RobertO
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Posts posted by RobertO
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That translates into *601 meaning pick a call up and 5775 being the call-identifier.
The PBX automatically tells the phone what code to use for the next pickup. So you don't have to configure anything.
It is possible to see the number from the person who is calling and not *601<call-identifier>?
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We have groups of people who are able to pickup calls within their pickup-group. This is done with the buton feature from snom/pbxnsip. When someone pickup a call from another an increasing originator-numer is shown.
1. pickup "*6015775"
next "*6015781"
next "*6015784"
....
On the called phone the right originator is shown. Do I have something to configure?
Regards
Robert
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Is it possible to redirect all incomming external calls for a few accounts to the switchboard. But internal direct calling to this accounts should be possible.
In other words an DND with redirection only for external calls. It would be nice if this could be controlled with a button on the phone (Snom 360). Thanks
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We are using Snom 360 phones with Firmware 7.1.30 and PBXnSIP (2.1.5.2357 (Win32)). The Snom Phones are factory default and have only the PNP settings applied. So I think nothing very exotic.
As I know snom phones support RFC 4916. But I am not shure!
I have no more Ideas what to do? Please help!!!!
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Here is the SIP Protocol from Phone ( C ) There is an INFO message but the Caller ID from Phone ( A ) ist never seen...
Phone ( A ) External
Phone ( B ) IP 192.168.232.117
Phone ( C ) IP 192.168.232.90
PBX IP 192.168.232.10
An other phenomen is that the button of the called person is blinking.
Any Idea????
<---------- Snip -----
Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:53:370 (1038 bytes):
INVITE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-4ab17271000e1823454fdfbdecd95195;rport
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>
Call-ID: 02622e7d@pbx
CSeq: 16185 INVITE
Max-Forwards: 70
Contact: <sip:4406@192.168.232.10:5061;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/2.1.5.2357
Alert-Info: <http://127.0.0.1/Bellcore-dr2>
Content-Type: application/sdp
Content-Length: 378
v=0
o=- 40095 40095 IN IP4 192.168.232.10
s=-
c=IN IP4 192.168.232.10
t=0 0
m=audio 58138 RTP/AVP 8 0 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:MHLtx19IYi7nF6NjlHBXJIZRbYLG6MgJ5LnW5sbG
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
--------------------------------------------------------------------------------
Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:53:420 (532 bytes):
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-4ab17271000e1823454fdfbdecd95195;rport=5061
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16185 INVITE
Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-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:192.168.232.10:5061 at 7/2/2008 20:09:53:679 (500 bytes):
MESSAGE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-1f3c1d62036b0d14ee3486e175f7cbc9;rport
From: "ROE " <sip:4406@sip.domain.de>;tag=65162
To: "ROE " <sip:4406@sip.domain.de>
Call-ID: iu5xtytk@pbx
CSeq: 12216 MESSAGE
Max-Forwards: 70
Contact: <sip:192.168.232.10:5061;transport=tls>
Subject: buttons
Content-Type: application/x-buttons
Content-Length: 53
k=40
c=pickup
x=ext
i=4451
n=*60114
a=invite
--------------------------------------------------------------------------------
Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:53:810 (270 bytes):
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-1f3c1d62036b0d14ee3486e175f7cbc9;rport=5061
From: "ROE " <sip:4406@sip.domain.de>;tag=65162
To: "ROE " <sip:4406@sip.domain.de>
Call-ID: iu5xtytk@pbx
CSeq: 12216 MESSAGE
Content-Length: 0
--------------------------------------------------------------------------------
Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:53:817 (433 bytes):
PRACK sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-12d35d0e49ec9c6110c9ffcb59878593;rport
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16186 PRACK
Max-Forwards: 70
Contact: <sip:4406@192.168.232.10:5061;transport=tls>
RAck: 1 16185 INVITE
Content-Length: 0
--------------------------------------------------------------------------------
Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:53:827 (366 bytes):
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-12d35d0e49ec9c6110c9ffcb59878593;rport=5061
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16186 PRACK
Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1
Content-Length: 0
--------------------------------------------------------------------------------
Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:57:708 (1025 bytes):
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-4ab17271000e1823454fdfbdecd95195;rport=5061
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16185 INVITE
Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1
User-Agent: snom360/7.1.30
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, callerid
Content-Type: application/sdp
Content-Length: 417
v=0
o=root 1426542497 1426542498 IN IP4 192.168.232.90
s=call
c=IN IP4 192.168.232.90
t=0 0
m=audio 52886 RTP/AVP 8 0 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:rs98bfKUTQhtHDhr6tNCmv1ZrIP7RmFHplOK0Y66
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=encryption:optional
a=sendrecv
--------------------------------------------------------------------------------
Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:57:958 (407 bytes):
ACK sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-2acacff56c07853ba001d3fe4cf46e3c;rport
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16185 ACK
Max-Forwards: 70
Contact: <sip:4406@192.168.232.10:5061;transport=tls>
Content-Length: 0
--------------------------------------------------------------------------------
Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:57:963 (468 bytes):
MESSAGE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-c538840cd1bf6ec221877a463f16c28f;rport
From: "ROE " <sip:4406@sip.domain.de>;tag=35457
To: "ROE " <sip:4406@sip.domain.de>
Call-ID: srjdn69q@pbx
CSeq: 19739 MESSAGE
Max-Forwards: 70
Contact: <sip:192.168.232.10:5061;transport=tls>
Subject: buttons
Content-Type: application/x-buttons
Content-Length: 21
k=40
c=on
x=ext
--------------------------------------------------------------------------------
Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:57:1000 (270 bytes):
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-c538840cd1bf6ec221877a463f16c28f;rport=5061
From: "ROE " <sip:4406@sip.domain.de>;tag=35457
To: "ROE " <sip:4406@sip.domain.de>
Call-ID: srjdn69q@pbx
CSeq: 19739 MESSAGE
Content-Length: 0
--------------------------------------------------------------------------------
Received from tls:192.168.232.10:5061 at 7/2/2008 20:10:01:175 (498 bytes):
MESSAGE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-0d3ac7578e7fdfe873e61040634b3c07;rport
From: "ROE " <sip:4406@sip.domain.de>;tag=15184
To: "ROE " <sip:4406@sip.domain.de>
Call-ID: jlb2mrq5@pbx
CSeq: 34430 MESSAGE
Max-Forwards: 70
Contact: <sip:192.168.232.10:5061;transport=tls>
Subject: buttons
Content-Type: application/x-buttons
Content-Length: 51
k=40
c=hold
x=ext
i=4451
n=*60115
a=invite
--------------------------------------------------------------------------------
Sent to tls:192.168.232.10:5061 at 7/2/2008 20:10:01:215 (270 bytes):
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-0d3ac7578e7fdfe873e61040634b3c07;rport=5061
From: "ROE " <sip:4406@sip.domain.de>;tag=15184
To: "ROE " <sip:4406@sip.domain.de>
Call-ID: jlb2mrq5@pbx
CSeq: 34430 MESSAGE
Content-Length: 0
--------------------------------------------------------------------------------
Received from tls:192.168.232.10:5061 at 7/2/2008 20:10:03:551 (528 bytes):
INFO sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-6dc49b483f259e6b15537191cf31cbb2;rport
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16187 INFO
Max-Forwards: 70
Contact: <sip:4406@192.168.232.10:5061;transport=tls>
Content-Type: message/sipfrag
Content-Length: 87
From: "ROE-Demo " <sip:4451@sip.domain.de>
To: "ROE " <sip:4406@sip.domain.de>
--------------------------------------------------------------------------------
Sent to tls:192.168.232.10:5061 at 7/2/2008 20:10:03:594 (365 bytes):
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-6dc49b483f259e6b15537191cf31cbb2;rport=5061
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16187 INFO
Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1
Content-Length: 0
--------------------------------------------------------------------------------
Received from tls:192.168.232.10:5061 at 7/2/2008 20:10:03:602 (1008 bytes):
INVITE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-5259f7fb7d7dc0c792e37d0e4f2fdeec;rport
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16188 INVITE
Max-Forwards: 70
Contact: <sip:4406@192.168.232.10:5061;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/2.1.5.2357
Content-Type: application/sdp
Content-Length: 378
v=0
o=- 40095 40096 IN IP4 192.168.232.10
s=-
c=IN IP4 192.168.232.10
t=0 0
m=audio 58138 RTP/AVP 8 0 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:MHLtx19IYi7nF6NjlHBXJIZRbYLG6MgJ5LnW5sbG
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
--------------------------------------------------------------------------------
Sent to tls:192.168.232.10:5061 at 7/2/2008 20:10:03:674 (1025 bytes):
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-5259f7fb7d7dc0c792e37d0e4f2fdeec;rport=5061
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16188 INVITE
Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1
User-Agent: snom360/7.1.30
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, callerid
Content-Type: application/sdp
Content-Length: 417
v=0
o=root 1426542497 1426542499 IN IP4 192.168.232.90
s=call
c=IN IP4 192.168.232.90
t=0 0
m=audio 52886 RTP/AVP 8 0 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:rs98bfKUTQhtHDhr6tNCmv1ZrIP7RmFHplOK0Y66
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=encryption:optional
a=sendrecv
--------------------------------------------------------------------------------
Received from tls:192.168.232.10:5061 at 7/2/2008 20:10:03:826 (407 bytes):
ACK sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0
Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-2acacff56c07853ba001d3fe4cf46e3c;rport
From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173
To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw
Call-ID: 02622e7d@pbx
CSeq: 16188 ACK
Max-Forwards: 70
Contact: <sip:4406@192.168.232.10:5061;transport=tls>
Content-Length: 0
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When I set the Failover Behavior to "only on 5xx codes" a field named "Request timeout" appears. Has anyone a recommendation for this value? What unit is used; secons or milliseconds?
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Ok, I have understood where the problem is. But is there a solution? We are using Snom 360 (Fw7.1.30) with factory default plus PNP-Settings and the latest PBXnSIP Software.
Is there a setting at the phones or the pbx that I have to change? The only thing I found is "Change names in To/From-headers:" in the domain-Settings. And I have not found any description for this setting.
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When an external caller ( A ) calls an internal destination ( B ) the phone ( B ) shows the right caller ID from ( A ). After an attended transerfer to an other internal Destination ( C ) the phone ( C ) shows the Caller ID from Phone ( B ).
Is this by design or is there a way to control this behavior.
In my opinion this should be optimal:
Phone ( C ) see in the ringstate Caller ID ( B ) and after a successful transfer the phone ( C ) see caller ID ( A )
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In the Domain admin pages (v2.1.5.2357) the link goes to http://wiki.pbxnsip.com/help/help and you get a 404 not found error.
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Mailbox escape is a greate idea. But after the announcement "press 0 to be connected to the switchboard" the caller have about 1 second to decide. If you wait longer you are connected with the mailbox and prssing 0 only generate sound on the Mailbox ;-) Is it possible to have the announcement of the Mailbox excape char at the beginning.
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PNP
in snom Phones
1. Yes I agree to this. This settings should be timezone dependent. In our German enviroment the after a PNP prov. the Phone have the following settings.
<timezone perm="RW">USA</timezone>
<tone_scheme perm="RW">USA</tone_scheme>
This settings could also be tied to the timezone setting of the PBX.
2. Can you advice how to configure with a numer of lines ;-)
Thanks
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Nothing conspicuous to see. Memory and CPU usage is extremly low. Have heart that at this time a user has a problems with callbach on busy. So today I have disabled "Offer Camp On". May be this could be a hint.
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Buttons are wonderful for me to configure :-)
I think the girls at the switchboard would love me if the Buttons are a combination of "monitor extension (on phone)" and "Speed Dial". It is possibe to get that feature very quick... best in the next release ;-)
Thanks
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PNP
in snom Phones
Sorry I dont get it....
I have Snom setting that should be applied on any phone that get it's config with PNP. These are:
<transfer_on_hangup perm="">on</transfer_on_hangup>
<call_waiting perm="">off</call_waiting>
<time_24_format perm="">on</time_24_format>
<date_us_format perm="">off</date_us_format>
:
:
Do anyone have an advice for Dummys what is to do?
- What file I have to be changed?
- What entrys have to be changed?
- What file I have to be placed in what directorys?
Any help would be fine!
Thanks
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Today our PBX crashes (2.1.5.2357 (Win32)). The only thing I can see is an entry in the System-Eventlog. Service "pbxnsip PBX" stopps unexpectly (translated). In the Application-Eventlog there is no entry. Are there any other log-files that I can send you to identify the problem?
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Is it possible to tell the PBX to dial out directly when the URL is called. In the moment we have to press "1" to start calling.
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PNP
in snom Phones
How does the pnp.xml file in the html-directory works together with the config files ind the generated directory? I want to set a phone parameter on all phones. Adding the parameter under
<file name="snom_3xx_phone.xml" encoding="xml">
<pattern>!snom_3xx_phone-([0-9A-F]{12})\.xml!\1!</pattern>
<parameter name="vlan"></parameter>
<parameter name="admin_pin">0001</parameter>
section has no effect.
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PNP
in snom Phones
Ok, I generate a xml file with my additional phone settings an put this file in the html directory. What is to do to bind this file to the pnp config. For a new phone I only want to activate pnp and the phone should get the PNP-config and the "custom" settings from my xml file.
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Thanks, this solves the problen
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PNP
in snom Phones
It is possible to add custom settings to the PNP settings? For example to have an other XML-File with custom settings that is send to the phone? It would be nice to define such a file globaly and per phone.
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We also wrote something up a thttp://wiki.pbxnsip.com/index.php/Click_To_Dial.
Is it possible to have a domain admin account that is allowed to dial in behalf of an other account. We have a software where the users are already authenticated. It would be nice to have one technical user account on the server that is allowed to dial for another user. So we don't have to put all passwords from all users in our software.
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Okay, lets keep a "Speed Dial" mode in the back of our mind - or use the one which is arleady available... But that one does not light up the LED. Might be not so diffilult to do that. Lets see if others also feel the need for that.
I Agree with Kristan. Buttons with Speed-Dial would be nice. We already use this but with buttons it more easy to configure.
Call Pickup
in snom Phones
Posted
Sorry, I think we are talking about different things. I will discribe it in more details:
Phone "A" Internal user "A" in Button-Group 1
Phone "B" Internal user "B" in Button-Group 1
Phone "C" Internal user "C" with Number "1234"
"C" is calling "A" and "B" picks this call with a Button on the phone. The Snom display ( Phone B ) shows "*6015775" and not the number from Phone "C" as the calling party.