search for: snom320

...off ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tommaso Calosi Sent: Thursday, May 04, 2006 4:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unwanted conference with snom320 and asterisk 1.07bristuffed I have 13 Snom 320 with asterisk 1.07 bristuffed. The problem is that sometimes on random basis, when one customer is placed on hold and another call arrives, the customers are put in conference with each other. This look very strange to me, but I've disabled th...

...and another call arrives, the customers are put in conference with each other. This look very strange to me, but I've disabled the confernce button on the snom phones to prevent the human errors, but it still occurs. Investigating I've discovered that a similar problem was fixed with the Snom320 Release 5.2 (http://www.snom.com/snom320_release_notes.html ) It says: fixed unwanted conference bug in offhook/enter during ringback with an incoming call BUT my phones are already running 5.2 firmware. Any idea? Am I the only one with this problem? Do you think is the usual buggy-snom...

...0D CSeq: 341 REGISTER=0D=0D Max-Forwards: 70=0D=0D Contact: <sip:5549 at 192.168.0.4:2048;line=3Ddz9s1zzw>;flow-id=3D1;q=3D1.0;+s= ip.instance=3D"<urn:uuid:6467e491-3f54-4b76-8777-359712b5e388>";audio;mobil= ity=3D"fixed";duplex=3D"full";description=3D"snom320";actor=3D"principal";e= vents=3D"dialog";methods=3D"INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBS= CRIBE,PRACK,MESSAGE,INFO"=0D=0D User-Agent: snom320/7.1.8=0D=0D Supported: gruu=0D=0D Allow-Events: dialog=0D=0D X-Real-IP: 192.168.0.4=0D=0D Expires: 3600=0D=0D C...

Hi, I am using SNOM phones with Asterisks for few years. They used go occasionaly NR, and I would reregister them from the phone web interface. But it started doing frequently for last few months. Here are SNOM Phone and the firmware version; snom190-SIP - Version-Code: snom190-SIP 3.56m snom320-SIP - snom320 jffs2 v3.36 snom300-SIP - snom300-SIP 6.5.2 Asterisk version - Asterisk CVS-01/29/04-16:41:27 I would appreciate any help to fix the NR problem. Thanks

Hello all, I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Any clue would be welcomed. Thanks. Voip-Crazy

Hi there - I'm having some echo problems on my snom 320 phones. Anybody experience this before ? I don't have any issues with the sipura 841s I have though. Any help is greatly appreciated. Thanks ! Nora Lavelle -------------- next part -------------- An HTML attachment was scrubbed... URL:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director _____________...

...call with the im message as a variable in it starting a local channel that goes into the relevant part of the dialplan. examples: sms: "Could you please call Mark" "0612345678" for sending an sms sip_entry: "500" "cdwtg_34$" "ALAW" "snom320" for adding a sip entry to sip.conf with number 500 and password cdwtg_34$ for a snom320 conference: "0591234567" "0201234567" "0612345678: for setting up a conference between this numbers All the logic has to be in the dialplan or scripts but it all should sta...

Hi, I want to check the status of a blind transfer (only sip endpoint) between various phones. Transfer is working perfectly, using ## from features.conf or using transfer key from phone, here SNOM320. My problem is that if party to transfer to is busy, the transfer fail and the call is ended. What I want to do is to return the call to the party who originate the transfer. I checked variable like ${BLINDTRANSFER} ${TRANSFERED_BY} or ${TRANSFER_CONTEXT}, they are all empty. What did I miss?...

...s0vh;rport..From: <sip:K922002626 at 62.134.xxx.xxx>;tag=vyx8c0trgx..To: <sip:12 at 62.134.xxx.xxx>;tag=as13e7cb7c..Call-ID: 3c2768d8487f-rbzdwjzdbgcs..CSeq: 1163 SUBSCRIBE..Contact: <sip:K922002626 at 192.168.55.31:2048>;reg-id=1..max-forwards: 70. .event: dialog..user-agent: snom320/8.2.25..expires: 60..Accept: application/dialog-info+xml..Content-Length: 0.... U 2010/03/02 11:34:06.053870 192.168.4.109:5060 -> 192.168.55.31:2048 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport;received=212.78.xxx.xxx..From: <sip:K922002...

...ll. The Equipment my trials were done on: SPA3000 Hardware Version: 2.0.1(7376), Software Version: 3.1.10(GWd), and also tried Software 3.1.7. Nothing plugged into the FXS port. Asterisk 1.2.4 running on FreeBSD 5.4 (i386), AMD Athlon 64 3200+, 1GB RAM. SNOM 320. Application-Version: snom320-SIP 5.3.6 Rootfs: snom320 jffs2 v3.36 Polycom IP501 <don't have access to the software/hardware version from where I am right now> Cellphone All SIP equipment is running on a dedicated LAN. Network "splitters" were used to run two parallel LANs through the existing cablin...

...;subscribecontext=xxxxx" in the user's config in sip.conf so that SIP peers could only subscribe to extensions in that particular context. See the output below: asterisk_test*CLI> -- Registered SIP '959' at 82.135.81.211 port 5062 expires 3600 -- Saved useragent "snom320/4.4" for peer 959 Jan 6 20:11:29 ERROR[8677]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 008971940494 in context sanset Jan 6 20:11:29 ERROR[8677]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without...

We have the latest version of asterisk, on a xeon dell server (2gb ram), with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable firmware) and are having a few problems. We have a basic menu that transfers calls to different extensions. The problems can be found on all extensions. We have 2 different incoming providers and the problem happens on both prov...

Is anyone using the SNOM 360 as a reception console with Asterisk? We are trying to have the ability to view whether an extension is on or off hook, or ringing with the Snom, which seems to work fine. The issue is that we are having difficulty picking up calls and transferring. Anyone have experience / insight? Darrell S. Long Director of Technology BestWeb Corporation Phone 877-777-2932

Hi, I used Grandstream (gxp2000, gxp2020) and Snom (370) SIP Phones, but with 2 extensions BLF status does not work correctly. have someone ever tested a Sip Phone with more then 60 BLF without problems? Can someone suggest me model and brand? Thanks, bye. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...Addr->IP : 10.0.12.21:2067 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: Reception1 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw:20,alaw:20) Auto-Framing : No Status : OK (23 ms) Useragent : snom320/8.7.3.19 Reg. Contact : sip:blabla0 at 10.0.12.21:2067 As you see, Def. Username should be tel-221 and not Reception1 ! We have around 20 SNOM300 and 320, only with this one we have this problem. Firmware version is the same in other SNOM320. We restart Asterisk, the server himself, the phon...

...quest 102: Match Found ...... Apr 23 12:42:39 DEBUG[6850]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3c26700f6ddd-eeq2u57j5kuf@snom320-00041324ABE7' of Request 112: Match Found P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE] Apr 23 12:42:43 DEBUG[7625]: channel.c:3275 ast_generic_bridge: Didn't get a frame from channel: mISDN/1-1 Apr 23 12:42:43 DEBUG[7625]: channel.c:3550 ast_channel_bridge: Bridge stops bridging channels mISDN...

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Probation passed

...gy To: "[ MyName in the Snom ], " <"[ MyName in the Snom ], >;tag=as6fec2de7 Call-ID: 055f1d8f752fcd8b52f0f3b71f89ef36@[ MyName in the Snom ].dyndns.org CSeq: 2 BYE Max-Forwards: 70 Contact: <sip:44@[ I P - A N G E R U F E N E R ]:5060;line=7anx8ofw>;reg-id=1 User-Agent: snom320/7.3.14 Content-Length: 0 As you can see - a BYE is sent. I tested it out many times, it only occures if a call comes from the patton, only sip calls can greatly be holded and transferred. The whole SIP DEBUG is available here, i dont wanted to post this stuff.. ( http://www.agethen.com/sip-de...

...: 3c28a76e73cf-gp9nioi8zdci CSeq: 12 REGISTER Max-Forwards: 70 Contact: <sip:test3 at 192.168.114.200:2048;line=nagqjxpx>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:8d4403c1-af97-4139-b535-de7d460a9187>";audio;mobility="fix ed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/8.4.18 Allow-Events: dialog X-Real-IP: 192.168.114.200 Supported: path, gruu Expires: 3600 Content-Length: 0 <------------...