pbx support

No, that has to be done wherever you are collecting the CDRs.

The link talks about the static mechanism of stopping the process, changing the xml and then starting it. The easy thing to do is to use the mechanism to change any global variable as explained here - http://wiki.snomone.com/index.php?title=Global_Configuration_File: So, you would login to the PBX as system admin and then type http://<pbx-ip>/reg_status.htm?save=save&out_of_dialog_options=true

45 is the agent and 73 is agent group as explained in the message that I posted <agentID>45@domain.com</agentID> <acdGroup>73@domain.com</acdGroup>

Looking at the version you are using I would recommend http://downloads.snom.net/snomONE/win32/pbxctrl-2011-4.3.0.5021.exe . This version is very close to what you have with few fixes. But if you want a forklift upgrade, then you can use http://wiki.snomone.com/index.php?title=Upgrades page to choose the latest. If you decides to move to any of 4.5 versions, then i suggest you to backup the PBX working directory using the OS file manager.

Generally domain delete should clean up the trunks too. In any case, you can check the 'trunks' folder for xml file that contain the domain index of the deleted domain. If you do not have any trunk xml files that contain deleted domain, then a PBX restart might help.

That is a random issue that we saw in the older versions of the PBX. That issue was fixed in the newer versions.

"Domain->Settings->Authentication user" is the user name for the phone web UI.

Not sure what you folks are dreaming about! This is what the GetAgentState request and response look like <?xml version="1.0" encoding="UTF-8"?> <GetAgentState xmlns=“http://www.ecma-international.org/standards/ecma-323/csta/ed5"> <device>45@domain.com</device> <acdGroup>73@domain.com</acdGroup> </GetAgentState> <?xml version="1.0" encoding="UTF-8"?> <GetAgentStateResponse xmlns=“http://www.ecma-international.org/standards/ecma-323/csta/ed5"> <agentStateList> <agentStateEntry> // max 32 <agentID>45@domain.com</agentID> <loggedOnState>true</loggedOnState> <agentInfo> <agentInfoItem> <acdGroup>73@domain.com</acdGroup> <agentState>Ready</agentState> // agentNotReady, agentNull, agentReady, agentBusy, agentWorkingAfterCall </agentInfoItem> </agentInfo> </agentStateEntry> ... ... <agentStateEntry> ... </agentStateEntry> </agentStateList> </GetAgentStateResponse>

If you are talking about the phone's web interface, then you have to use the Domain->Settings->Admin password. If you are talking about the user portal on the PBX (extension's PBX web portal), then use the Extension->Web Password.

It is scheduled for the next release, next quarter. We really do not want to introduce this change in any of the minor versions to confuse folks.

You can't really use the call id from the ajax call_list to perform CSTA call Transfer call. You really have to use the CSTA steps (start application, monitor device etc) before you can use the TransferCall. The call id that you get in the ajax call_list is local index, not the SIP call id.

Wrong forum?

If you see "Tr" in the log message, that means PBX is repeating the same message again because it did not receive any response. [5] 20120822130040: SIP Tr udp:216.115.69.144:5060: INVITE sip:12034267166@sip.flowroute.com;user=phone SIP/2.0 If this was a working system and suddenly stopped (without any software upgrade), then it is generally something outside the software causing it. Could be external firewall or if the system was rebooted, then sometimes system firewall can cause this.

It is not the same problem. In your case, the none of the messages are reaching "216.115.69.144". So this is either network issue or DNS issue resolving to the wrong address.

No.

To summarize, on both 720 and 320, the pre-configured DND button does not work. Is that correct? You can put then a notepad/wordpad and attach them. BTW, check SIP log on the phone too when you press the DND button.

There is a setting under AA - "Accounts that cannot be called:". If you put those extensions under that setting, then they won't appear in "Dial by name" list. This can be a workaround for now.

In 4.2.1.4025 INVITE sip:2586286@24.96.139.170;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.36.1.18:5060;branch=z9hG4bK-51894bc0201b0a0fcd1f844b0562cbdd;rport From: <sip:8502345271@24.96.139.170>;tag=63215 To: <sip:2586286@24.96.139.170;user=phone> .... Contact: <sip:8502345271@10.36.1.18:5060;transport=udp> User-Agent: snom-PBX/2011-4.2.1.4025 Remote-Party-ID: "Ron Hardy" <sip:18502345271@pbx.gulfworld.com;user=phone>;party=calling;screen=yes In 4.5.0.1090 INVITE sip:2586286@24.96.139.170;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.36.1.18:5060;branch=z9hG4bK-385ccc20f762fcb4bbbaaf4ecb4646bf;rport From: "Ron Hardy" <sip:209@pbx.gulfworld.com>;tag=3522 To: <sip:2586286@pbx.gulfworld.com;user=phone> ... Contact: <sip:8502345271@10.36.1.18:5060;transport=udp> ... User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Remote-Party-ID: "Ron Hardy" <sip:18502345271@pbx.gulfworld.com;user=phone> Privacy: id P-Charging-Vector: icid-value=;icid-generated-at=10.36.1.18;orig-ioi=pbx.gulfworld.com Looking at these 2 traces, in 4.2, both From header and the Remote-Party-ID header had same number. In 4.5, From header was as extension. Also, in 4.5, there are other headers (the provider may not like some of those too). So, after you upgrade to 4.5, please modify these on the trunk Set the "Remote Party/Privacy Indication:" to "Customs Headers" and set the following drop-down to Request-URI: Let the system decide (default) From: Based on incoming call To: Let the system decide (default) P-Asserted-Identity: Don't use header P-Preferred-Identity: Don't use header Remote-Party-ID: Based on incoming call P-Charging-Vector: Don't use header Privacy Indication: Don't use header Basically, play with the "From" header drop-down, just in case if they don't like the above mentioned header. BTW, you can also, choose "Other" for the "From" header and set something like "My company name" <sip:{trunk-ani}@{domain};user=phone> or "My name" <sip:{ext-ani}@{domain};user=phone> or any other combination of tag/template from http://wiki.snomone.com/index.php?title=Trunk_Custom_Headers

We have the same mapping and it works fine. Did you check the phone logs when you press that button? Does it happen on just 1 phone/model or on other phones/models too?

Forgot to upload the file? Yes, the newer version has better control on the logs. For the older version please use http://wiki.snomone.com/index.php?title=Retrieving_SIP_logging for help.

I am assuming you are talking about the snomONE plus appliance. Here is the download link. Please use this for test purpose only (as it is still being tested here with other changes). http://downloads.snom.net/snomONE/centos32/v4.6/pbxctrl-centos5-4.6.0.1008

The license key that you posted seems to fine. Did you restart the service just to be sure?

Seems to work fine on the 720 here, running SIP 8.7.3.10 Setting DND with *78/*79 works as expected - Works here too. Setting DND as one of the 18 Programmable keys works as expected (Sends *78/*79 as DND on/DND off) - Did not test it here. Setting DND using the DND key on the phone has no effect at all - Works fine here. Can you log into the phone's web interface and check the function key mappings?

PBX generally connects to the remote server and if the connection is successful, then it will send the CDR. So, make sure the remote server is accepting the connections from the PBX.

Unfortunately, today there is no option to exclude specific extensions from dial by name feature. Maybe something for the future.