intercom feature

reco · 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

Vodia PBX · February 18, 2009

what am i missing?

Hmm. Works here... Also on version 7.3.14.

Do you see "No permission for intercom to ..." in the log (level 5)?

nate · February 20, 2009

I'm having a hard time getting the intercom feature to work on snom handsets. for example, extension 600, i dial *90600 it should auto pick up that phone. instead, it just rings the extension.

Any one have any input?

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...

Vodia PBX · February 21, 2009

settings from the target extension:

intercom_enabled!: on

intercom_connect_type!: intercom_connect_type_handsfree

The settings on the phone should not make the big difference. Can you get the SIP INVITE from the phones web trace log?

Bill H · February 21, 2009

I discovered this too on the CS-410 with the 3144 version.

Dial *90# and it will ask you to "Please enter the extension number"

Then it will work like an intercom.

Why has this been changed?

mattlandis · February 21, 2009

I discovered this too on the CS-410 with the 3144 version.

Dial *90# and it will ask you to "Please enter the extension number"

Then it will work like an intercom.

Why has this been changed?

For us when I dial *90XXX (xxx=ext.) is just rings the phone. (or do it the way you explianed)

This WORKED for us in the previous version. (we are running windows version)

we are using provisioned snom360s.

tx

matt

reco · February 24, 2009

glad you guys see the same problems.

looking forward for a fix

reco

pbx support · February 25, 2009

glad you guys see the same problems.
looking forward for a fix

reco

I tested it here and when the "Intercom to the following extensions:" is empty, it rings the phone and when I set it to "*" it intercoms (snom 320 & 360)

mattlandis · February 25, 2009

i have * in the "Intercom to the following extensions:" for the extension initiating the intercom and the phone getting the intercom just rings...

snom 360

there remains a problem.

tx

matt

pbx support · February 25, 2009

i have * in the "Intercom to the following extensions:" for the extension initiating the intercom and the phone getting the intercom just rings...
snom 360

there remains a problem.

tx

matt

What OS? I tested it on Windows and CS410. Both seem to work ok. BTW, what is the called UA? (snom, polycom, cisco?)

pbx support · February 25, 2009

What OS? I tested it on Windows and CS410. Both seem to work ok. BTW, what is the called UA? (snom, polycom, cisco?)

Checkout http://wiki.snom.com/Settings/answer_after_policy. Also, http://wiki.snom.com/Settings/intercom_enabled

mattlandis · February 25, 2009

What OS? I tested it on Windows and CS410. Both seem to work ok. BTW, what is the called UA? (snom, polycom, cisco?)

-all phones involved are snom 360

-this worked before upgrading to the latest version

-version=windows.

tx

matt

nate · March 3, 2009

I am still having this problem... so does anyone have a fix? on the phones, i have ' answer after policy' always.

enable intercom on

i don't know what these settings were prior to latest firmware update.

this was a very nice feature. Can someone please help?

Vodia PBX · March 3, 2009

I am still having this problem... so does anyone have a fix? on the phones, i have ' answer after policy' always.

enable intercom on

i don't know what these settings were prior to latest firmware update.

this was a very nice feature. Can someone please help?

I would factory-reset the phone and provision it automatically. There is no need to change the configuration; the PBX will signal the intercom feature in the INVITE.

nate · March 3, 2009

Ok, so i reset the phone back to factory defaults. it still does the same thing. the phone won't auto anwer, and now, i can't login to the web interface on the phone.

it prompts me for a login, and my extension doesn't work, neither does admin and no password.

reco · 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

Vodia PBX · March 3, 2009

what do you need?

The INVITE from the phone's log (the one that receives the INVITE) would be interesting!

reco · 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

pbx support · 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):
Interesting.. the invite sent by the PBX to the called party should have either "Call-Info" header or "Alert-Info" header.

Call-Info: <sip:41@pbx.company.com>;answer-after=0
Or
Alert-Info: Auto Answer

I just tested this on pbxnsip-PBX/3.2.0.3143 (Mac OS) and it worked fine. PBX and both phones are on the same network, snom-320 using version 7.1.35 (and 7.3.14)

reco · 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?

pbx support · 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?

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.

Vodia PBX · March 5, 2009

Check the status of the setting called "answer_after_policy" on the phone. It should be "idle", not "off".

nate · March 9, 2009

Check the status of the setting called "answer_after_policy" on the phone. It should be "idle", not "off".

Could someone post a ini file of a working snom phone. i don't think it matters what phone. i'm thinking this happened after latest firmware update.

nate · March 9, 2009

Could someone post a ini file of a working snom phone. i don't think it matters what phone. i'm thinking this happened after latest firmware update.

I just tried, *90

it asks "please enter the extension number"

i dial "ext"

it says, you are not allowed to place this call.

intercom feature

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