hosted

m3 is a snom red headed step child

We will be supporting G.723 sip trunks end of January 2011.

add ulaw codec to your IP phone. then it wont transcode which is good.

We use a honeypot application at various IP's throughout our network to find scanners. most scanners first send an OPTION message to see if your SIP port is open. our honeypot detects this and processes a block to our core router. This has been very effective at stopping unwanted traffic before it becomes an outage issue.

WHAT????? Is this crippled on pbxnsip keys as well? I would certainly hope not.

yea makes sense. I think they are just trying to put something in a box and thought distribution..

you can buy Yellow version and get 4 non-snom registrations. Thats still a great value.

this seems pretty simple to me. snomONE is a subsidized product buy the purchase of a SNOM IP telephone. much the same way cisco sells a discounted PBX with the use of a cisco IP telephone. IF you want to use 3rd party devices and dont take advantage of a subsidized product that is easy as well. buy PBXNSIP which is open to all devices. sure the software is the same. its just not subsidized.. All we sell is snom phones, because of the tight integration. For this is is great! now we can give a free (subsidized) product and sell more phones..

pbxnsip re-registered 1/2 of what i set it to. so if I set the sip server to 360 seconds pbxnsip re-registers at 180. is that normal?

actually fax is much better in 2010 it redirects the fax to a fax server.. which means it works now

how does DNSnaptr work with Polycom PNP? is that a SRV or TXT record?

I have the trunks set to re-register every 10 minutes, but it keeps doing it every 30 seconds.. how to I change this?

It would be super cool if in the calls status page you could display the codec the call is using. and if it is transcoding.

our hosted server backups are not even 500 meg. hard to think a single pbx would be even close. be curious what OS and hardware also. never heard of these issues.

I requested this a long time ago but PNP per domain would be awesome

but how do i disable the phone from sending it. I need DND for this application to stay local (to the phone).

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I mapped DND in the buttons and then took it off. however the phone keeps sending these. </DoNotDisturbEvent> [5] 16/9/2010 14:34:42: DND-Value: true [5] 16/9/2010 14:34:42: Switch on DND [5] 16/9/2010 14:34:44: rtp_port::set_port [5] 16/9/2010 14:34:44: DND xml: <DoNotDisturbEvent xmlns="http://www.ecma-international.org/standards/ecma-323/csta/ed3"> <device>200@burr.sipnerd.com</device> <doNotDisturbOn>false</doNotDisturbOn> Even though the snom says Event Key F_DND how do i disable that?

yes it is stuck in ALERTING. linksys phones as well.

Curious.. If I put a DNS/SRV record in this field "Explicitly list addresses for inbound traffic" will it allow traffic from the all the SRV records? or do I just need to put in all the IP's..

INVITE sip:603@192.168.10.55:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 6.5.4.3:5060;branch=z9hG4bK-f6dad48abfa80aab2b7ebc47d6a55afa;rport From: "Chris" <sip:601@6.5.4.3>;tag=830686032 To: "Yesica" <sip:603@6.5.4.3> Call-ID: b67a76fa@pbx CSeq: 26368 INVITE Max-Forwards: 70 Contact: <sip:603@6.5.4.3:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: nexSIP99/4.0.1.3499 Answer-Mode: Auto Content-Type: application/sdp Content-Length: 298 v=0 o=- 1197571218 1197571218 IN IP4 6.5.4.3 s=- c=IN IP4 6.5.4.3 t=0 0 a=oa:offer m=audio 16686 RTP/AVP 0 18 101 a=rtpmap:0 pcmu/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv

yea we loaded it. Aastra intercom/paging still does not work. linksys is ok.

Answer: Yes

Can I get an updated file that makes Aastra auto-answer/intercom function work?

when you call to a hunt group and answer the BLF and the web status says the phone is alerting. V4.x