Re: [asterisk-users] Vocera Comm Badges
I think you would only need a headset if you need privacy. That is the catch-22 in healthcare environments -- HIPAA would prevent Dr. Smith calling nurse Jones to ask the temperature of patient Susan White because nurse Jones could be with another patient. Support for the Vocera server to communicate via a PRI to a PBX has long been in the software. I'm not sure what they use for signaling. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von Klitzing Sent: Saturday, July 24, 2010 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vocera Comm Badges Hi! I've worked with these before. They are designed to run a whole hospital shift, so there should be no worries regarding the battery. Sounds good. The speaker phone quality is acceptable (the speaker is quite small and points forward, not upwards in the direction of the ear), or would you rather recommend a headset? I'm not aware of the server having any kind of SIP support -- I think you would need to have a PRI trunk to another PBX. That appears to be a newly added feature. The last time I talked to them they had their own proprietary codec to deal with the occasional packet loss of Wi-Fi, and the codec was encrypted, too. Interesting - any clue on the signalling method? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vocera Comm Badges
I've worked with these before. They are designed to run a whole hospital shift, so there should be no worries regarding the battery. I'm not aware of the server having any kind of SIP support -- I think you would need to have a PRI trunk to another PBX. The last time I talked to them they had their own proprietary codec to deal with the occasional packet loss of Wi-Fi, and the codec was encrypted, too. They really work -- just hit the button and say Call Bob Smith and they call Bob. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von Klitzing Sent: Friday, July 23, 2010 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vocera Comm Badges Hi! I´ve seen them at trade shows, I think I remember it being proprietary. What about using Dect handsets? That Star Trek device has always interested me. Too bad they chose WiFi over DECT, though. Vocera badge: * WLAN b/g * Talktime 2-2.5 hours, standby 20-27 hours * headset jack * OLED display (why don't they ever show this on the pics) * Linux based Back side: http://farm4.static.flickr.com/3412/3496101115_32319840a5.jpg System: * Windows server * Nuance ASR biometrics Dictaphone * Dialogic T1/ISDN/Analog cards * SIP interface * iPhone app I am also curious to hear some user reports, in particular about battery performance: Does it last one entire working day, or does it need to be exchanged (once?) per day? What is the smallest feasible installation? What speech protocol does the badge use to talk to the server? First quarter of 2010 was the third consecutive profitable quarter for the company. In addition, since the beginning of 2010, the company has announced the addition of 22 new employees across North America and the United Kingdom. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
It's been a few years ago, but Network Computing had tests results showing that VoIP over a VPN was measurably better than outside a VPN. Why? Because the latency was low enough that lost UDP packets (within the VPN tunnel) could be re-transmitted before the jitter buffer had expired. Since most jitter buffers are on the order for 10 to 80 msec, if your one-way latency is any greater than a third of your jitter buffer, it's of no use. For example, if the one-way latency is 15 msec, the best-case scenario is that with single-time packet loss, the other packet would arrive at the destination in ~45 msec. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Garth van Sittert Sent: Friday, May 08, 2009 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] QoS VPN I would think that VoIP over VPN is a bad idea as UDP packets need to be in realtime not corrected by the TCP of the VPN. Garth van Sittert Technical Director BitCo 08600 24826 www.bitco.co.za Aurimas Skirgaila wrote: Despite the VPN overhead, running VOIP through VPN is good idea because VPN reorders encapsulated UDP packets in correct order. Security matters as well. I'd suggest to route VNC packets rather over internet than VPN (so do I), as VPN usually has the highest priority. On Thu, May 7, 2009 at 11:33 PM, Roberto Piola roberto.pi...@visiant.it mailto:roberto.pi...@visiant.it wrote: I do not have examples, but if you are using the 1700 series router in order to originate the ipsec vpn, you may use command qos pre-classify (please search for it on cco.cisco.com http://cco.cisco.com) On Thu, May 7, 2009 at 9:54 PM, Brent Davidson br...@texascountrytitle.com mailto:br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mobile centrex solution
Two of the wireless carriers have a Centrex-like solution: http://www.networkcomputing.com/channels/wireless/showArticle.jhtml?articleI D=202200832pgno=5 Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort Sent: Tuesday, March 17, 2009 3:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] mobile centrex solution anyone know of a solution where mobile handsets out roaming the pstn cellular network can be used and treated as full fleged centrex extentions, i.e. I can transfer a call that comes in on a wired centrex copper pair out to a cell phone and the cell phone can transfer the call back or vice versa where the cell phone recieves the call directly and can transfer to the office all without hairpinning the call? essentially when the call is transfered I'd like to have asterisk get out of the call path but still have the capability to transfer the call back to asterisk and it's attached office phones. Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WiFi SIP phone w/VPN?
Not in the form factor that you would expect. Can I ask why? Most modern VoFi phones support WPA2. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, February 11, 2009 5:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] WiFi SIP phone w/VPN? Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN abilities? Failing that, a WiFi phone that runs Linux? I already know one phone that does meet my requirements -- the iPhone. The new software comes with a Cisco VPN client, and a SIP client can be had from third-party vendors for jailbroken phones. And, while I'm not averse to the idea, a) it ain't cheap, and b) it's a bit hack. I've googled my heart out, but haven't found anything else that (I'm sure) meets all three requirements. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
You're the miracle worker! Thanks! Frank From: Andres [mailto:and...@telesip.net] Sent: Tuesday, January 06, 2009 11:19 AM To: Frank Bulk Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Frank Bulk wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend Your problem is you are trying to do authenticate by host and by username at the same time. That does not work in asterisk. You should be seeing a Warning message in the console saying something like: check_auth: username mismatch, have ACME, digest has username That means you already matched to sip.conf entry ACME, but the digest has a different username, so it fails. You can fix it by setting the paramters in the CS1500 to have the username = ACME. That way the digest will come in as: Digest username=ACME ...bla bla bla Andres http://www.telesip.net I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which SIP/2.0 401 Unauthorized is issued after the un-authenticated INVITE and SIP/2.0 403 Forbidden after the authenticated INVITE. When I add insecure=very, this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: sip:5551236...@172.16.10.40 sip:5551236...@172.16.10.40 It isn't very clear (to me) from the success how the insecure=very helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Frank Bulk - iName.com wrote: The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: sip:5552022...@172.16.10.40 sip:5552022...@172.16.10.40;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b ba4 To: sip:+15552027...@sip.acme.com sip:+15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40;user=phone sip:5552022...@172.16.10.40;user=phone Privacy: none Remote-Party-ID: sip:5552022...@172.16.10.40;user=phone sip:5552022...@172.16.10.40;user=phone; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: sip:5552022...@172.16.10.40 sip:5552022...@172.16.10.40 Authorization: Digest username=username,realm=asterisk,nonce=118af2b0,uri=sip:+15552027020 @ sip.acme.com,response=111e63ec2a1f3ebabefe4f7dae4087a1,algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
After many hours of fiddling around, Andres gave me the final piece. For those looking to implement SIP Trunks on a CS-1500 with Asterisk, here are the pieces: Diagram: CS-1500 -- customer PBX (172.16.10.40)(172.16.10.195) HOST: should be the DNS name assigned to the CS-1500's SIP interface. e.g. sip.acme.com NUSR: user name used for the CS 1500 to login into the customer PBX. Needs to match up FreePBX's Trunk Name. For those who use the CLI, this section in sip.conf is encased in square brackets. i.e. [customername] NPSW: password used for the CS 1500 to login into the customer PBX. Needs to match up with the secret= line. i.e. secret=password IP: IP address of the customer PBX. i.e. 172.16.10.195 LUSR: user name used for the customer PBX to login into the CS 1500. Needs to match up with the username= line. i.e. username=customername LPSW: password used for the customer PBX to login into the CS 1500. Needs to match up with the secret= line. i.e. secret=password. For simplicity we made NUSR/LUSR the same and NPSW/LPSW the same. Since you need to define a trunk per customer, it makes the most sense and it easiest to support and implement. Here's what you need to add to Asterisk's sip.conf (yes, just those few lines!) [customername] host=sip.acme.com type=friend username=customername secret=password And the CS-1500 output: TYP TG NUM 1234 TGTP 2WAY TGNM SIP MG NO SIGT SIP STSI 0 HNPA 555 RC 0 RTP 0 TRNL PRFX PRFX 24 APFX NONE TRFC NONE 4XCD YES ACKA NO TYPC NOCO NXX UNKN LATA 000 CMCT NO TGID NONE SIT NO CNAR NO LRN NONE TNDM NO LDAT NO TRFC NONE EOAT NO ATIC NO CMCO NO TGMU NO HOST sip.acme.com NUSR customername NPSW password IP 172.16.10.195 PORT 5060 PROT UDP T38F NO AUTH YES LUSR customername LPSW password CLIM 7 CPBY 0 Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Bulk - iName.com Sent: Monday, January 05, 2009 6:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: sip:5552022...@172.16.10.40;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db ba4 To: sip:+15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40;user=phone Privacy: none Remote-Party-ID: sip:5552022...@172.16.10.40;user=phone; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: sip:5552022...@172.16.10.40 Authorization: Digest username=username,realm=asterisk,nonce=118af2b0,uri=sip:+15552027020@ sip.acme.com,response=111e63ec2a1f3ebabefe4f7dae4087a1,algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=audio 36224 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: sip:5552022...@172.16.10.40;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db ba4 To: sip:+15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40;user=phone Privacy: none Remote-Party-ID: sip:5552022...@172.16.10.40;user=phone; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: sip:5552022...@172.16.10.40 Authorization: Digest username=username,realm=asterisk,nonce=118af2b0,uri=sip:+15552027020@ sip.acme.com,response=111e63ec2a1f3ebabefe4f7dae4087a1,algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=audio 36224 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
