTP/AVP 18 3 2 0 8 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iIE4YWtoEmL5VNDncRdIeA2SKR2GKZly23Qc+k83 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080: INVITE sip:65215@127.0.0.1:5080;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone> Call-ID: 7b24e046@pbx CSeq: 18780 INVITE Max-Forwards: 70 Contact: <sip:102@127.0.0.1:5080;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pgcomms-PBX/3.0.0.2998 P-Asserted-Identity: "Normala" <sip:102@pbx.pgcomms.com.my> Content-Type: application/sdp Content-Length: 331 v=0 o=- 20980 20980 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 58324 RTP/AVP 18 3 2 0 8 9 101 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [5] 2008/09/04 17:26:40: Received loopback request without tag [7] 2008/09/04 17:26:40: UDP: Opening socket on port 57660 [7] 2008/09/04 17:26:40: UDP: Opening socket on port 57661 [5] 2008/09/04 17:26:40: Identify trunk (IP address/port match) 23 [9] 2008/09/04 17:26:40: Resolve 17492: aaaa udp 127.0.0.1 5080 [9] 2008/09/04 17:26:40: Resolve 17492: a udp 127.0.0.1 5080 [9] 2008/09/04 17:26:40: Resolve 17492: udp 127.0.0.1 5080 [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 INVITE Content-Length: 0 [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 INVITE Content-Length: 0 [9] 2008/09/04 17:26:40: Resolve 17493: aaaa udp 127.0.0.1 5080 [9] 2008/09/04 17:26:40: Resolve 17493: a udp 127.0.0.1 5080 [9] 2008/09/04 17:26:40: Resolve 17493: udp 127.0.0.1 5080 [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 INVITE User-Agent: pgcomms-PBX/3.0.0.2998 WWW-Authenticate: Digest realm="pbx.pgcomms.com.my",nonce="4cf3b7a72fbee5868e7e410c0e4ee2e4",domain="sip:65215@127.0.0.1:5080;user=phone",algorithm=MD5 Content-Length: 0 [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080 From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 INVITE User-Agent: pgcomms-PBX/3.0.0.2998 WWW-Authenticate: Digest realm="pbx.pgcomms.com.my",nonce="4cf3b7a72fbee5868e7e410c0e4ee2e4",domain="sip:65215@127.0.0.1:5080;user=phone",algorithm=MD5 Content-Length: 0 [7] 2008/09/04 17:26:40: Call 7b24e046@pbx#15046: Clear last INVITE [9] 2008/09/04 17:26:40: Resolve 17494: url sip:127.0.0.1:5080 [9] 2008/09/04 17:26:40: Resolve 17494: udp 127.0.0.1 5080 [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080: ACK sip:65215@127.0.0.1:5080;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046 To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb Call-ID: 7b24e046@pbx CSeq: 18780 ACK Max-Forwards: 70 Contact: <sip:102@127.0.0.1:5080;transport=udp> P-Asserted-Identity: "Normala" <sip:102@pbx.pgcomms.com.my> Content-Length: 0

hi support, any update on above request? thx regards, kelvin

for this case, what alternate way i can use to route inter branch calls. thx regards, kelvin

hi Support, i am make use of tel alias as global alias for inter branch call, so that branch extension able contact each others using this global alias but after upgrade to version 3.0.0.2998. It stop working, everytime call global alias pbx return not found message. kindly advice the solution. thx regards, kelvin

ok. thanks regards, kelvin

ok. thx regards, kelvin

hi support, may i know pbxnsip support dtmf sip info relay? if yes, how to change it on trunk? we experience dtmf issue on quintum DX2030 with dtmf sip info relay. when call connect to IVR system, dtmf does not recognize by ivr system. if change to h245 outband then no issue but due to environment setup we cant use h245 outband. thx regards, kelvin

hi Support, what will be the code for Codec G723? in wiki only tell codecs ulaw (0), alaw(8), G.722 (9), G.726 (2) or GSM 6.10 FullRate (3). thx regards, kelvin

noted. thx