pbx support
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We have seen cases before where the PBX is sending the "group name + calling party name", but the phone displays just the calling name. That's why I wanted to see what is in the INVITE that is sent to the phone.
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If the PnP is happening properly, then you will see that folder. If the files in that folder are not getting updated or they are not being generated at all, then there is some issue with the PnP.
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Does the phone requests that file? Did you check the phone log to see if it is requesting that file?
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If the phone publishes the www contact like below in the REGISTER message, these links are enabled.
WWW-Contact: <http://192.168.2.12:80>
WWW-Contact: <https://192.168.2.12:443>
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2 things -
- Did you really see the packet is leaving the system where the PBX running? This can be done using a wireshark (or some other network tracers)
- Did you verify that the provider has not filtering packets based on user agent string?
- Did you really see the packet is leaving the system where the PBX running? This can be done using a wireshark (or some other network tracers)
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The star code (*93) is interfering with couple of other features. We want to see if we can remove that star code support. But before that we want to see who and how many are using this feature.
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Not sure what you are looking for. As I mentioned none of the emails are editable, generally.
Have you tried editing the an email that you received from someone else (other than PBX CDR)? You can edit only if you hit "reply" or "forward"
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Yes, it is going to be in the next release.
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i am not sure which one is better - if listening to recordings from the phone or login to the web interface and playing from there. We have added the web interface support for the next version.
We can take a look at you visio diagram too if you have one.
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Only Windows versions have that support. Mac and other Linux versions, it is not supported.
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v4.5 should allow you to configure the 720 phones. Do you have any customization done on the html or the pnp file? If you have, please back them up(for later use) and revert everything back to default.
Also, the 720 phones come with built in certificates that are trusted by the PBX. So, whether the phone in the LAN(same network as the PBX) or WAN (different network), you should be able to perform the PnP.
On the PBX, turn off the "Admin->Settings->PnP: Trust MAC address" and reboot the phone.
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Looks like you have some customized html file (check either 'html' or 'webpages' folder under PBX working directory). Please back up those directories and then delete them, restart the PBX.
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I assume this is the INVITE from PBX to Hunt extension. Can you post the INVITE that is received by the PBX, or simply, post the full SIP log?
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Thanks. We will make the default as "off" instead of "on" in the next version.
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Not a PHP expert, but can you see everything is sent out on the wire as explained here? - http://wiki.snomone.com/index.php?title=CSTA_example for details.
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Can you let us know the size of the document? Or else, could you please email the document to support@snom.com? We should be able to either upload it on your behalf and/or put it on the wiki too.
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Use !(\+[0-9]*)!\1!t!300 for handling +.
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We do have another parameter called 'user' that you can use.
<tr>
<td>Extension:</td>
<td>{ssi variable user}</td>
</tr>
But today it does not display the user account info. Instead it displays the user ID (index for the table), which is not very useful on the email. We will add that to the next version.
We probably did not print the extension info because, if you receive an email, you already know which extension you have received the emails for. We can see issue, if you get emails for multiple extensions.
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It's been logged at level 9 with "Log general events". Maybe you have level 8 turned on?
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i do not think hunt group behavior has changed in that area. PBX always rings all the "free" extensions at each stage. If the extension in question was busy when stage where it belongs is being processed, then it will not be called for that particular call.
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The call log is displayed from 'cdrt' folder. The extension call logs are displayed only under the respective user portals.
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For some reason, the match is failing and the call is always going to the default destination 300.
After the log
Incoming call: Request URI sip:1111111111@{customersIP}:5060, To is <sip:5555555555@{customersIP}:5060>you should see something like
Incoming: formatted To is....
. I wanted to see that.
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Not sure if this is a OS related issue. If PBX is getting 603 Declined, then someone on the trunk side rejecting it. Maybe thy did not like the number itself.
But if you want to try it on another OS, Linux is popular. Lot of folks are using CentOS-64 version.
We also have the virtual appliance for vmware. Check out http://snomone.wordpress.com/2012/02/21/snomone-virtual-appliance/
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Yes. You can verify this by checking the pbx.xml file.
remote call initiation on 4.5 swapping headers if connect=false
in General Setup
Posted
Can you also post the URL that you are using (the sample is enough to see the syntax) so that recreate what you are doing?