Martyn

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?

Hi, I have a customer who has installed a licenses version of snomONE Yellow 4.5.0.1090 and I need to build a standby PBX server in case the main unit fails. So my question is, how do I obtain a copy of the installer for Windows 7 please? Thanks Martyn

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

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

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

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

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

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

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

Katerina, You may well have nailed this one for me. When i listened to the file, it is the noise heard when making a call. I've renamed the file and re-started the pbx. I'll ask my customer to test tomorrow Regards Martyn

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

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

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

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

Katerina, I've already tried that, I set up a new admin password and pin in the pbx, rebooted but I still have the same issue. The phone asks for an Administrators Login and presents the save button I reverted to no admin password and pin, reset the handsets and still the same issue Regards Martyn