voipguy

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

My Grandstream HT502's came today and the bug is not in the HT502's = great

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.

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.

....correction to my posted problem. When the incoming call waiting call is ringing my HT701 I reported that my current call is muted which is correct but only on my side is it muted - the person I was speaking with can still hear me but I can't hear them until the call waiting caller gives up or goes to voicemail.

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.

snom One - can you explain more on that limitation? I don't get why we would be limited to X calls for one domain.

No, they are not in an agent group.

me too, we have over a hundred users on our system with spa2102 adapters using analog phones and we need the *93 to remain.

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.

Don't disable the mailbox after 3 failed attempts - just disconnect the caller after 3 failed attempts.

Can you post the release notes?

Thanks!

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

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.

...or build it into the web interface so we can manage call recordings from Sales q's etc.

thanks but I'm already running that version, the posts above show version .5016 so I was asking if a build could be done for that version but for centos 32 bit.

can you make a centos 32bit pbxnsip build?

...here's a pic of my graph from Sunday nights email report - you can see the difference because no speaker phone calls. That graph represents about 900 calls for Sunday.

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.

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.

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

Maybe try new snom firmware 8.4.32 - I know it fixed a couple issues I had.

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.

Thanks. Wow...tons of bug fixes and new features in this release.