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Kristan

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Posts posted by Kristan

  4. Snom subscriptions

    in snom Phones

    Posted

    Hi,

     

    We have a problem with one of our installations. They have a service flag setup (700) which is used as the night mode on the main hunt group. Several of the phones subscribe to this (definied as an "extension" on a function key) so when night mode is on, the light is on. Unfortunately, this seems to stop working for some reason after a week or so and only a restart of the service seems to clear it, it's like the PBX stops sending out NOTIFYs to the phones. They then get annoyed because night mode is on, but they don't know it is, and their clients are all being directed to vm.... Any ideas why this might be? They're on the very latest version of the PBX exe (2.0.9.2059)

     

    They're phones are all on 6.5.10 firmware, we're trying v7 on a couple of our office phones as I know it has better options to show subscriptions, but it doesn't seem stable enough yet to try on clients...

     

    Thanks,

     

    Kristan

  5. Problem with addressbook

    in General Setup

    Posted

    We did have more than one domain, but it's since been deleted. I did try manually changing the domain in the phonebook entries to "1" in the xml files and restarting the PBX, but it didn't seem to make a difference. Definitely not using the NANPA dialplan, the numbers are national format UK ones.

     

    I've tried doing a new install from scratch, and I still get the same behaviour. Incidentally, upgrading to the new version broke incoming calls via one of the SIP registration type trunks we had, but that's a different issue! I've now gone back to the old version.

     

    The only thing I can point this at is the SIP INVITE, the from header now looks like this:

     

    From: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>

     

    whereas previous it was just

     

    From: <sip:07624xxxxxx@10.1.0.7>

     

    Same with the contact. Is the PBX taking this as the name and ignoring the addressbook entry?

  7. Problem with addressbook

    in General Setup

    Posted

    Thanks for that, still no luck though. The database files are now in the correct format, and display number and number both match the incoming number, however I still don't get a match (full logs this time in case it's any help):

     

    INVITE sip:0000@sip.mydomain.com SIP/2.0

    Via: SIP/2.0/UDP 10.1.0.7:5060;rport;branch=z9hG4bK1718576833

    From: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>;tag=66672835

    To: <sip:0000@sip.mydomain.com>

    Call-ID: 405359025@10.1.0.7

    CSeq: 20 INVITE

    Contact: <sip:07624xxxxxx@10.1.0.7:5060>

    Max-Forwards: 70

    User-Agent: Voxtream Parlay VoXip-1-8-1

    Expires: 120

    Remote-Party-ID: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>;party=calling;screen=no;privacy=off

    P-Preferred-Identity: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>

    Supported: replaces

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

    Content-Type: application/sdp

    Content-Length: 237

     

    v=0

    o=userX 11423117 20000001 IN IP4 10.1.0.7

    s=A call

    c=IN IP4 10.1.0.7

    t=0 0

    m=audio 5006 RTP/AVP 0 2 0 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:2 G726-32/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [7] 2007/07/20 08:38:48: UDP: Opening socket on port 50320

    [7] 2007/07/20 08:38:48: UDP: Opening socket on port 50321

    [5] 2007/07/20 08:38:48: Identify trunk (IP address/port and domain match) 3

    [9] 2007/07/20 08:38:48: Resolve destination 967: a udp 10.1.0.7 5060

    [9] 2007/07/20 08:38:48: Resolve destination 967: udp 10.1.0.7 5060

    [7] 2007/07/20 08:38:48: SIP Tx udp:10.1.0.7:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 10.1.0.7:5060;rport=5060;branch=z9hG4bK1718576833

    From: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>;tag=66672835

    To: <sip:0000@sip.mydomain.com>;tag=56f2ac33f7

    Call-ID: 405359025@10.1.0.7

    CSeq: 20 INVITE

    Content-Length: 0

     

     

    [6] 2007/07/20 08:38:48: Sending RTP to 10.1.0.7:5006

    [9] 2007/07/20 08:38:48: Resolve destination 968: a udp 10.1.0.118 2051

    [9] 2007/07/20 08:38:48: Resolve destination 968: udp 10.1.0.118 2051

    [5] 2007/07/20 08:38:48: Trunk ISDN sends call to 250

    [8] 2007/07/20 08:38:48: Play audio_moh/noise.wav

    [7] 2007/07/20 08:38:48: Hunt Group: Moving to next stage

    [9] 2007/07/20 08:38:48: Resolve destination 969: a udp 10.1.0.108 2051

    [9] 2007/07/20 08:38:48: Resolve destination 969: udp 10.1.0.108 2051

    [9] 2007/07/20 08:38:48: Resolve destination 970: a udp 10.1.0.105 2051

    [9] 2007/07/20 08:38:48: Resolve destination 970: udp 10.1.0.105 2051

    [9] 2007/07/20 08:38:48: Resolve destination 971: a udp 10.1.0.111 2051

    [9] 2007/07/20 08:38:48: Resolve destination 971: udp 10.1.0.111 2051

    [9] 2007/07/20 08:38:48: Resolve destination 972: a udp 10.1.0.100 2054

    [9] 2007/07/20 08:38:48: Resolve destination 972: udp 10.1.0.100 2054

    [9] 2007/07/20 08:38:48: Resolve destination 973: a udp 10.1.0.106 2054

    [9] 2007/07/20 08:38:48: Resolve destination 973: udp 10.1.0.106 2054

    [5] 2007/07/20 08:38:48: Using codecs pcmu g726-32 telephone-event

    [9] 2007/07/20 08:38:48: Resolve destination 974: a udp 10.1.0.7 5060

    [9] 2007/07/20 08:38:48: Resolve destination 974: udp 10.1.0.7 5060

    [7] 2007/07/20 08:38:48: SIP Tx udp:10.1.0.7:5060:

    SIP/2.0 183 Ringing

    Via: SIP/2.0/UDP 10.1.0.7:5060;rport=5060;branch=z9hG4bK1718576833

    From: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>;tag=66672835

    To: <sip:0000@sip.mydomain.com>;tag=56f2ac33f7

    Call-ID: 405359025@10.1.0.7

    CSeq: 20 INVITE

    Contact: <sip:0000@10.1.0.40:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.0.9.2059

    Content-Type: application/sdp

    Content-Length: 196

     

    v=0

    o=- 58399 58399 IN IP4 10.1.0.40

    s=-

    c=IN IP4 10.1.0.40

    t=0 0

    m=audio 50320 RTP/AVP 0 2 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:101 telephone-event/8000

    a=sendrecv

     

    The phonebook entry looks like this :

      <?xml version="1.0" encoding="utf-8" ?> 
- <row>
 <display_number>07624xxxxxx</display_number> 
 <domain>2</domain> 
 <first>Kristan</first> 
 <name>Test</name> 
 <number>07624xxxxxx</number> 
 <speed /> 
 <type /> 
 <user /> 
 </row>

     

    And the domain entry looks like this (star codes removed or brevity) :

    <?xml version="1.0" encoding="utf-8" ?> 
- <row>
 <adrbook_match>loose</adrbook_match> 
 <area_code /> 
 <cfn_timeout>10</cfn_timeout> 
 <country_code /> 
 <default_dialplan>1</default_dialplan> 
 <display>localhost</display> 
 <dp>enter</dp> 
 <email_from>pbx@mydomain.com</email_from> 
 <email_pass /> 
 <email_pop3 /> 
 <email_smtp>10.1.0.12</email_smtp> 
 <email_user /> 
 <from_style /> 
 <lang_audio /> 
 <lang_tones /> 
 <lang_web /> 
 <mailbox_escape /> 
 <max_accounts /> 
 <max_calls /> 
 <max_extensions /> 
 <max_mb_duration /> 
 <mb_enter_pin>false</mb_enter_pin> 
 <mb_pinsize>4</mb_pinsize> 
 <mb_prefix>8</mb_prefix> 
 <mb_size>50</mb_size> 
 <mb_timeout>20</mb_timeout> 
 <moh>1</moh> 
 <name>localhost</name> 
 <pickup_policy>false</pickup_policy> 
 <record_annoucement>true</record_annoucement> 
 <soap_extcall /> 
 <star_prefix /> 
 <to_style /> 
 <tz /> 
 <voicemail /> 
 </row>

     

    If I copy the number from the call log, open the addressbook page and do a "find", it works, so the number is definitely correct! Any other ideas or settings I can try?

     

    Thanks for your help,

     

    Kristan

  10. Problem with addressbook

    in General Setup

    Posted

    Nope, pnp dialplan is set to "User must press enter" (we're in the UK). I've had a look at the address book xml files, but they don't have a display name or internal name, just a domain, name and number..

     

    The gateways are good, ISDN based with up to 8 BRI ports. We've used them in several installations and find them to work really well, you can do some quite clever call routing on it, had a built in ISDN monitor (which is very handy when trying to troubleshoot call problems) etc. All in all a well featured box.

  11. Problem with addressbook

    in General Setup

    Posted

    Hi,

     

    We had a 1.5 system which started to failing to match incoming numbers in the address book. We upgraded to 2.0 as we were going to do this anyway, but still have the same problem.

    In this case I have an address book entry setup for 07624xxxxxx where xxxxxx is my number, with my name as the first and last names.

     

    The invite from our ISDN gateway looks like this:

     

    INVITE sip:0000@sip.mydomain.com SIP/2.0

    Via: SIP/2.0/UDP 10.1.0.7:5060;rport;branch=z9hG4bK633500735

    From: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>;tag=1887432543

    To: <sip:0000@sip.mydomain.com>

    Call-ID: 926951278@10.1.0.7

    CSeq: 20 INVITE

    Contact: <sip:07624xxxxxx@10.1.0.7:5060>

    Max-Forwards: 70

    User-Agent: Voxtream Parlay VoXip-1-8-1

    Expires: 120

    Remote-Party-ID: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>;party=calling;screen=no;privacy=off

    P-Preferred-Identity: 07624xxxxxx <sip:07624xxxxxx@10.1.0.7>

    Supported: replaces

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

    Content-Type: application/sdp

    Content-Length: 238

     

    v=0

    o=userX 728681760 20000001 IN IP4 10.1.0.7

    s=A call

    c=IN IP4 10.1.0.7

    t=0 0

    m=audio 5006 RTP/AVP 0 2 0 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:2 G726-32/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    The numbers of the phone book entry and the incoming number match exactly, yet the invites go out to the phones like this:

     

    INVITE sip:201@10.1.0.105:2051;line=kbq2n6ok SIP/2.0

    Via: SIP/2.0/UDP 10.1.0.40:5060;branch=z9hG4bK-892898d350aed058da43c63944ab0a54;rport

    From: "07624xxxxxx" <sip:07624xxxxxx@10.1.0.7>;tag=20996

    To: <sip:0000@sip.mydomain.com>

    Call-ID: 072d8d14@pbx

    CSeq: 13227 INVITE

    Max-Forwards: 70

    Contact: <sip:201@10.1.0.40:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.0.3.1715

    Alert-Info: <http://127.0.0.1/Bellcore-dr3>

    Content-Type: application/sdp

    Content-Length: 260

     

    v=0

    o=- 35429 35429 IN IP4 10.1.0.40

    s=-

    c=IN IP4 10.1.0.40

    t=0 0

    m=audio 52078 RTP/AVP 0 8 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-11

    a=sendrecv

     

    Obviously this means the call shows up on the phone as the number, not the name. I've no idea why it's not working, can anyone suggest anything to try or something I've missed?

     

    Thanks

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