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voipguy

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Posts posted by voipguy

  1. SNOM M3 Base Station power adapter

    in snom Phones

    Posted

    Hi,

     

    Does anyone know where I can buy a replacement power adapter for the SNOM M3 base station? Not the cordless phones cradle but the actual main SNOM M3 base station? It doesn’t even have to be a SNOM power adapter as long as it’s the right power specs and the same shaped end so it will plug into the SNOM M3 base station.

     

    Thanks

     

     

     

     

  3. Delta Aurigids

    in New Features/Versions

    Posted

    I see 0 messages for those users.

     

     

    I'm not using the new version but after reading this thread I would like to throw out a possible reason why this is happening.

     

    The new version now allows you to play recorded calls in the user login gui - the customer didn't have 25 voicemail msgs from people calling him but does he have "user" or "domain" level call recording turned on? If he does then it's my guess the new version has a bug in it where it's counting the users call recordings in the system as voicemail messages and the voicemail max messages limit he was only allowed to have was 25 so thats why he got the email saying he reached his max limit but in fact has no vmails in his box.

     

    I don't have the new version so I can't tell if the user would even be able to see the call recordings in his user login gui because it might be permission based and he might not have permission to see them and thats why he's reporting no messages in his web login gui?

     

    Thats just a guess.

  4. Grandstream HT701 - incoming call waiting call mutes original call

    in ATA Devices

    Posted

    UPDATE

     

    The Engineers over at Grandstream said this:

     

    "I managed to reproduce the issue not only on a PBX that provides early media, but also on other PBXs that do not. I have filed this bug and it should be fixed soon. We will keep you posted on this and let you know when we have a build ready for you to test. Thank you."

     

     

    Earlier this morning they said after looking at my pcap logs that it might be an early media problem - looks like they confirmed this.

     

    The HT701 is one of their newer models...I have some HT502's coming tomorrow which has different firmware, I'll test and report if it has the same bug.

  6. Grandstream HT701 - incoming call waiting call mutes original call

    in ATA Devices

    Posted

    Hi,

     

    We are using Centos, PBXNSIP Hosted Pro + ver 4.3.0.5022

     

     

    Using a Grandstream HT701 ATA:

    The call waiting doesn't work like it should - I could be on a call talking and then I receive a call waiting beep and I can also see the incoming caller id but while this call waiting call is ringing my HT701 I can no longer hear or speak with the person I was originally speaking with - soon as the call waiting caller gives up(or call goes to voicemail) and stops calling I can now hear and speak with my original person....we never got disconnected just muted while the call waiting caller was trying to call my HT701.

     

     

    This is my setup:

     

    1. HT701 registered to a PBXNSIP account - the account uses a dial plan - the dial plan uses it's own trunk.

     

    2. The trunk is setup as SIP Proxy, inbound/outbound, generic sip server, username and password for the account in my sip proxy server, proxy ip address, lock codec = no, strict rtp = no, generate unique = no, accept redirect = no, interpret sip uri = yes, requires busy tone = no, trunk requires out of band = yes, global=no, RFC3325(P-Asserted-Identity), no failover, is secure=no, inter-office=no, ringback =media, force local=no.

     

     

    I can make and receive calls and the call quality is excellent, dtmf works, caller id works.

     

    Here's the weird part - I can bypass PBXNSIP and just register my HT701 to my voip proxy server with the exact same settings untouched and everything works 100% correctly - the ring pattern is normal and even the call waiting beep doesn't mute my conversation with my original party.

     

    So one would think the problem is a setting in PBXNSIP that needs to be tweaked and that might be the case but all my Cisco SPA2102 and PAP2-NT's work with PBXNSIP - no problems - all the trunk settings are the same.

     

    I do have a wireshark trace of the HT701 registered to PBXNSIP with the above call example and a trace with the HT701 registered to my voip proxy server with the above call example. I also have a PBXNSIP log level 8 file for the HT701 registered to PBXNSIP call. I'm not knowledgeable enough to see any problems in these pcap trace files. If any of the SNOMONE techs can take a look at them I would appreciate it - I can PM the pcap files or PM a link for you to download them.

     

    If anyone else has come across this call waiting problem before please post.

     

    Thanks again.

  10. 4.5 ready for primetime?

    in General Setup

    Posted

    Dont be fooled by the name "Beta Corondidis". This is not a beta version, this is the name of a comet. We are happy with 4.5 so far; the biggest hickup was the upgrade of the trunks, where some manual work might be required with the trunk settings.

     

     

    We are still on PBXNSIP ver 4.3.0.5022 centos 32 bit. We run hosted and have hundreds of Trunks - the above comment is the reason why we are waiting to upgrade. Hoping the SNOM team comes out with an update that defaults the Trunks to the way they have been in all previous versions.

     

    We don't have time right now to manually go into each Trunk and configure them.

  13. v2011-4.3.0.5020 is here

    in New Features/Versions

  14. v2011-4.3.0.5020 is here

    in New Features/Versions

    Posted

    All,

     

    It's been 4 months since we last released a bugfix version on v4 for snomONE. So, here is latest bugfix version - 2011-4.3.0.5020. This includes fixes for some of the known defects and couple of minor feature addition.

     

    The release notes, download links can be found here

     

    http://wiki.snomone.com/index.php?title=Release_notes

     

    Please let us know if you wanted any specific version that is not listed here.

     

    Thank you for the support!!!

     

     

    Can you post a PBXNSIP linux centos 32 bit version? Thanks.

  15. Phone screen

    in Web Interface

    Posted

    There was an issue with the LDAP query in snomONE version. To avoid this issue you can clear the entry(389) from Admin->Settings->Ports: LDAP section and restart the PBX.

     

    Otherwise, you can wait till next week for the updated version.

     

    Are you still planning on releasing the updated versions this week? Looking for PBXNSIP centos 32 bit, would love to run 64bit but not sure if the G729 call recording bug has been fixed yet.

  20. MOS Score

    in General Setup

    Posted

    Does this only happen on the 3xx series? Our MOS graph is looking very similar, sometimes even more worse and I never figured out what causes the system to behave like that on an internal Cat6 Gbit LAN with 870 devices...

     

    Really looking forward to the day when Snom changes the graphs to something like jqPlot which will allow to chose a datapoint from the graph and it will display the relevant data of....*dreams*... B)

     

    EDIT: See attachement.

     

    It happens to all VoIP phones when your in speaker phone mode. We also have a mix of phones - snom 320, 370, 821 and 870's and ata's like spa2102's.

     

    There is nothing wrong with your system - it's just speaker phone calls. It took me awhile to figure this out.

     

    From what I've seen ver 5 will have a MOS graph for each extension - in the account under the registration tab a MOS graph will be at the bottom. This will be great for when customers complain about call quality you can then look at the graph for their extension. I also think their will be a MOS graph on the trunk account.

     

    I attached a pic of my MOS graph - we are a VoIP hosting provider so we use G729 - that is why you see the line around 3.5....G711 the line is around 4.1 in the graph. We support G711 to avoid transcoding so you will see a few calls at the 4.1 mark. The lower marks are speaker phone calls.

     

    We do both business and residential....on the weekends for the graphs you never see the calls go below the 3.5 mark because my residential customers don't have voip speaker phones - only spa2102 ata's.

     

    Hope this helps clear this up for you.

  21. MOS Score

    in General Setup

    Posted

    Hi,

     

    I have one PBX with 10 extensions, all snom 320s with 8.4.31 installed, that has been showing me a MOS graph with all the trunks great but a few spots on the extensions around 2.5. See the attached image. Does this mean that I have an extension acting up or some heavy LAN traffic? The PBX and Phones are on the same LAN. There is a POTS adapter involved, which is the only interface to the outside world, it is also on the same LAN. The PBX version is .4025 running on Ubuntu. Any help or a direction to look in would be appreciated.

     

    Thanks,

    Steve

     

     

    I bet that is from users using their snom 320's in speakerphone mode....it's easy to test....make a 5 minute speaker phone call to a friend and talk for 5 minutes but do this after hours so you know the call in the graph was your speaker phone call.

  22. issue with HTML Templates - userpage control

    in General Setup

    Posted

    Hi,

    the editor in the HTML template page does not load the entire template, it cuts off after certain number of characters.

    For example, the snom_820_phone.xml, it cuts off after some of the multicast address fields, even though there are much more after it (by looking at the generated config file).

     

    It makes it impossible to overwrite some of the settings.

     

    Pls take a look and let me know of any workaround or fix.

     

    thank you

     

    Try the Chrome web browser and I think you will find you won't have this problem.

  24. No organization to the call log list

    in CDR

    Posted

    Having trouble figuring out how they organize the call list..? The user GUI that I am seeing is not organized at all, it is skipping days. How can you list them in order by day?

     

    I reported this as a bug over a year ago...created a ticket...sent screen shots...gave them access to the end user web account but nothing was ever done. When i login to an end users account and look at the missed calls tab call log or the regular call log it's all random. When I login to my system as admin the call logs are in order.

  25. thundering loud burst of static for several seconds

    in General Setup

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