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Martyn

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Posts posted by Martyn

  1. Umm, so we've spent over £800 on snom ONE Yellow and there is no support which is nice to know. I've read on this forum previously that your pushing v5, so can I have it from the horses mouth your refusing to support a request for the v4.5 installer which you will have but won't supply?

  2. My recommendation is to always set the country code in the domain settings. That way, the PBX can re-format the numbers into always the same format. For example, 2121234567 and 12121234567 and +12121234567 are the same if the PBX knows that this number must be read in the US number context (country code 1). That makes it easier to match numbers, for example for hunt group alias names.

     

    In 5.0.10 we have introduced a drop-down for the settings that you are looking at. ERE are powerful, but not very easy to understand. In 5.0.10 the PBX would internally generate the pattern !([0-9]{10})$!\\1!500! to match the last 10 digits of the number. Then you can just assign the DID as the alias to the account number. For example, "400 2121111190" would mean that the account has the number 400 (which would be the primary number) and 2121111190.

     

    Hi,

     

    I must admit I'm really disappointed with the upgrade costs and pricing of the V5 PBX software, so much so, we're actively looking at other options. Also the support on this forum for pre v5 seems to recommend one thing... upgrade to V5. What happened to the support for customers who purchased v4, made a significant investment and now cannot get the answers they require?

     

    Anyway, stepping off my soapbox, the solution to my issue was very ease to resolve. I changed the firewall software on the PC which acts as the ipPBX and it all works correctly now.

     

    Martyn

  3. Hi,

     

    I am in the process of working out configuration of implementing DID's on a customer deployment with my own company PBX which is SnomONE v4.5.0.1090 Epsilon Geminids (Win64)using Snom 300's and 320's. The customer is using the same ipPBX software and phones.

     

    I have the following dial plan configured which picks up the last digit of a range of 5 DID's (0-4) and sends this to a Alias (1-4 against configured internal extension numbers (100-104). The DID's are in the format xxxx xxx xx90 - xxxx xxx xx94 i.e the last DID digit of 1 maps to Extension 101 etc. The DID with the last digit of '0' is for the hunt group telephone number (the main company telephone number).

     

    Dial Plan 9 is !([0-9]{1}$)!\1!t!500 - The Dial Plan is configured in the relevent Trunk 'Send to Extension'

     

    The issue is that when the hunt group extension is configured (the customer uses a hunt group 500 which has 3 stages, each stage ringing different phones in different order for differing periods of time) the call to the hunt group number works for the 1st attempt (it rings 2 phones as it is initially configured to do) but then subsequent DID calls do not ring the relevent phone extention unless the 500 is moved.

     

    Any guidance would be appreciated

     

    Regards

     

    Martyn

  4. LoL. Yes in theory the ALG should be transparent. But we had to find out that many SIP ALG vendors act after the motto "if I don't understand it, I'll block it". That is where TLS comes into play. TLS has the advantage that the ALG has no chance to see the SIP packet and start messing with it. The ALG does not have to patch the packets for snom ONE anyway, as the SBC does all neccessary steps to deal with devices behind NAT.

     

    Bottom like: Use PnP for phones as much as possible. If you use the automatic provisioning, the PBX will set the phones to use TLS anyway (well, for snom phones). Then the ALG will be taken out of the picture.

    Hi,

     

    Yes, agreed. I've even found the Vigor SIP ALG has different effects from one model to the next.

     

    I would use TLS, however the ITSP (Orbtalk) doesn't appear to support it. Its something i'll be asking them tomorrow, but when I configure it via the proxy address in the trunk interface configuration using '<sip gw ip address>:5061;transport=tcp' i get a 408 error.

     

    Regards

     

    Martyn

  5. If you want to answer the challenge from the firewall AND from the provider you obviously need two username/passwords. Right now, the PBX has only one. The multiple proxy authentication was specified in the RFC from day one; however I believe that only very few SIP implementations are able to deal with several username and passwords. Also, considering that many SIP registrations use TLS anyway, I don't see the security gain with this exercise.

    I'm not sure where you are coming from here, AFAIK the SIP ALG on the Vigor does not present any challenge back to the SnomONE as it is transparent in its operation, hence why i don't have the issue with my other customers PBX and also why I don't have any issue with it enabled on 2 x further Virtual IPPBX services we have deployed. An ALG should be transparent and not a 'full' proxy which would require usernames and password to operate. An ALG should only rewrite the header information to allow correct NAT transversal for datagrams that require it.

     

    Regards

     

    Martyn

  6. All,

     

    Has anyone come across the issue, where when you enable the SIP ALG in a firewall (in this case a Vigor 2920) you receive a 407 error on the handset when trying to make an outbound call?

     

    Trace with Vigor SIP ALG Enabled

    [8] 2013/02/01 00:51:53: Call from an user 107

    [8] 2013/02/01 00:51:53: From user 107

    [8] 2013/02/01 00:51:53: Call state for call object 22: idle

    [5] 2013/02/01 00:51:53: Dialplan "Standard": Match 9xxxxxxxxxxx@xxx.xx.x.xxx to sip:xxxxxxxxxxx@xxx.xxx.xxx.xxx;user=phone on trunk Orbtalk

    [8] 2013/02/01 00:51:53: Allocating for call port 93, SIP call id c004af05@pbx

    [5] 2013/02/01 00:51:53: set codec: codec pcma/8000 is set to call-leg 92

    [7] 2013/02/01 00:51:53: Call c004af05@pbx: Clear last INVITE

    [5] 2013/02/01 00:51:53: INVITE Response 407 Proxy Authentication Required: Terminate c004af05@pbx

    [8] 2013/02/01 00:51:53: Clearing call port 93, SIP call id c004af05@pbx

    [8] 2013/02/01 00:51:53: Remove leg 94: call port 93, SIP call id c004af05@pbx

    [8] 2013/02/01 00:51:54: Clearing call port 92, SIP call id 3c3b99ec3d46-gdx883ihv2qz

    [8] 2013/02/01 00:51:54: Remove leg 93: call port 92, SIP call id 3c3b99ec3d46-gdx883ihv2qz

    [8] 2013/02/01 00:57:13: Allocating for call port 94, SIP call id 3c3b9b2c6c24-epy3o4fuw20l

     

    Trace with Vigor SIP ALG Disabled

    [8] 2013/02/01 00:57:13: Call from an user 107

    [8] 2013/02/01 00:57:13: From user 107

    [8] 2013/02/01 00:57:13: Call state for call object 23: idle

    [5] 2013/02/01 00:57:13: Dialplan "Standard": Match 9xxxxxxxxxxx@xxx.xx.x.xxx to sip:xxxxxxxxxxx@xxx.xxx.xxx.xxx;user=phone on trunk Orbtalk

    [8] 2013/02/01 00:57:13: Allocating for call port 95, SIP call id 4e563542@pbx

    [5] 2013/02/01 00:57:13: set codec: codec pcma/8000 is set to call-leg 94

    [8] 2013/02/01 00:57:14: Call state for call object 23: alerting

    [7] 2013/02/01 00:57:15: Call 4e563542@pbx: Clear last INVITE

    [5] 2013/02/01 00:57:15: set codec: codec pcma/8000 is set to call-leg 95

    [8] 2013/02/01 00:57:15: Call state for call object 23: connected

    [8] 2013/02/01 00:57:19: Clearing call port 95, SIP call id 4e563542@pbx

    [8] 2013/02/01 00:57:19: Remove leg 96: call port 95, SIP call id 4e563542@pbx

    [7] 2013/02/01 00:57:19: Call 3c3b9b2c6c24-epy3o4fuw20l: Clear last request

    [5] 2013/02/01 00:57:19: BYE Response: Terminate 3c3b9b2c6c24-epy3o4fuw20l

    [8] 2013/02/01 00:57:19: Clearing call port 94, SIP call id 3c3b9b2c6c24-epy3o4fuw20l

    [8] 2013/02/01 00:57:19: Remove leg 95: call port 94, SIP call id 3c3b9b2c6c24-epy3o4fuw20l

     

    For reference, we use exactly the same firewall with SIP ALG enabled with a Linksys SPA9000 and it its rock solid and does not have the issue.

     

    Thanks in advance

     

    Martyn

  7. Hi,

     

    We have deployed Snom-One 4.5.1070 with Snom 320 handsets and are suffering from indiscriminate calls being dropped for no apparent reason. When this last happened I obtained the attached trace and the call dropped when the users handset 'appeared' to re-authenticate itself with the PBX which 'appeared' to be successful, but the only successful transaction that occured was the called was dropped.

     

    Could someone please assist in decoding the trace for user 107 and advising on possibilities on where configuration changes can be made to 'cure' this issue

     

    Thanks in advance

     

    Regards

     

    Martyn

    Dropped Call Trace.txt

  8. Hi,

     

    The issue is, although its comfort noise, my client (and his staff) find it uncomfortable as do their callers when they hear the blast of noise (which is described as loud and alarming) when they are placed on hold (before the MoH cuts in) or are being transferred.

     

    Are there any other comfort 'noises' available to choose from?

     

    I'll run with it switches off and guage my clients view of the difference

     

    Regards

     

    Martyn

  9. Katerina,

     

    Thanks. Now i've got the phones set up without SRTP and RTP Encryption switched off, we're still experiencing the white noise issue when the received is picked up and a number dialed. When the 'tick' is pressed, you hear a blast of white noise until the call is connected and the called number starts ringing, so the white noise is not consistent in time, it depends on how quickly the call is connected.

     

    Regards

     

    Martyn

  10. Katerina,

     

    I've now configured the phone back to admin mode and can configure the individual Identities.

     

    I've completed the upgrade to the latest 8.7.3.10 firmware on 3 of the phones, and they still display the issue noted in white noise when the 'tick' button is pressed to dial the selected number.

     

    For reference, i've just changed the original IPPBX (based on Asterisk 1.4) for the Snom ONE Yellow and the asterisk pbx/snom 320 combination didn't have the issue.

     

    Could you also advise on how to change the Phone pnp file to stop tls and RTP encryption being enabled by default. When ever I reboot the phone, I have to go back and change the settings in the phone

     

    Thanks in advance

     

    Regards

     

    Martyn

  11. Hi,

     

    It doesn't matter if we set the password up globally in the domain settings gui or locally in the account extension gui in the PBX, we still have the same issue. I've reset the phone numerous times and the issue still exists, and it only started when the phone firmware was upgraded.

     

    The remaining phones can be reset and they don't display the log on issue when accessed

     

    Regards

     

    Martyn

  12. Katerina,

     

    No, the password I set (and reset to default i.e. no password) would not get me into the advanced settings on either of the firmware revision levels. I set the password and on the actual extension tab in the PBX, i entered the admin username and password and still no access.

     

    I've attached a screen shot of what i'm seeing. I've tried the password and pin i've set and i don't get access to any of the tab settings.

     

    Thanks

     

    Martyn

    post-21660-0-04012100-1347559031_thumb.png

  13. Hi,

     

    I attempted to update from 8.4.18 to 8.7.x which failed. When the phone rebooted, I now cannot access any of the advanced settings, the only option I have presented to me is Administators Password, a text box to type into and a save button. I tried the same with a phone running 7.3.30 and its the same result.

     

    Any thoughts?

     

    Regards

     

    Martyn

  14. Hi,

     

    Thanks for the response.

     

    For info, I have upgraded the snom ONE from the initial install version of 4.2.0.3958 to 4.5.0.1090 all on Win 7 pro 64Bit

     

    8 of the snom 320's are using snom320-SIP 7.3.30.6059, the other phone has been upgraded to snom320-SIP 8.4.18 42566

     

    The upgraded phone displays exactly the same issue as the other phones. It did occur to me, that perhaps the 8-10 beeps are related to a voicemail being present.

     

    A single hunt group is configured as is a service flag for working hours. the only voice mail on the system is when the service flag is set and an out of hours message given and forwarded to a 'dummy' extension for out of hours voicemail. All 'real' extensions are configured for access to the 'dummy' extension's voicemail. When i say 'dummy', the extension has no physical phone configured.

     

    I'm happy to forward on the configuration, minus extension passwords and trunk details etc, as well as a configuration from 1 x snom 320 using 7.3.30 and the phone configuration from 1 x snom 320 using 8.4.18

     

    Regards

     

    Martyn

  15. Hi all,

     

    We've set up a Snom ONE Yellow with 10 snom 320 extensions for a client. Each extension has had RTP Encryption switched off, but on pressing the soft button for an external line, all user extensions hear white noise for a few seconds, with some of the phones beeping rapidily for 8-10 beeps before giving a stead dial tone.

     

    When the user picks up a call and transfers it to another phone, the caller hears a blast of white noise which apparently is very loud

     

    I've read that a few users on the forum have described the white noise issue and followed the advice but to no avail. My ITSP is Orbtalk in the UK.

     

    Thanks in advance

     

    Martyn

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