Happy
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Posts posted by Happy
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I think the hunt group is just not the right way to do this. The ACD has a lot more possibilities to keep callers lined up nicely and you can record some prompts informing them and keeping them lined up. Then you have all the time to process them one by one. In case there is no line, the caller would still get the ringback tone immediately.
I read the admin manual and indeed Automatic Call Distribution seems to be the better way to go. I'll give it a try !
Marc
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Right now, the park orbit remembers which extension (= neck to grab) parked the call. If we would have to send the call back to the group, the call might end up with someone else, or, even worse, after nobody wanted to pick up, in a redirected number, maybe even outside.
I would just not send the call to a park orbit. Maybe just send them to a ACD or to a mailbox.
Our problem is this :
- we would like to get a call put on hold by pbxnsip, if the operator is busy on the phone. It is frustrating to pick up the next line, tell them to be patient and then put them on hold, then going back to the first caller.
- but there should be a possibility to be warned IF the operator forgets the second pending call afterwards. Wireless phones for example don't have BLF to warn you.
Maybe there is a solution or workaround to get this done, but I don't find it in the (wealth of) tools pbxnsip is offering.
Best Regards,
Marc.
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If an extension parks a call, the call normally gets back to the extension after some time.
Using this logic to a HG, I would expect the call to go back to the HG that initiated the park. But that is not what is happening : the call remains in the park orbit forever.
Is it because I put the park orbit number in the last stage of the HG ? I know that would create a loop, but if it checks the presence of the call inbetween, that is not a problem. It is nice to be able to put a call in a park orbit if you are not able to handle it instantly, but having no reminder except a BLF is a bit over the edge...
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Hello,
I want when i call some extension ( such as 15 ) ring the extension 15 and 17, such as Follow me in Asterisk. Is it possible here???
Thank you
Why not make a hunt group and ring the hunt group instead ??
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Phones Snom320 in the system have 3 BFL, monitoring specific CO-lines CO1-2-3.
CO1 and CO2 are related to a Isdn Trunk. CO3 is related to a voip trunk with several other CO lines (3-4-5-6).
The voip-trunk is used for outgoing calls. The problem with this setup is that the first user calling out gets CO3. Next user jumps to the isdn-trunk on CO1. It looks like the system is not offering CO4-5-6 because these are not on the phone buttons. According to the dial-plan, they should be used.
If I make the CO3 button an active line, it will also be linked to incoming calls on CO1 and CO2.
I suppose there is something wrong in the setup; maybe someone has a better way to reach the same goal : being able to monitor the isdn-trunk AND being warned that there is phone traffic on the voip-trunk ...
Best Regards,
Marc.
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If you use "buttons" it should be all automatic. Apart from the *87 feature code (e.g. *8771 to pick up from accoúnt 71) there is another "hidden" feature code that connects to a call, but for that one you need to know the call number; which is provided by the PBX within the button protocol.
Do you mean the pickup also works when a button is connected to account 71 and account 71 is a IVR or AA ?
Does Pbxnsip make a difference there between 'normal' extensions and Ivr or AA nodes ?
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Once the call comes into the system you can grab the call by pressing the co-line or by dialing *87 which is the star code feature to pick up the calls but if a call is been routed to the IVR or AA then you have to wait until the calls has been routed out to an extension in other to retrieve it.
So all I can do is send the call to a park orbit with a suitable music on hold and corresponding BLF and let it get to the hunt group afterwards. Then it should be possible to pick up the call on the corresponding park button.
Thanks anyway,
Marc
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Is it possible to pick up a call from a Ivr or AutoAttendant ?
Current situation : line comes in, lights up a CO-line button, gets to a Ivr, and then we have to wait until the Ivr has finished and the phone is ringing to be able to pick it up. I guess it must be possible to pick up the line from the Ivr, but I can't find out how ...
Any advice highly appreciated.
Marc
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You are right of course, on the other hand, even the snomone forum says it's possible...
Linksys routers of the type Wrt54G are capable to run the Tomatovpn firmware by Keith Moyer.
Essencially they create one or more vpn-tunnels to different networks. Networks can be connected over the internet as if they were one. Even a dhcp-server on one end serves computers on the other end, so look out if you start connecting.
A snom 320 on the 'other' end of the vpn is provisioned as if it is in the network. Connections are very stable, sound quality as well. You should of course play a bit with the qos settings, as other traffic passing the tunnel can easely interfere with the sound quality.
Succes,
Marc
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Are you asking about this? - http://wiki.snomone.com/index.php?title=General_Logging_Settings
I actually mean the call-history : I can only retrieve one page there of about 20 lines and that is it.
It surely must be possible to have a call-history of for example the last month ?
Best Regards,
Marc
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The trace states clearly it is an unsupported codec issue.
Either get them to swap to G711u/a OR buy an upgrade license.
Provider claims the G711u/a codec is available.
Looks anyway like it is not used then.
I'll check the snom sales department.
Thanks,
Marc
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If this is snom ONE free, it does not include the G.729A codec. Unfortunately, this codec is not free and much be puchased; that's why we cannot include it in the free edition.
Provider Twytel provides also codec G711u - G711a.
Is it possible the error has another background ?
Best Regards,
Marc
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If this is snom ONE free, it does not include the G.729A codec. Unfortunately, this codec is not free and much be puchased; that's why we cannot include it in the free edition.
It is Pbxnsip - 10 user licence. But it does not have that codec either.
Is there a way to convert our licence into a snom licence with the codec ?
Best regards,
Marc
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I'm registering to a new provider Twytel.
Incoming calls are received, but rejected with error 415.
Could anyone jump in with a solution for the problem ? Provider did try several codec, but just did not work.
In addendum the log-file of a incoming call
Thanks,
Marc Happaerts
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You can set it using "Logging->Log Length". But it is just for the web interface, i.e., how many lines of log you want to see on the "Logfile" page. It does not have any impact on the log file (if being written).
Is it written somewhere by default, or what log settings do I have to use ? I can't find much of a clue in the log settings on the admin tab page.
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CO lines becomes tricky once you start having more than 1 trunk. If you have multiple trunks with CO lines and if you map the buttons to those CO-lines, then only way to really select the "right" trunk is by pressing the co-line (shared line) button corresponding to that trunk.
That is the first clear answer I get regarding CO-lines. Seems indeed the 'better' way to do it, as it works without any problem.
If you just pick the phone and dial the number, then PBX choses the first free co-line (could be the trunk that you are not expecting).
This needs some further explanation : my dial-plan says - first take trunk B (CO lines 3-6) / if not available fallback to trunk A (CO lines 1 - 2).
Now what is the first CO-line to pbxnsip : CO line 1, or the first in the first trunk of the dial-plan ? Ok, sounds messy, but I hope you get my point...
Also, I would suggest to skip the 1st button while assigning the co-lines on the phone. It is much cleaner that way. Otherwise, it is bit confusing to see the lights and say if the phone is using co-line or the light is just related to the identity.
Swiched indeed the phone buttons to that setup after a while. Is quite confusing the other way.
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http://pbxnsip.com/protect/beta/win32/pbxctrl-4.2.0.3970.exe should take care of this issue.
I'm using now version 4.2.0.3981 (Win32). Still my CO-lines block the dial plan from dialing out with anything else then the default gateway.
Is somethimes really annoying, as every now and then we have to be able to choose a certain trunk.
Any fix in sight ?
Thanks,
Marc
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Is it possible to set the log length for the snomone call log ?
Thanks,
Marc
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I want to create an 870 for a remote user and was told that VPN is overkill.
Is there an idiot's guide to remote srtp?
Will Snom One provision remotely or will it even provision for this use?
Basically, treat me like an idiot. It would seem to be an accurate description at the moment.
I can only tell what I'm using. I'm creating an openvpn connection with 2 linksys wrt54G modems, flashed with Keith Moyer's TomatoVpn firmware. My snom 320 connects through the openvpn just like it's connected to a 'normal' internal network. It is simple, cost-effective and above all, works like a charm.
Best Regards,
Marc.
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One think that strikes out is the number "011-33333". In SIP the "-" is a part of the URI, in other words "011-33333" is not equal to "01133333". That might cause a lot of confusion.
The PBX likes to display numbers with "-" in it, but only for display purposes! Internally, the processing goes on without the "-". But when the number comes from a non-human device like the PSTN gateway, then the PBX will take it literally.
Hi,
I did a replace on the phone numbers, the '-' is there because of that. All phone numbers are in the original logfile in a complete numerical format. I am sorry for the confusion here.
The situation now is that the IVR connects directly to the HG for transferring the call. It seems to me that - for as long as you are in the IVR - pbxnsip does not detect a hangup. Would it be better to put in an extra IVR with a small message ? Probably Pbxnsip checks the status of the line in between the 2 IVR nodes?
Marc.
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Yes, you can use your pbxnsip key with snom ONE. You will get the snom ONE limitations on non-snom devices though.
pbxnsip update appears to be here:
Thanks Matt !
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In addendum log-file, I did log to a file directly, so nothing will be purged.
Ok, no log file added.
We'll give it a second try.
Regards,
Marc
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All,
We released first version of snom ONE(4.2.0.3958) during Oct 2010. After some good feedback from the you folks, we worked hard to fix most of the defects reported. The result of that is the latest bug fix release(4.2.0.3981).
For details such as release notes, downloads, upgrade procedure etc, please refer the below link
http://wiki.snomone.com/index.php?title=Upgrades
Thank you everyone and greatly appreciating the cooperation!
- Team snom ONE
Hi,
Is there a pbxnsip upgrade available too ?
Is it, by the way, possible to convert a pbxnsip licence to a Snomone licence. Since (almost) all my phones are Snom, Snomone Yellow would be great to me.
Regards,
Marc
Missed calls / Received calls
in Hunt Group Setup
Posted
Pbxnsip (latest version) with Snom 320 phones (firmware 8.4.18).
HG sends call to several extensions, one picks up the line.
Other extensions all show the incoming call in the 'missed calls' listing !
To me, these are not missed calls, so list is somewhat useless.
Hope someone has me an answer
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This is what shows in the sip trace log of a snom 320 phone, where the call is recorded as missed :
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Received from tls:192.168.1.15:5061 at 7/9/2011 11:48:01:199 (401 bytes):
CANCEL sip:12@192.168.1.22:2077;transport=tls;line=k3azpaae SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK-b74cb9969f17298dba519f36657880d4;rport
From: "Bureel" <sip:0123456789@pbx.praktijk.be:5060;user=phone>;tag=56269
To: <sip:92@pbx.praktijk.be>
Call-ID: 56d6b7ca@pbx
CSeq: 15711 CANCEL
Max-Forwards: 70
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0