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penta s.r.l.

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Posts posted by penta s.r.l.

  5. Snom ONE 3rd party registrations problem

    in General Setup

    Posted

    Read my article, especially point 3:

    http://windowspbx.blogspot.com/2011/02/gotchas-to-watch-for-when-migrating.html

     

    "Remember that these allowed non snom devices will have a limitation that they will not support a SIP REFER. In the real world this means that, for one, the Transfer button on software and hard phone will likely not work. This is not the end of the world though as with snom ONE you can put the call on hold and use the built in star code to transfer as a work around."

    Thank you Matt for the clear answer.

  6. Snom ONE 3rd party registrations problem

    in General Setup

    Posted

    Hi,

    we have a snom ONE yellow license installed and we are using a mix of snom/non snom phones.

     

    With great pleasure I've read that "snom ONE yellow supports 10 third-party devices", so I've installed it, since we have 5 thid-party devices.

    The problem is that phones still cannot transfer calls and the log still reads

    "REFER from device type "Linksys/SPA941-5.1.8" is not supported in this product"

     

    We are running version 2011-4.2.0.3981 (Linux)

     

    Could you help me?

    Thank you.

  7. Call transfer ended

    in Call Treatment

    Posted

    Yes, a third party registration is designed for ATA (for fax), SIP phones etc Standard SIP phones will allow a registration, calls in and out and hold.

     

    The free version allows 2 3rd party registrations.

    The strange thing is that it allows the registration of these devices, but not the transfer operation. :huh:

    If you park and pickup between 2 of these "not supported" devices, it works. :blink:

    Avoiding the registration process should make more sense, IMHO.

    From this my perplexity.

  8. Call transfer ended

    in Call Treatment

    Posted

    Hi,

    I'm trying Snom ONE and if I try to transfer a call, the pbx terminates the call.

     

    From the log I can read this line:

    REFER from device type "Linksys/SPA941-5.1.8" is not supported in this product

     

    Is this the 3rd party limit applied from Snom ONE?

     

    Thank you in advance.

  9. pbxnsip now part of snom

    in Announcements

    Posted

    I think that snom phones integration with the pbx should constitute a "big" reason to sell snom phones.

    For this reason, Snom choice to provide a free pbx that supports (almost) only snom phones is understanble.

     

    What I don't understand is the not surmontable limit of non snom phones.

    Why don't they simply sell a license in order to use a non snom agent?

     

    Do you want to use 5 snom phones and 9 non snom phones? Pay for 9 non snom agent.

     

    Just my 2 cents

  13. pbxnsip now part of snom

    in Announcements

    Posted

    No, they convert 1:1. You get exactly what you had before, but when switching to the snom ONE you have the third-party restriction (which would then allow zero third party registrations...). Probably a bad deal, so I would recommend to stay with the pbxnsip builds if you have old pbxnsip license.

    Hi,

    what are the differences from the last pbxnsip version and the snom one version?

  14. Transferring all incoming calls to a cell phone on request

    in Using Cell Phones

    Posted

    You have to do 2 things here

    • Create a service flag that has proper time settings based on your requirements.
    • In the AA 60, there is a Night Service section. Put the service flag account number that you just created under "Service Flag Account" field & the cell phone number under "Night Service Number" field

     

    This will send the call to cell phone without playing welcome message. If you want to play a welcome message, then follow the similar steps to send the call to an IVR node first (instead of cell phone), play a message and then out to cell phone.

     

    If you just want to play an greetings, then you can use the IVR page on the AA account.

    Is this working in pbxnsip v3 too?

  16. Transferring all incoming calls to a cell phone on request

    in Using Cell Phones

    Posted

    Is it possible to let the pbx automatically transfer all incoming calls to a cell phone on request?

     

    Scenario:

    The incoming trunk send calls to extension 60.

    60 is an Auto Attendant that after the welcome message, redirects to account 70 (an Hunt Group).

     

    Is it possible to tell the system (via a Service Flag for example) that all calls must be redirected to an external number (or registered cell phone) instead of the 70 account?

    This should happen after the welcome message has been played.

     

    Thank you.

     

    --

    Nicola M.

  19. suggestion for a DECT system

    in VoIP Phones

    Posted

    with the DECT phone your using with the ATA, can you transfer calls from one handset to another?

     

    Sure, using pbxnsip transfer not the transfer offered by the base.

    Consider that we have 4 bases and 4 handsets.

     

    how many calls can you take on one device? how many callers can you put on hold?

     

    I haven't tried to use more than one call, so I cannot give you any hint on this.. sorry.

     

    i am looking for a good dect phone for a client, they have used the aastra and are very unhappy

     

    I don't know if this could be a good solution for you, anyway I solved my problems with C470IP.

    I'm evaluating the snom m3, and I opened a topic asking info about it.

     

    Bye

     

    --

    Nicola

  20. Gigaset C470IP and Call transfer via R key

    in VoIP Phones

    Posted

    Yes, that's the problem. The handset tells the PBX that some key "16" has been pressed. But the PBX does not know what to do with that... (sorry for the 200 Okay, which essentially means "got it").

     

    If the handset is able to send REFER then you have to use that mode...

    As you can see from the screenshots posted in the first message, there are two options related to REFER.

    Not one combination of these options is working. :(

     

    --

    Nicola

  22. suggestion for a DECT system

    in VoIP Phones

    Posted

    Hi Nicola,

    I was wondering which problems / bad experience you are talking about.

    Rudi

    Hi Rudi,

    the system I had problem with, included pbxnsip plus 2 C470IP, each with 2 C47H handsets.

     

    The first problem is the compatibility with the pbx.

    You cannot use the R key to hold ora transfer a call, your only option (AFAIK) is to use the Options->External menu and then send to pbxnsip the correct extension. To transfer the call, again Options->Transfer (!?)

    This way to work slows down the employes a lot.. apart from that the phone (or its display) itself is really slow, you are always waiting for it.

     

    The base is operating like a small pbx.. you can easily transfer calls between handsets registered within the same base, but the interaction with other handsets registered on other bases is difficult (like explained before)

     

    The C470IP can (theorically) sustain 2 VOIP calls at the same time, so I configured 2 VOIP profiles on each base and assigned a profile to each handset.

    So, why in the world is extension 40 busy, if 41 is speaking? :-)

     

    Anyway we have solved.

    I decided to try the old DECT (always Siemens) phones coupled with LinkSys PAP2.

    They work! They are fast and the employes are happy.

     

    Maybe those C470IP can be integrated with pbxnsip and all these problems are due to my lacking of knowledge.

    Anyway is just my experience, and hopely it will be helpful to others.

     

    bye.

     

    --

    Nicola

  23. suggestion for a DECT system

    in VoIP Phones

    Posted

    I'd need to buy a DECT system for a customer.

     

    After the bad experience with C470IP phones, I'd need to be sure about which DECT phones have total compatibilty with pbxnsip.

    Could you tell me wich products I could take into consideration??

     

    Thank you.

     

    --

    Nicola

  24. Gigaset C470IP and Call transfer via R key

    in VoIP Phones

    Posted

    Those settings seem to make sense...

     

    Do you see a REFER going out to the PBX? The PBX does not perform the job of collecting the digits; this must be done by the SIP endpoint (making it also possible to edit it before sending it). Then from the PBX perspective the way of collecting the transfer information should not matter--this is a implementation detail of the endpoint user interface.

     

    Hi,

    I don't see a REFER going out.

    Here it is the piece of pbxnsip's log collected when I press the R key.

    You can see the INFO DTMF 16 and the DTMF R, but these info are sent to the pbx immediately, the phone does not wait for the user to compose a number.

     

    Any hint?

    Thank you

     

    [7] 2009/10/05 10:00:31: SIP Rx udp:10.xxx.xxx.19:5060:

    INFO sip:41@10.xxx.xxx.1:5060 SIP/2.0

    Via: SIP/2.0/UDP 10.xxx.xxx.19:5060;branch=z9hG4bK48ab1f49244014cb3f8d9b34707cf0b;rport

    From: "Francesca" <sip:41@10.xxx.xxx.1>;tag=2221630226

    To: <sip:41@10.xxx.xxx.1>;tag=674e3c5180;user=phone

    Call-ID: 2757463928@192_168_1_19

    CSeq: 4 INFO

    Contact: <sip:41@10.xxx.xxx.19:5060>

    Max-Forwards: 70

    User-Agent: C470IP021910000000

    Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY

    Content-Type: application/dtmf-relay

    Content-Length: 22

     

    Signal=16

    Duration=86

    [8] 2009/10/05 10:00:31: Received INFO DTMF 16

    [9] 2009/10/05 10:00:31: Resolve 281: aaaa udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:31: Resolve 281: a udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:31: Resolve 281: udp 10.xxx.xxx.19 5060

    [7] 2009/10/05 10:00:31: SIP Tx udp:10.xxx.xxx.19:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 10.xxx.xxx.19:5060;branch=z9hG4bK48ab1f49244014cb3f8d9b34707cf0b;rport=5060

    From: "Francesca" <sip:41@10.xxx.xxx.1>;tag=2221630226

    To: <sip:41@10.xxx.xxx.1>;tag=674e3c5180;user=phone

    Call-ID: 2757463928@192_168_1_19

    CSeq: 4 INFO

    Contact: <sip:41@10.xxx.xxx.1:5060>

    User-Agent: pbxnsip-PBX/3.4.0.3201

    Content-Length: 0

     

     

    [6] 2009/10/05 10:00:31: Received DTMF R

    [9] 2009/10/05 10:00:31: Resolve 282: aaaa udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:31: Resolve 282: a udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:31: Resolve 282: udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:32: Resolve 283: aaaa udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:32: Resolve 283: a udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:32: Resolve 283: udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:35: Resolve 284: aaaa udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:35: Resolve 284: a udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:35: Resolve 284: udp 10.xxx.xxx.19 5060

    [7] 2009/10/05 10:00:35: SIP Rx udp:10.xxx.xxx.19:5060:

    BYE sip:41@10.xxx.xxx.1:5060 SIP/2.0

    Via: SIP/2.0/UDP 10.xxx.xxx.19:5060;branch=z9hG4bK207e79b1ac5dd431d0cdd532dec7d3b;rport

    From: "Francesca" <sip:41@10.xxx.xxx.1>;tag=2221630226

    To: <sip:41@10.xxx.xxx.1>;tag=674e3c5180;user=phone

    Call-ID: 2757463928@192_168_1_19

    CSeq: 5 BYE

    Contact: <sip:41@10.xxx.xxx.19:5060>

    Authorization: Digest username="41", realm="10.xxx.xxx.1", algorithm=MD5, uri="sip:41@10.xxx.xxx.1:5060", nonce="b4a9162aef36ee740423c0c11a30c07f", response="d55237bd7a3f24ae3cd92df9af807f48"

    Max-Forwards: 70

    User-Agent: C470IP021910000000

    Content-Length: 0

     

     

    [9] 2009/10/05 10:00:35: Resolve 285: aaaa udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:35: Resolve 285: a udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:35: Resolve 285: udp 10.xxx.xxx.19 5060

    [7] 2009/10/05 10:00:35: SIP Tx udp:10.xxx.xxx.19:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 10.xxx.xxx.19:5060;branch=z9hG4bK207e79b1ac5dd431d0cdd532dec7d3b;rport=5060

    From: "Francesca" <sip:41@10.xxx.xxx.1>;tag=2221630226

    To: <sip:41@10.xxx.xxx.1>;tag=674e3c5180;user=phone

    Call-ID: 2757463928@192_168_1_19

    CSeq: 5 BYE

    Contact: <sip:41@10.xxx.xxx.1:5060>

    User-Agent: pbxnsip-PBX/3.4.0.3201

    RTP-RxStat: Dur=16,Pkt=811,Oct=138244,Underun=0

    RTP-TxStat: Dur=16,Pkt=805,Oct=138460

    Content-Length: 0

     

     

    [9] 2009/10/05 10:00:45: Resolve 286: aaaa udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:45: Resolve 286: a udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:45: Resolve 286: udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:58: Resolve 287: aaaa udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:58: Resolve 287: a udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:58: Resolve 287: udp 10.xxx.xxx.20 5060

    [9] 2009/10/05 10:00:58: Resolve 288: aaaa udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:58: Resolve 288: a udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:00:58: Resolve 288: udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:01:02: Resolve 289: aaaa udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:01:02: Resolve 289: a udp 10.xxx.xxx.19 5060

    [9] 2009/10/05 10:01:02: Resolve 289: udp 10.xxx.xxx.19 5060

  25. Gigaset C470IP and Call transfer via R key

    in VoIP Phones

    Posted

    Hi,

    is anyone using these phones with pbxnsip??

     

    I'm able to transfer a call using "Options->External->Transfer", but it takes too much time.

    I'd like to use the "R" key, but all my attempts have been futile.

     

    I've also tried to do it using the "Hook flash" setup but I don't know what to enter inside the "Application Type" and "Application Signal" fields.

     

    Firmware version is 021910000000 / 043.00

     

    Could you help me?

     

    Here below you can see a pair of screenshots of the C470IP configuration page.

     

    c470ip1.jpg

    c470ip2.jpg

     

    Thank you.

     

    --

    Nicola

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