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Since all of the call recordings are stored locally on the server...has anyone come up with a good solution to push the recordings out to the customer? Im running Software-Version: 5.1.2d (CentOS64) Ive tried getting a bash script to send the files out via FTP, but im no programmer and have not been successful. Does SnomONE have a built in feature for this? Any help is appreciated.

Looks like 20 is the default. Can anyone confirm?

Ive got a customer complaining that their voice mailbox is full. I see that their "Maximum number of messages" is set to default. I cant find this default value. Can someone please educate me on this? Are there any other factors as far as mailbox quotas go? Any other settings to take into account?

Too bads right. When can we expect the update?

Looks like the busy call was due to the PBX sending a CANCEL right after the INVITE. This issue has been opened in another forum.

No dead air, but getting busy tone wen dialing out on a few of these calls

Well, Looks like its ringing now. Ive not had any dead air calls. But on about 25% of calls there is no audio either way. Looking over the pcap I still see only inbound RTP. There was no audio on either side of the call.

Weird, So she switched phones with a known working phone. Now the new phone she has is not working and the old is.... Thought that all of the config files were generated off the same template so im not sure why it would be acting like this. Any ideas?

And what password? I assume the trunk secret but want to be sure. Ive got a lot of customers on this box so I want to make sure Im understanding.

Will this require a restart of the service?

How do you set the trunk up like that? I assume you mean on the PBX settings.

Ive not had this issue. My cust is complaining that he cannot log in. So he hits the login page and uses known good creds and it just keeps taking him to the same login screen. This is on firefox an chrome If he uses IE, it works Running Version: 5.0.10 (CentOS64)

I honestly do not know what we are getting at here? Sockets? The above link is confusing.

Got the following string off a Polycom forum: [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT So far so good.

Yeah mines the same: [2-7]x|8[1-7]x|[2-9]11|1xxxxxxxxxx|011x.T|*x.T What would cause it to connect after '80'? What do you suggest changing?

Ive got a cust that is having issues logging in to the web interface with Firefox or Chrome. I use Firefox exclusively and Ive never had issues. Anyone got any ideas? Typically he is able to log in once, but when he tries again later in the day it fails. It does work with IE (boo)

Here is the dial plan config: <dialplan dialplan.1.impossibleMatchHandling="0" dialplan.1.removeEndOfDial="1" dialplan.1.applyToUserSend="1" dialplan.1.applyToUserDial="1" dialplan.1.applyToCallListDial="0" dialplan.1.applyToDirectoryDial="0" dialplan.2.impossibleMatchHandling="0" dialplan.2.removeEndOfDial="1" dialplan.2.applyToUserSend="1" dialplan.2.applyToUserDial="1" dialplan.2.applyToCallListDial="0" dialplan.2.applyToDirectoryDial="0" dialplan.3.impossibleMatchHandling="0" dialplan.3.removeEndOfDial="1" dialplan.3.applyToUserSend="1" dialplan.3.applyToUserDial="1" dialplan.3.applyToCallListDial="0" dialplan.3.applyToDirectoryDial="0" dialplan.4.impossibleMatchHandling="0" dialplan.4.removeEndOfDial="1" dialplan.4.applyToUserSend="1" dialplan.4.applyToUserDial="1" dialplan.4.applyToCallListDial="0" dialplan.4.applyToDirectoryDial="0" dialplan.5.impossibleMatchHandling="0" dialplan.5.removeEndOfDial="1" dialplan.5.applyToUserSend="1" dialplan.5.applyToUserDial="1" dialplan.5.applyToCallListDial="0" dialplan.5.applyToDirectoryDial="0" dialplan.6.impossibleMatchHandling="0" dialplan.6.removeEndOfDial="1" dialplan.6.applyToUserSend="1" dialplan.6.applyToUserDial="1" dialplan.6.applyToCallListDial="0" dialplan.6.applyToDirectoryDial="0"> <digitmap dialplan.1.digitmap="" dialplan.1.digitmap.timeOut="" dialplan.2.digitmap="" dialplan.2.digitmap.timeOut="" dialplan.3.digitmap="" dialplan.3.digitmap.timeOut="" dialplan.4.digitmap="" dialplan.4.digitmap.timeOut="" dialplan.5.digitmap="" dialplan.5.digitmap.timeOut="" dialplan.6.digitmap="" dialplan.6.digitmap.timeOut=""/> <routing> <server dialplan.1.routing.server.1.address="" dialplan.1.routing.server.1.port="" dialplan.2.routing.server.1.address="" dialplan.2.routing.server.1.port="" dialplan.3.routing.server.1.address="" dialplan.3.routing.server.1.port="" dialplan.4.routing.server.1.address="" dialplan.4.routing.server.1.port="" dialplan.5.routing.server.1.address="" dialplan.5.routing.server.1.port="" dialplan.6.routing.server.1.address="" dialplan.6.routing.server.1.port=""/> <emergency dialplan.1.routing.emergency.1.value="" dialplan.1.routing.emergency.1.server.1="" dialplan.2.routing.emergency.1.value="" dialplan.2.routing.emergency.1.server.1="" dialplan.3.routing.emergency.1.value="" dialplan.3.routing.emergency.1.server.1="" dialplan.4.routing.emergency.1.value="" dialplan.4.routing.emergency.1.server.1="" dialplan.5.routing.emergency.1.value="" dialplan.5.routing.emergency.1.server.1="" dialplan.6.routing.emergency.1.value="" dialplan.6.routing.emergency.1.server.1=""/> </routing> </dialplan> I have my C3PO unit at the shop or Id have him translate it. Can anyone see whats causing my issue? Anyone able to replicate the issue on their PBX?

Does anyone have a polycom that they could test this on? Try 'live' dialing (801) 373-9120 (Provo UT Temp and Time)

Its hard to say if this is a SnomONE issue or a Polycom issue. I test a 'live dial' from the polycom and it connects right after '80' is typed. works if '1801' is dialed. I tried this from a Snom360 but this phone acts a little differently. I pick up the headset and dial the full number (801-XXX-XXXX) but it makes me press the check mark to complete the call. If I dial 1801 it completes the call automatically without having to press the check mark. So its hard to tell if its a setting on the PBX or phone. Ill look at the Polycoms settings.

So on my Version: 5.0.10 (CentOS64) pbx im having a global issue. We are located in UT where the predominant area code is 801 When dialing an 801 number, the call connects immediately after dialing '80'. (Picking up headset and dialing) If we dial 1-801, it works fine. Alternatively you can type the number in, 801-XXX-XXXX and then hit dial on the phone and the call connects. This does not happen on any other area code. Meaning you can live dial any other number without adding the 1, i.e. 1-801. It seems to be special to the string '80' Anyone else experience this issue? I know that '8' is the voicemail prefix. Is this perhaps messing this up? How would I go about fixing this. Thanks in advance Sudo

Ive got a customer that has recently got a wireless headset (Zabra) She was complaining that it was not working fully on her phone. She was able to answer calls only if it was in its charger. When it was off the charger and on her head, the calls ring through as though she is on the phone. I had her test the headset on a different ext and it was working fine. We re provisioned the 2 extensions and swapped ext numbers so the working Polycom was now the receptionists ext. The interesting thing is that it is now failing in the same was as it did on the initial phone. Now, Im pretty sure there are no setting on the PBX that would affect the headsets relationship to the hard phone. But with this new development Im not sure. Any extension setting that would affect the way a wireless headset answers calls?

There was a video on creating a .csv for importing extensions. Is that process outlined somewhere on the site?

Basically I have a request to setup a system that acts the following way. They want it to ring 5 extensions. The thing is they want then to ring individually. So it rings the first ext twice, then the next without including the previous extension. After the first cycle, they want an escape message to play leave a voicemail. Then they want it to cycle again the same way and after the second cycle it forces them to the mailbox. Any suggestions on this? It doesn't look like the agent queue will ring the extensions individually. Would modifying the hunt groups be the best idea?

Can you please tell me what template file to modify and where to modify it to add additional ring stages in a hunt group? Is this possible?

I used to be able to hit onlinetraining.snomone.com but cant anymore. Ive tried onlinetraining.snomone.com, onlinetraining.snom.com, training.snomone.com etc, but I cant hit it. Is this site still up? I really hope that it is.