reco

March 11, 2009

callcentric only accepts 11digit numbers so my dial plan looks like this:

prio trunk pattern replacement

98 callcentric 411

99 Try Loopback xx

200 callcentric 011* 011*

202 callcentric xxxxxxxxxx 1*

203 callcentric xxxxxxxxxxx *

version 3.2.0.3143 works as expected

dial: 222 333 4444 the trunk gets 1 222 333 4444

after upgrading to 3.3.0.3156

dial: 222 333 4444 the trunk gets 222 333 4444

i know this is already another topic... but... sorry

thanx

March 11, 2009

ATTENTION!!!

after upgrade outgoing calls do not work anymore...

seems like something with the dialplans did get messup....

i am getting a message form callcentric that the number cannot be dialed as entered.

back to 3.2.0.3143 and trunks are working again.

9] 20090311120259: SIP Rx tls:PBXIP:2092:

INVITE sip:2223334444@domain.com;user=phone SIP/2.0

Via: SIP/2.0/TLS PBXIP:2092;branch=z9hG4bK-hzyu0mnjw7ur;rport

From: "Christof Haemmerle" <sip:33@domain.com>;tag=uz3ub5iepq

To: <sip:2223334444@domain.com;user=phone>

Call-ID: 3c267f040947-s0n92pf3dhh4

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:33@PBXIP:2092;transport=tls;line=3lalk32m>;reg-id=1

P-Key-Flags: keys="3"

User-Agent: snom320/7.3.14

Accept: application/sdp

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

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 448

v=0

o=root 1261499776 1261499776 IN IP4 PBXIP

s=call

c=IN IP4 PBXIP

t=0 0

m=audio 49258 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XqGHT9IY82/nH54UAbnMsBdqMEe7u+hlebOorv2d

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[8] 20090311120259: Packet authenticated by transport layer

[9] 20090311120259: UDP: Opening socket on 0.0.0.0:51702

[9] 20090311120259: UDP: Opening socket on 0.0.0.0:51703

[9] 20090311120259: UDP: Opening socket on [::]:51702

[9] 20090311120259: UDP: Opening socket on [::]:51703

[1] 20090311120259: UDP: TOS could not be set

[1] 20090311120259: Last message repeated 2 times

[8] 20090311120259: Could not find a trunk (3 trunks)

[9] 20090311120259: Using outbound proxy sip:PBXIP:2092;transport=tls because of flow-label

[9] 20090311120259: Last message repeated 2 times

[9] 20090311120259: SIP Tx tls:PBXIP:2092:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS PBXIP:2092;branch=z9hG4bK-hzyu0mnjw7ur;rport=2092

From: "Christof Haemmerle" <sip:33@domain.com>;tag=uz3ub5iepq

To: <sip:2223334444@domain.com;user=phone>;tag=1aa0ae9577

Call-ID: 3c267f040947-s0n92pf3dhh4

CSeq: 1 INVITE

Content-Length: 0

[7] 20090311120259: Set packet length to 20

[6] 20090311120259: Sending RTP for 3c267f040947-s0n92pf3dhh4#1aa0ae9577 to PBXIP:49258

[9] 20090311120259: Dialplan: Evaluating !^(411)@.*!sip:\1@\r;user=phone!i against 2223334444@domain.com

[9] 20090311120259: Dialplan: Evaluating !^([0-9]{10})@.*!sip:1\1@\r;user=phone!i against 2223334444@domain.com

[5] 20090311120259: Dialplan voip: Match 2223334444@domain.com to <sip:12223334444@callcentric.com;user=phone> on trunk callcentric-domain

[9] 20090311120259: UDP: Opening socket on 0.0.0.0:54758

[9] 20090311120259: UDP: Opening socket on 0.0.0.0:54759

[8] 20090311120259: Play audio_moh/noise.wav

[9] 20090311120259: UDP: Opening socket on [::]:54758

[9] 20090311120259: UDP: Opening socket on [::]:54759

[1] 2009

March 11, 2009

i want to set the following parameters which are not in the PnP settings in the admin interface.

holding_reminder!: off

user_ringer1!: Ringer7

ring_sound!: Ringer7

alert_internal_ring_sound!: Ringer7

alert_external_ring_sound!: Ringer7

i looked at this post but cannot figure out how to get this work

http://forum.pbxnsip.com/index.php?showtop...8&hl=ringer

any help? thanx

March 11, 2009

Updated to 3156. And just works again.

with7.1.x and 7.3.x firmware

thanx!

Reco

Since it is working in our lab with the same versions, it is hard to declare it as a bug yet. We are looking into other aspect that could impact the intercom behavior now.
BTW, there is a new version for both windows and mac - http://www.pbxnsip.com/protect/pbxctrl-3.3.0.3156.exe, http://www.pbxnsip.com/protect/pbx-darwin9.0-3.3.0.3156.zip

If you want to give the Mac version a spin, feel free(make a back up of the old data). Intercom works ok on both the versions here.

March 4, 2009

hi there,

pbx running on mac os x 10.5.6

phone and pbx on the same subnet

phone firmware version 7.3.14

is this a bug?

March 4, 2009

here the phone log:

Received from tls:10.0.24.2:5061 at 3/3/2009 23:05:02:428 (956 bytes):

INVITE sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364
To: "Tina Preschitz" <sip:34@nex9.com>
Call-ID: 31be71b6@pbx
CSeq: 24165 INVITE
Max-Forwards: 70
Contact: <sip:34@10.0.24.2:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.2.0.3143
Content-Type: application/sdp
Content-Length: 354

v=0
o=- 1469141092 1469141092 IN IP4 10.0.24.2
s=-
c=IN IP4 10.0.24.2
t=0 0
m=audio 62894 RTP/AVP 0 8 3 2 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:JnUQo6gyigtj8YS0iNRIXM89hzVkINALVaPk6rUH
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
Sent to tls:10.0.24.2:5061 at 3/3/2009 23:05:02:472 (525 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport=5061
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364
To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr
Call-ID: 31be71b6@pbx
CSeq: 24165 INVITE
Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-id=1
Require: 100rel
RSeq: 1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

Received from tls:10.0.24.2:5061 at 3/3/2009 23:05:02:613 (420 bytes):

PRACK sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-3bb0c2280d9f577775208f94e989a13e;rport
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364
To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr
Call-ID: 31be71b6@pbx
CSeq: 24166 PRACK
Max-Forwards: 70
Contact: <sip:34@10.0.24.2:5061;transport=tls>
RAck: 1 24165 INVITE
Content-Length: 0

Sent to tls:10.0.24.2:5061 at 3/3/2009 23:05:02:624 (359 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-3bb0c2280d9f577775208f94e989a13e;rport=5061
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364
To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr
Call-ID: 31be71b6@pbx
CSeq: 24166 PRACK
Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-id=1
Content-Length: 0

Received from tls:10.0.24.2:5061 at 3/3/2009 23:05:04:091 (337 bytes):

CANCEL sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364
To: "Tina Preschitz" <sip:34@nex9.com>
Call-ID: 31be71b6@pbx
CSeq: 24165 CANCEL
Max-Forwards: 70
Content-Length: 0

Sent to tls:10.0.24.2:5061 at 3/3/2009 23:05:04:099 (288 bytes):

SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport=5061
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364
To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr
Call-ID: 31be71b6@pbx
CSeq: 24165 CANCEL
Content-Length: 0

Sent to tls:10.0.24.2:5061 at 3/3/2009 23:05:04:111 (376 bytes):

SIP/2.0 487 Request Terminated
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport=5061
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364
To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr
Call-ID: 31be71b6@pbx
CSeq: 24165 INVITE
Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-id=1
Content-Length: 0

Received from tls:10.0.24.2:5061 at 3/3/2009 23:05:04:219 (394 bytes):

ACK sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364
To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr
Call-ID: 31be71b6@pbx
CSeq: 24165 ACK
Max-Forwards: 70
Contact: <sip:34@10.0.24.2:5061;transport=tls>
Content-Length: 0

here the pbxnsip log

[9] 20090303225858: SIP Rx tls:10.0.24.24:2056:
INVITE sip:*9034@nex9.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-vwhdetk7intn;rport
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj
To: <sip:*9034@nex9.com;user=phone>
Call-ID: 3c2712760257-tf8ewlridkyh
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:33@10.0.24.24:2056;transport=tls;line=4zhgbysz>;reg-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/7.3.14
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 448

v=0
o=root 1240381312 1240381312 IN IP4 10.0.24.24
s=call
c=IN IP4 10.0.24.24
t=0 0
m=audio 63900 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:9QkeySQSHNoHQxwWRkkQO3F9u1Zs4/lIEMgvFsFF
a=rtpmap:9 g722/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[8] 20090303225858: Packet authenticated by transport layer
[9] 20090303225858: UDP: Opening socket on port 50530
[9] 20090303225858: UDP: Opening socket on port 50531
[9] 20090303225858: UDPv6: Opening socket on port 50530
[9] 20090303225858: UDPv6: Opening socket on port 50531
[1] 20090303225858: UDP: TOS could not be set
[1] 20090303225858: Last message repeated 2 times
[8] 20090303225858: Could not find a trunk (3 trunks)
[9] 20090303225858: Using outbound proxy sip:10.0.24.24:2056;transport=tls because of flow-label
[9] 20090303225858: Last message repeated 2 times
[9] 20090303225858: SIP Tx tls:10.0.24.24:2056:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-vwhdetk7intn;rport=2056
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj
To: <sip:*9034@nex9.com;user=phone>;tag=d6ea99ba48
Call-ID: 3c2712760257-tf8ewlridkyh
CSeq: 1 INVITE
Content-Length: 0


[7] 20090303225858: Set packet length to 20
[6] 20090303225858: Sending RTP for 3c2712760257-tf8ewlridkyh#d6ea99ba48 to 10.0.24.24:63900
[8] 20090303225858: Play audio_moh/noise.wav
[9] 20090303225858: UDP: Opening socket on port 54934
[9] 20090303225858: UDP: Opening socket on port 54935
[9] 20090303225858: UDPv6: Opening socket on port 54934
[9] 20090303225858: UDPv6: Opening socket on port 54935
[1] 20090303225858: UDP: TOS could not be set
[1] 20090303225858: Last message repeated 2 times
[9] 20090303225858: Using outbound proxy sip:10.0.24.23:2056;transport=tls because of flow-label
[9] 20090303225858: SIP Tx tls:10.0.24.23:2056:
INVITE sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-d7be59d02d9fb308cef2e19b334d015f;rport
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=311957039
To: "Tina Preschitz" <sip:34@nex9.com>
Call-ID: cb1fa236@pbx
CSeq: 28978 INVITE
Max-Forwards: 70
Contact: <sip:34@10.0.24.2:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.2.0.3143
Content-Type: application/sdp
Content-Length: 352

v=0
o=- 376158563 376158563 IN IP4 10.0.24.2
s=-
c=IN IP4 10.0.24.2
t=0 0
m=audio 54934 RTP/AVP 0 8 3 2 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FMw/tCREc0JA5/5jLpkctEANAGxJX6bVwcLGQ1Cz
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

[7] 20090303225858: Set packet length to 20
[9] 20090303225858: SIP Rx tls:10.0.24.23:2056:
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-d7be59d02d9fb308cef2e19b334d015f;rport=5061
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=311957039
To: "Tina Preschitz" <sip:34@nex9.com>;tag=3tezmb6ixm
Call-ID: cb1fa236@pbx
CSeq: 28978 INVITE
Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-id=1
Require: 100rel
RSeq: 1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


[9] 20090303225858: SIP Tx tls:10.0.24.23:2056:
PRACK sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-ae98c15052991447af66ffbb02e6b539;rport
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=311957039
To: "Tina Preschitz" <sip:34@nex9.com>;tag=3tezmb6ixm
Call-ID: cb1fa236@pbx
CSeq: 28979 PRACK
Max-Forwards: 70
Contact: <sip:34@10.0.24.2:5061;transport=tls>
RAck: 1 28978 INVITE
Content-Length: 0


[8] 20090303225858: Play audio_en/ringback.wav
[9] 20090303225858: SIP Tx tls:10.0.24.24:2056:
SIP/2.0 183 Ringing
Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-vwhdetk7intn;rport=2056
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj
To: <sip:*9034@nex9.com;user=phone>;tag=d6ea99ba48
Call-ID: 3c2712760257-tf8ewlridkyh
CSeq: 1 INVITE
Contact: <sip:33@10.0.24.2:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.2.0.3143
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 366

v=0
o=- 1421313999 1421313999 IN IP4 10.0.24.2
s=-
c=IN IP4 10.0.24.2
t=0 0
m=audio 50530 RTP/AVP 0 8 3 2 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fLIdE/rUu5cghG97S14nBTgil4tkNU66VstPOaVQ
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[9] 20090303225858: SIP Rx tls:10.0.24.24:2056:
PRACK sip:33@10.0.24.2:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-ieriregozcix;rport
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj
To: <sip:*9034@nex9.com;user=phone>;tag=d6ea99ba48
Call-ID: 3c2712760257-tf8ewlridkyh
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:33@10.0.24.24:2056;transport=tls;line=4zhgbysz>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons
Content-Length: 0


[8] 20090303225858: Packet authenticated by transport layer
[9] 20090303225858: SIP Tx tls:10.0.24.24:2056:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-ieriregozcix;rport=2056
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj
To: <sip:*9034@nex9.com;user=phone>;tag=d6ea99ba48
Call-ID: 3c2712760257-tf8ewlridkyh
CSeq: 2 PRACK
Contact: <sip:33@10.0.24.2:5061;transport=tls>
User-Agent: pbxnsip-PBX/3.2.0.3143
Content-Length: 0


[7] 20090303225859: Receiving DTMF on codec 101
[9] 20090303225859: SIP Rx tls:10.0.24.23:2056:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-ae98c15052991447af66ffbb02e6b539;rport=5061
From: "Christof Haemmerle" <sip:33@nex9.com>;tag=311957039
To: "Tina Preschitz" <sip:34@nex9.com>;tag=3tezmb6ixm
Call-ID: cb1fa236@pbx
CSeq: 28979 PRACK
Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-id=1
Content-Length: 0


[7] 20090303225859: Call cb1fa236@pbx#311957039: Clear last request
[9] 20090303225900: SIP Rx udp:10.0.24.20:5060:
SUBSCRIBE sip:10.0.24.2 SIP/2.0
Via: SIP/2.0/UDP 10.0.24.20;branch=z9hG4bK33hx4q971oe1oe
Max-Forwards: 70
From: <sip:32@buero-newyork.com>;tag=niwan
To: <sip:32@buero-newyork.com>;tag=79e2a83e4f
Call-ID: 3f6gv27wo85af@buero-newyork.com
CSeq: 8616 SUBSCRIBE
Contact: <sip:32@10.0.24.20>
Accept: application/simple-message-summary
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK
Allow-Events: dialog,message-summary
Authorization: Digest username="32", realm="buero-newyork.com", nonce="bcb648cf721f1c55464482e81d2fabce", uri="sip:10.0.24.2", response="1e01d5b0a7a737753a7555c74fd01bb2", algorithm=MD5
Event: message-summary
Expires: 28
Supported: replaces
User-Agent: snom-m3-SIP/01.22 (MAC=0004132A0F44; HW=1)
Content-Type: text/plain
Content-Length: 0


[9] 20090303225900: Resolve 1217647: aaaa udp 10.0.24.20 5060
[9] 20090303225900: Resolve 1217647: a udp 10.0.24.20 5060
[9] 20090303225900: Resolve 1217647: udp 10.0.24.20 5060
[9] 20090303225900: SIP Tx udp:10.0.24.20:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.0.24.20;branch=z9hG4bK33hx4q971oe1oe
From: <sip:32@buero-newyork.com>;tag=niwan
To: <sip:32@buero-newyork.com>;tag=79e2a83e4f
Call-ID: 3f6gv27wo85af@buero-newyork.com
CSeq: 8616 SUBSCRIBE
Contact: <sip:10.0.24.2:5060;transport=udp>
Expires: 32
Content-Length: 0

March 3, 2009

hi pbxnsip,

also just reset both phones again

dialed *9034

phone was ringing... i think we need to digg deeper.

what do you need?

thanx reco

February 24, 2009

glad you guys see the same problems.

looking forward for a fix

reco

February 20, 2009

all log levels to 9 and every option turned on.

nothing with ``intercom`` or ``permission`` in the logfiles.

again the feature code for intercom is: *90

the target extension: 34

the origin extension: 33

on EXT 33 I dial: *9034

-> ext 34 just rings.

settings from the target extension:

intercom_enabled!: on

intercom_connect_type!: intercom_connect_type_handsfree

no clue what's happening here...

February 18, 2009

i had a working intercom setup.

after the upgrade to 3.2 the phones only ring but do not auto answer

domain intercom feature code *90

extension i try to intercom 34

extension intercom permission on both phones: *

calling *9034 makes the phone only ring

3.2.0.3143 (Darwin)

snom 320 7.3.14

phone setting

intercom_enabled!: on

intercom_connect_type!: intercom_connect_type_handsfree

what am i missing?

thanx

February 18, 2009

i have the same problem here.

i was never worried about the co lines but also intercom does not work anymore

will write a new post since its not related to this.

thanx

January 29, 2009

this is wired,

is there a way to tell the pbx to flush that cache. this just started to happen since callcentric did some work on their network.

no changes on the pbx.

after a restart it works again for some time.

christof

January 20, 2009

i have the crash report attached.

any idea what happened?

i had most of the log stuff turned off no real info in the log.

attached there is a crash report.

pbxctrl_darwin9.0_2009_01_20_170634_voice.crash.zip

Pbxnsip_error.log.zip

December 2, 2008

i am on version 3.0.1 and this does not work. unfortunately.

any idea how this needs to be done?

You need to declare each extension as a tel:alias.

So you would create in the alias column of extension 200 the following "tel:200".

This is used as a Global Scope alias and therefore allows you to cross-domains

October 21, 2008

hi hki

zip is the standard compression format build into the mac os x server. its pretty standard since this feature was introduced in the os 2-3 years back.

christof

September 30, 2008

Yea, we found that one already. The workaround for time being is to use another email server. Not sure if we have a MacOS build ready right now.

cool thanx looking forward for the a new release

September 29, 2008

We have added STARTTLS, which is probably causing the problem. It works with google, but that seems to be only half the battle. We have another case where it does not work, but maybe you can also send me (PM) a test account for that server and we can test it from the lab.

hi there i am using google apps mail service to send all mails with pbxnsip.

most of the part this works pretty well. but sometimes its stuck in a loop where it repots in the logs that the connection got refused but the mail was sent. so i end up getting hundres of messages ;(

platform: mac os x 10.5.5

Version: 3.0.0.2998 (Darwin)

smtp server config in domain: smtp.gmail.com:587

log:

[8] 20080929140103: SMTP: Connect to 74.125.93.109:587

[8] 20080929140103: SMTP: Received 220 mx.google.com ESMTP 6sm218032qwk.1

[8] 20080929140103: SMTP: Received 250-mx.google.com at your service, [pbxip]

250-SIZE 35651584

250-8BITMIME

250-STARTTLS

250 ENHANCEDSTATUSCODES

[8] 20080929140103: SMTP: Received 220 2.0.0 Ready to start TLS

[8] 20080929140103: SMTP: Received

[8] 20080929140103: Last message repeated 2 times

[8] 20080929140103: SMTP: Received 334 VXNlcm5hbWU6

[8] 20080929140103: SMTP: Received 334 UGFzc3dvcmQ6

[8] 20080929140104: SMTP: Received 235 2.7.0 Accepted

[8] 20080929140104: SMTP: Received 250 2.1.0 OK 6sm218032qwk.1

[8] 20080929140104: SMTP: Received 250 2.1.5 OK 6sm218032qwk.1

[8] 20080929140104: SMTP: Received 354 Go ahead 6sm218032qwk.1

[8] 20080929140105: SMTP: Received 250 2.0.0 OK 1222711265 6sm218032qwk.1

[5] 20080929140105: SMTP: Connection refused on 74.125.93.109:587

September 10, 2008

thanx for the xml file but i still could not figure out how to change this for snom phones.

<vendor ua="Snom.*" type="alert-info">Custom 1</vendor>

</tone>

could the value of the alert-info also be an url to a wav or mp3 file?

thanx christof

September 10, 2008

it would be nice to be able to have the option to upload professionally recorded voicemail greetings like we can with an AA , hunting and pecking through the file system to replace the .wav files is a pain for those who want to have them pre-recorded ..

y

yes i second that plus and option to record more than one greeting and choose one ad active greeting through web interface and the voice menu.

i actually thought this was included in version 3 but i cannot find it.

christof

September 2, 2008

did somebody try this with the snom 320? could not find a template and does not seem so easy to me to configure.

i would love just to choose form the ringers in the phone "ringer 7"

thanx

September 1, 2008

hi pradeep,

just installed pbxnsip on mac. works good so far.

any idea when you will be using my installer scheme with launchd and install path /Library/Pbsnsip?

thanx

christof

August 31, 2008

if my pbxnsip looses the internet the trunks do not reregister automatically.

trunk page displays:

500 Address Resolution Failed (Registration failed, retry after 60 seconds)

but it does not reregister. also clicking `reregister` does not do the trick.

a reboot of pbxnsip does solve this.

the internet connection is up while clicking `reregister`

Version: 3.0.0.2993 (Linux)

christof

August 20, 2008

nice, if this is kind of stable i would switch from 2 mac mini's running debian to maosx right away.

christof

We are working on the installer

August 16, 2008

i am afraid i need a link for the debian version

August 15, 2008

he guys amazing....

amazing guys, finally...

requests:

- lets include the address book dialer posted on this forum.

creating install package:

1. download apple developer tools

2. in /Developer/Applications/Utilities/Packagemaker.app is your friend.

no clue about the time issue.

christof

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intercom feature

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