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I did set the default codec to g711u on pirelli end to g711u. Can it have anything to do with alert info Ring tones, the only other thing I did other than upgrade was played with the custom4 ringtone in the ringtones.xml file.

Is anyone using the Pirelli wifi phone with version 2.1.6? Mine stopped working about the time of the 2.1.5 to 2.1.6 upgrade. Outbound calls work. however inbound it rings, however if I try to answer it does not answer it just goes back to the idle screen. If the call is picked up elsewhere e.g. at another phone logged in to the same extension, the Pirelli will continue ringing even after the call is pick up, until I click the ignore button on the phone. Here is the pbxnsip log [0] 2008/02/20 16:01:38: SIP Rx tls:192.168.0.28:2181: PRACK sip:18@192.168.0.10:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.0.28:2056;branch=z9hG4bK-71609n4h8czk;rport From: "Shana " <sip:18@localhost>;tag=rms1z38m2o To: <sip:10@localhost;user=phone>;tag=2ad886fee1 Call-ID: 3c27bd535d71-enyv17r10bz3 CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:18@192.168.0.28:2056;transport=tls;line=2ahorl0j>;flow-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons Content-Length: 0 [0] 2008/02/20 16:01:38: SIP Tx tls:192.168.0.28:2181: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.28:2056;branch=z9hG4bK-71609n4h8czk;rport=2181 From: "Shana " <sip:18@localhost>;tag=rms1z38m2o To: <sip:10@localhost;user=phone>;tag=2ad886fee1 Call-ID: 3c27bd535d71-enyv17r10bz3 CSeq: 2 PRACK Contact: <sip:18@192.168.0.10:5061;transport=tls> User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Length: 0 [0] 2008/02/20 16:01:38: SIP Rx tls:192.168.0.38:2182: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.10:5061;branch=z9hG4bK-537f737a2cb8049fae1eeff3d7cb7145;rport= 5061 From: "Shana " <sip:18@localhost>;tag=58061 To: "Nechama " <sip:10@localhost>;tag=gkd9r3qljt Call-ID: c98e4a77@pbx CSeq: 5798 PRACK Contact: <sip:10@192.168.0.38:2065;transport=tls;line=2ahorl0j>;flow-id=1 Content-Length: 0 [5] 2008/02/20 16:01:38: Final transport on 290540 [0] 2008/02/20 16:01:38: SIP Rx udp:192.168.0.24:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-81df691e26fdb3082513d8e5c060f700;receiv ed=192.168.0.10;rport=5060 From: "Shana " <sip:18@localhost>;tag=27826 To: "Nechama " <sip:10@localhost> Call-ID: a073c51e@pbx User-Agent: Pirelli D910.0.3.99d FS_D910.0.2.83_PIS CSeq: 11786 INVITE Content-Length: 0 [0] 2008/02/20 16:01:39: SIP Rx udp:192.168.0.24:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-81df691e26fdb3082513d8e5c060f700;receiv ed=192.168.0.10;rport=5060 From: "Shana " <sip:18@localhost>;tag=27826 To: "Nechama " <sip:10@localhost>;tag=3-3516497182 Call-ID: a073c51e@pbx User-Agent: Pirelli D910.0.3.99d FS_D910.0.2.83_PIS CSeq: 11786 INVITE Contact: <sip:10@192.168.0.24:5060;transport=udp> Require: 100rel RSeq: 1 Content-Length: 0 [0] 2008/02/20 16:01:39: SIP Tx udp:192.168.0.24:5060: PRACK sip:10@192.168.0.24:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-7b39b4283864b823e4f0a37ebf9ba2ba;rport From: "Shana " <sip:18@localhost>;tag=27826 To: "Nechama " <sip:10@localhost>;tag=3-3516497182 Call-ID: a073c51e@pbx CSeq: 11787 PRACK Max-Forwards: 70 Contact: <sip:10@192.168.0.10:5060;transport=udp> RAck: 1 11786 INVITE Content-Length: 0 [0] 2008/02/20 16:01:39: SIP Rx udp:192.168.0.24:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-7b39b4283864b823e4f0a37ebf9ba2ba;receiv ed=192.168.0.10;rport=5060 From: "Shana " <sip:18@localhost>;tag=27826 To: "Nechama " <sip:10@localhost>;tag=3-3516497182 Call-ID: a073c51e@pbx User-Agent: Pirelli D910.0.3.99d FS_D910.0.2.83_PIS CSeq: 11787 PRACK Content-Length: 0 [0] 2008/02/20 16:01:40: SIP Rx udp:192.168.0.24:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-81df691e26fdb3082513d8e5c060f700;receiv ed=192.168.0.10;rport=5060 From: "Shana " <sip:18@localhost>;tag=27826 To: "Nechama " <sip:10@localhost>;tag=3-3516497182 Call-ID: a073c51e@pbx User-Agent: Pirelli D910.0.3.99d FS_D910.0.2.83_PIS CSeq: 11786 INVITE Supported: timer,100rel,replaces Session-Expires: 1800;refresher=uac Min-SE: 90 Contact: <sip:10@192.168.0.24:5060;transport=udp> Content-Type: application/sdp Content-Length: 229 v=0 o=Pirelli 2208988817 4 IN IP4 192.168.0.24 s=Pirelli c=IN IP4 192.168.0.24 t=0 0 m=audio 50000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [0] 2008/02/20 16:01:40: SIP Tx udp:192.168.0.24:5060: ACK sip:10@192.168.0.24:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-b1d883fa9a395d9a51e085f90585c908;rport From: "Shana " <sip:18@localhost>;tag=27826 To: "Nechama " <sip:10@localhost>;tag=3-3516497182 Call-ID: a073c51e@pbx CSeq: 11786 ACK Max-Forwards: 70 Contact: <sip:10@192.168.0.10:5060;transport=udp> Content-Length: 0 [0] 2008/02/20 16:01:40: SIP Tx udp:192.168.0.24:5060: BYE sip:10@192.168.0.24:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-79e3ddb8ac76c9f068b7d4ad6b14e6ad;rport From: "Shana " <sip:18@localhost>;tag=27826 To: "Nechama " <sip:10@localhost>;tag=3-3516497182 Call-ID: a073c51e@pbx CSeq: 11788 BYE Max-Forwards: 70 Contact: <sip:10@192.168.0.10:5060;transport=udp> RTP-RxStat: Dur=3,Pkt=0,Oct=0,Underun=0 RTP-TxStat: Dur=6808121,Pkt=0,Oct=0 Content-Length: 0 [0] 2008/02/20 16:01:41: SIP Rx udp:192.168.0.24:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-79e3ddb8ac76c9f068b7d4ad6b14e6ad;receiv ed=192.168.0.10;rport=5060 From: "Shana " <sip:18@localhost>;tag=27826 To: "Nechama " <sip:10@localhost>;tag=3-3516497182 Call-ID: a073c51e@pbx User-Agent: Pirelli D910.0.3.99d FS_D910.0.2.83_PIS CSeq: 11788 BYE Content-Length: 0 [5] 2008/02/20 16:01:41: BYE Response: Terminate a073c51e@pbx

I double checked this. I snom 360 phones configured using plug and play. There are 2 address book buttons, one the 3rd button under the display, that does not show any entries. The other the directory button under the redial button shows all the internal extensions, however it does not show enties from the personal or company address book.

Did you try http://www.videolan.org/streaming-features.html ?

I always set service to auto restart. I did that the first time the service crashed. I am not sure if I ever had a service crash since, as it would start right back up, before the complaints come in.

I also got several complaints about blank / hang up messages. I also got a few that voicemail volume is very low.

Using an IVR node instead of Auto attendant did the trick.

I agree Video support is becoming a must, everyone is asking about it.

If anyone has a quick fix or workaround for this, I would greatly appreciate it. Thanks,

No Not the MWI, the retrieve message botton, is forwarding to the wrong extension. A reset of the phone to defaults will definetly do the fix, however it should not need to be done, there is something wrong. DND automatically gets programmed to the correct *code when using plug and play, and it does genarally work to undo DND too, however ocassionally it does not undo it.

Snom 360 7.1.30, I can not get a trace as it happens so randomly, only 1 user complained, and it only happens 1-2 times per week.

I have 1 user complaining, that from time to time, when he grabs a caller off hold, she can not hear the other party, although the other party hears her fine. There is no NAT involved.

There is a ringtones.xml, however I am not sure how to use it. I saw a sample in another post. the bellcore 2 3 and 4 sound very similar, the snom standard ringers sound very different.

I have a station that hot desked to another extension, then turned off hot desk and everything went back to nornal except, the message button, still forwards to the hot desked extension. I also get many complaints, when users use the DND button on the snom, then turn it if, it sometimes stays on in the PBX, they need to disable using the * code or web interface.

I see there are several custom options on the ring tones, where do I set or upload these ring files. Can I set it to use ringer 2 or ringer 3 from snom?

I was able to reproduce this problem on my internal demo system.

Is there any way to sync the snom phones address book with the PBXnsip address book.

Upgraded to 2.1.6.2446 and the problem still exists. 4# redirects to auto attendant 04, however after 3 seconds, it cuts off and gives message please enter extension number. see attached.

Thanks, that worked, removing the tel: from the alias name made the trunk DID display.

I do have the DID in the trunk DID field, however the extension settings seem to override that. version 2.1.5.2357 (Win32)

Whichever option I try for the caller ID on the trunk setting. I either get the extension number or the tel:alias. I need the outbound caller ID to be the trunk DID.

Broadvox requires all outbound calls to display the mail number as the Caller ID in order to be included in their unlimited local plan. This worked fine until we set a tel: alias on an extension, now that an alias was set for an extension, that DID is being displayed on outbound calls. Is there any way to have the trunk DID overwrite the the tel: alias for outbound caller ID presentation.

Most 3rd Party that I have seen were designed for Asterisk http://www.voip-info.org/wiki/view/Asteris...nagerInterfaces or were released by PBX vendors for their own PBX such as Quadro Communications Manager www.epygi.com.

Were any 3rd party operator consoles tested with PBXnSip?

Just an update, if I change the outbound proxy to the IP instead of east.ga.broadvox.net it works, should it not query DNS, I do receive that IP when I ping it. I rather use DNS name as it does change as the have redundant systems.