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Problem with addressbook


Kristan
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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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Hmmm.... Do you have USA dial plan for the domain? If that is so, the PBX interprets the numbers in the "NANPA" style, meaning that numbers must be 10 digits and "international" numbers must start with 011. You can check the entry in the database, look at at the address book in the file system. There is a "display name" and a internal name that is used for comparisons. Maybe that name was converted to USA style.

 

BTW interesting PSTN gateway. Any comments on the gateway?

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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.

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It seems we fixed the issue in the latest version... 2.1 will have it.

 

Is that a confirmed bug then? Any ideas on a release date or if there's a workaround? I've got people shouting at me wanting their address book working again!

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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

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Strange.

 

What raises my eyebrow is that the domain index is "2" - do you have more than one domain? If the PBX sends the call to the wrong domain it is no surprise that the PBX does not find the address book there.

 

Also, make sure that you don't use the NANPA dialplan in the domain. Your numbers start with "0" and that does not look like a U.S. number.

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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?

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  • 2 weeks later...
Still having problems here guys, anyone got any ideas or is there anything further I can try or send in that would help?

 

Thanks,

 

Kristan

 

I have been trying to get the Address Book to work also with no success.

 

I started using the Address Book feature in the individual phones instead.

 

It isn't as vast (200 numbers) as the PBXnSIP's Address Book.

 

Depending on which manufacturer of phones you use, you can put the Address Book names and numbers in the <mac.cfg> for each phone

 

or use the <all.cfg>.

 

I use Aastra and there are 2 config files. The common one to all phones <aastra.cfg> (loads first) and the individual config files for each phone <mac address.cfg> (loads second).

 

I did notice, at one time, that the Name and Number stored in the PBX Address Book would show up on the CDR when ever the number was dialled manually or the Speed Dial Number (*123 example) was used. That kind of indicates that it is in the PBX system doing something, but not attributing the Caller ID Number to a name listed in the Address Book.

 

I hope this may help/

 

Bill H

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Thanks for that Bill, I think we're going to have to go down that route as the address book on PBXnSIP just isn't working, just a pain as it used to work ok.

 

Further to this - I have done further testing using SysInternal's FileMon. Looking at the output, I can see PBXCtrl.exe reading in the various XML files from the adrbook directory, presumably into memory. Calls come in, and do not resolve against this list. I have confirmed the domain index (2) matches the address book entries, but still no joy. Is there any kind of extended debug or something to explain why this isn't working ?

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  • 2 months later...

We're still seeing this not working I'm afraid in even in 2.1 - single entry in the address book and it doesn't match the number on incoming calls at all, as per the rest of the thread above.

 

More strangeness is that if I try to edit the number after adding the entry, it doesn't take the change - if I created it with 07624491xxx, then try deleting the area code 07624, save it, it works - but then if I try putting it back in again, it doesn't save, and just shows me the 491xxx.

 

Any ideas? Sounds like the addressbook really isn't happy!

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