RobertO

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.

It is possible to see the number from the person who is calling and not *601<call-identifier>?

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

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

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!!!!

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

Do you have an URL to request the CSTA webservice?

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?

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.

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 )

In the release notes from Version 2.1.5. I have read that there is a start of CSTA support. Is there code sniplet/example or something like this to see how it works?

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.

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.

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

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.

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

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

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?

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.

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.

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.

Thanks, this solves the problen

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.

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.

I Agree with Kristan. Buttons with Speed-Dial would be nice. We already use this but with buttons it more easy to configure.