[asterisk-users] trying to connect to asterisk with softphone (logs, etc)
In the Asterisk log I see: --- [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] <--- SIP read from UDP:198.38.7.34:5065 ---> SIP/2.0 200 OK To: ;tag=sd3D4swKRc From: ;tag=as07c833c5 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK13c68eb7;rport Call-ID: 5e070a0021f200c72308ddad6fe2521c@192.168.0.99 CSeq: 221 REGISTER Contact: ;expires=55 Content-Length: 0 <-> [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] --- (8 headers 0 lines) --- [Mar 23 19:25:29] NOTICE[4067] chan_sip.c: Outbound Registration: Expiry for nat5.babytel.ca is 55 sec (Scheduling reregistration in 40 s) [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] Really destroying SIP dialog '5e070a0021f200c72308ddad6fe2521c@192.168.0.99' Method: REGISTER [Mar 23 19:25:44] VERBOSE[4003] asterisk.c: [Mar 23 19:25:44] -- Remote UNIX connection [Mar 23 19:26:01] VERBOSE[15640] manager.c: [Mar 23 19:26:01] == Manager 'sendcron' logged on from 127.0.0.1 [Mar 23 19:26:01] VERBOSE[15640] manager.c: [Mar 23 19:26:01] == Manager 'sendcron' logged off from 127.0.0.1 [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] <--- SIP read from UDP:192.168.0.28:5060 ---> REGISTER sip:192.168.0.99 SIP/2.0 Call-ID: d7ab3099e71b65e0ae104cc441aecc25@0:0:0:0:0:0:0:0 CSeq: 4 REGISTER From: "201" ;tag=5fbdd638 To: "201" Max-Forwards: 70 User-Agent: Jitsi2.6.5390Mac OS X Expires: 600 Contact: "201" ;expires=600 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK-34-5e76c348412aa7cadf05777dd72d8a4d Authorization: Digest username="201",realm="asterisk",nonce="2577db3d",uri="sip:192.168.0.99",response="c3c4a08638f1ac928b1329b312038e75",algorithm=MD5 Content-Length: 0 <-> [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] --- (12 headers 0 lines) --- [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] Sending to 192.168.0.28:5060 (NAT) [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] <--- Transmitting (NAT) to 192.168.0.28:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK-34-5e76c348412aa7cadf05777dd72d8a4d;received=192.168.0.28;rport=5060 From: "201" ;tag=5fbdd638 To: "201" ;tag=as78b94599 Call-ID: d7ab3099e71b65e0ae104cc441aecc25@0:0:0:0:0:0:0:0 CSeq: 4 REGISTER Server: Asterisk PBX 1.8.29.0-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="43b1ba24" Content-Length: 0 <> [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] Scheduling destruction of SIP dialog 'd7ab3099e71b65e0ae104cc441aecc25@0:0:0:0:0:0:0:0' in 32000 ms (Method: REGISTER) [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] <--- SIP read from UDP:192.168.0.28:5060 ---> REGISTER sip:192.168.0.99 SIP/2.0 Call-ID: d7ab3099e71b65e0ae104cc441aecc25@0:0:0:0:0:0:0:0 CSeq: 5 REGISTER From: "201" ;tag=5fbdd638 To: "201" Max-Forwards: 70 User-Agent: Jitsi2.6.5390Mac OS X Expires: 600 Contact: "201" ;expires=600 Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK-34-364fb1c68f6d21e3f71292e300535c15 Authorization: Digest username="201",realm="asterisk",nonce="43b1ba24",uri="sip:192.168.0.99",response="ed23dc12d2effb6d02d5c7aa33a260d5",algorithm=MD5 Content-Length: 0 <-> [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] --- (12 headers 0 lines) --- [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] Sending to 192.168.0.28:5060 (NAT) [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] <--- Transmitting (NAT) to 192.168.0.28:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK-34-364fb1c68f6d21e3f71292e300535c15;received=192.168.0.28;rport=5060 From: "201" ;tag=5fbdd638 To: "201" ;tag=as78b94599 Call-ID: d7ab3099e71b65e0ae104cc441aecc25@0:0:0:0:0:0:0:0 CSeq: 5 REGISTER Server: Asterisk PBX 1.8.29.0-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <> [Mar 23 19:26:04] NOTICE[4067] chan_sip.c: Registration from '"201" ' failed for '192.168.0.28:5060' - Wrong password [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] Scheduling destruction of SIP dialog 'd7ab3099e71b65e0ae104cc441aecc25@0:0:0:0:0:0:0:0' in 32000 ms (Method: REGISTER) [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] <--- SIP read from UDP:192.168.0.28:5060 ---> REGISTER sip:192.168.0.99 SIP/2.0 Call-ID: d7ab3099e71b65e0ae104cc441aecc25@0:0:0:0:0:0:0:0 CSeq: 6 REGISTER From: "201" ;t
Re: [asterisk-users] [OT] switches
On 23/03/15 16:37, thufir wrote: > On Mon, 23 Mar 2015 10:11:54 +, Lukasz Sokol wrote: > >> No, ethernet switch works at lower / physical / MAC layer, NAT is >> 'above' >> that; >> so as long as everything is OK with your TCP/IP settings everywhere, >> a switch is entirely transparent to TCP/IP (or generally, when it's >> encapsulated into MAC traffic). > > > so how does a client pc find the server if there's no NAT? by IP > address?? That makes no sense, to me, if the switch isn't assigning > addresses. > > > -Thufir > > +1 to what Kevin said, and there is a protocol running on pretty much every ethernet based network, named ARP : Address Resolution Protocol, by which ALL the clients learn ALL the surrounding clients (including the one that is the GATEWAY) MAC/IP combinations. Simplified, the encapsulation of ethernet packets is sort-of | MAC Header | IP Header | Packet |[MAC Source address][MAC Destination Address]|[Source IP][Destination IP]|[The rest of packet] [order and number of fields not necessarily real-life, for illustration purposes only] now the MAC source/dest fields are added AND REMOVED as needed when the packet passes from card to computer/router, then from computer/router to card; as the MAC fields don't make sense in wider area networks; 'dumb' switches don't participate/snoop in ARP, only store a table of what card MAC address they encountered on source MAC field of packets coming from that interconnect manageable switches /can/ participate and filter in the ARP process if told so and have such option. HTH, el es -- _ -- 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
[asterisk-users] Auto Answer
Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone "Allow Auto Answer by Call-Info: yes Dialplan: exten => 501,1,SIPAddHeader(Call-Info: answer-after=2) exten => 501,n,Page(SIP/140&SIP/110,d) exten => 501,n,Hangup() not work for me, it ring but does the function of auto answer Any idea? -- rickygm http://gnuforever.homelinux.com -- _ -- 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] [OT] switches
On Mon, 23 Mar 2015, thufir wrote: so how does a client pc find the server if there's no NAT? by IP address?? That makes no sense, to me, if the switch isn't assigning addresses. The 'endpoint' (pc, softphone, mobile, desk set, etc.) 'finds' the server's IP address when: ) You configure the endpoint with the IP address or host name of the server. This happens either by a web page you fill out on the endpoint or a configuration file that is downloaded by TFTP, FTP, HTTP, etc. ) You configure SRV records in your DNS. I think the old IAXy did some sort of discovery on port , but I don't remember if it was device or server discovery. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] [OT] switches
> so how does a client pc find the server if there's no NAT? by IP > address?? That makes no sense, to me, if the switch isn't assigning > addresses. Switches have a MAC table that keeps track of which MAC addresses are on which ports. That's how they decide where to route packets. http://en.wikipedia.org/wiki/CAM_Table http://en.wikipedia.org/wiki/OSI_model-- _ -- 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] [OT] switches
On Mon, 23 Mar 2015 10:11:54 +, Lukasz Sokol wrote: > No, ethernet switch works at lower / physical / MAC layer, NAT is > 'above' > that; > so as long as everything is OK with your TCP/IP settings everywhere, > a switch is entirely transparent to TCP/IP (or generally, when it's > encapsulated into MAC traffic). so how does a client pc find the server if there's no NAT? by IP address?? That makes no sense, to me, if the switch isn't assigning addresses. -Thufir -- _ -- 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] PJSIP - Video Support for WebRTC
On Mon, Mar 23, 2015 at 8:55 AM, Gosmac wrote: > Hey i have an interesting topic to discuss here. > > The main goal here is to be able to make a video call between two WebRTC > endpoints registered on asterisk 13 it is a feature that definitely asterisk > 13 should support . > > the problems that i faced with this is the following and i hope i could get > an advise here. > > asterisk 13 vanilla version has some issues marking the video packets this > complain web browser specially VP8 codecs so a friend of mine help me to > patch res_rtp_asterisk and now asterisk is marking video streams :) it just > mark video packets not touch anything else and web browser show video on web > page now I’m using online demo http://tryit.jssip.net/ is stable and get more > updates than sipml5. so i try echo() dialplan test and everything work > perfect on echo test :). > > i have two questions and i hope you could give me some advise. > > 1) after marking video packet I’m able to make Dial() between two webrtc > peers but i get one way audio and video on callee party, “after 3 minutes on > call” i get two way audio and video on all parties seems to be not just a > problem on a missing keyframe. > > 1.1) the 3 minutes delay only happen using chrome stable , could be a dtls > problem when asterisk make an offer to other endpoint? > 1.2) when i use chrome-dev and i disable dlts encryption everything work > perfect on video call. > > 2) after marking video packets i realize that when you make a call with video > and you involve on dialplan an application like playback or music on hold any > application that played audio files (audio and video never work). > > 2.1) asterisk is muggling the audio and video streams ? > > This is good information for all guys out there that wants to support video > on webrtc in asterisk 13 > Please stop spamming the list with this e-mail. Resending it multiple times is clearly not yielding the results you'd like. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- 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
[asterisk-users] Question about hangup - Asterisk v11.15.0
Hello,
on previous versions of asterisk, extension h and H make us know who
ended a call (caller or callee). In the last * versions, seems that only
h extension is used, as stated here
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
In the last versions, how do we know which end terminate a call (SIP,
ISDN, Analog, ...) in h extension ? Will the
${HASH(SIP_CAUSE,${CDR(dstchannel)})} give the information ?
We also face a strange behavior: we are ringing few phones (~10) and
sometimes, once the call get answered, we see that 2~3 seconds after
this, music on hold is started on the channel! And 20 seconds after, the
call is terminated without that any party hanged up :-(
It's a Elastix 2.5 installation, we thought that problem could came from
Elastix so we set our own dialplan for incoming calls:
same =
n,Set(__phonesToRing=SIP/118&SIP/119&SIP/122&SIP/123&SIP/124&SIP/125&SIP/126&SIP/127&SIP/128&SIP/129&SIP/130&SIP/132)
same = n(startRing),Answer()
same = n,Dial(${phonesToRing},,it) ;no voicemail
or forward => ring indefenitely
same = n,Hangup
Incoming call give for instance in logs:
[2015-03-23 11:07:20] VERBOSE[1342][C-0e85] app_dial.c: --
SIP/126-43d8 is ringing
[2015-03-23 11:07:21] VERBOSE[1342][C-0e85] app_dial.c: --
SIP/118-43d3 connected line has changed. Saving it until answer for
SIP/bero_trunk-43d2
[2015-03-23 11:07:21] VERBOSE[1342][C-0e85] app_dial.c: --
SIP/118-43d3 answered SIP/bero_trunk-43d2
[2015-03-23 11:07:25] VERBOSE[1342][C-0e85] res_musiconhold.c:
-- Started music on hold, class 'default', on SIP/bero_trunk-43d2
[2015-03-23 11:07:27] VERBOSE[1342][C-0e85] res_musiconhold.c:
-- Stopped music on hold on SIP/bero_trunk-43d2
[2015-03-23 11:07:41] VERBOSE[1342][C-0e85] pbx.c: -- Executing
[h@from-trunk:1] Macro("SIP/bero_trunk-43d2", "hangupcall,") in new
stack
Thanks for any hint
--
Daniel
--
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[asterisk-users] Local channel + queue
hi, i'm facing problem with multiple calls to one agent when Local channels are used wireshark shows multiple invites to the agent's phone used versions asterisk 1.8/asterisk 13 agents are logged dynamically. interface state based on hints queue configuration ... ringinuse=no autofill = yes ... member => Local/99@route_phones_1,2,mila_jojovich,SIP/virtual_99 member => Local/88@route_phones_1,3,angelina_jolie,SIP/virtual_88 time between call to local channel and call to SIP device can be in seconds //Without local channel queue works good, but i need local channel for additional settings/actions i need working BLF (multiple states was problem), i need working transfers (cannot limit call to 1 via GROUP_COUNT) howto change state in queue immediately after calling local channel (similiary to after calling sip device) ? any tips? -- --- Marek Cervenka === -- _ -- 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] RTP sent to remote internal IP
Harel Cohen wrote: Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the public IP address as it is seen on the packet header. Signalling is flowing correctly with no issues. Could you please advise why is this happening and how to correct this? Here is the [peer] in my sip.conf and the SDP in the setup (INVITE + OK). I'll be happy to provide any other information if needed: I think before anyone can fathom a guess you'd need to include Asterisk level information. Such as SIP debug on its side, rtp debug. As well - have you opened the firewall so RTP can be received at the Asterisk that advertises the public IP? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
[asterisk-users] PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams :) it just mark video packets not touch anything else and web browser show video on web page now I’m using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :). i have two questions and i hope you could give me some advise. 1) after marking video packet I’m able to make Dial() between two webrtc peers but i get one way audio and video on callee party, “after 3 minutes on call” i get two way audio and video on all parties seems to be not just a problem on a missing keyframe. 1.1) the 3 minutes delay only happen using chrome stable , could be a dtls problem when asterisk make an offer to other endpoint? 1.2) when i use chrome-dev and i disable dlts encryption everything work perfect on video call. 2) after marking video packets i realize that when you make a call with video and you involve on dialplan an application like playback or music on hold any application that played audio files (audio and video never work). 2.1) asterisk is muggling the audio and video streams ? This is good information for all guys out there that wants to support video on webrtc in asterisk 13 Javier Riveros -- _ -- 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] [OT] switches
Remember that that zyxel 16 port switch is only 8 poe ports. If your phones are 802.3af or 802.3at, you could look at the ubiquiti line of switches. On Mar 13, 2015 9:34 PM, "Brian Franklin" wrote: > If your phones support PoE, > > I have had huge success with Zyxel: > http://www.amazon.com/ZyXEL-ES1100-16P-16-Port-Ethernet-Unmanaged/dp/B00 > 5GRETMM/ref=sr_1_3?ie=UTF8&qid=1426296572&sr=8-3&keywords=zyxel+poe > > If you want to go even cheaper, I have successfully used these as well: > http://www.amazon.com/TRENDnet-8-Port-100Mbps-Switch-TPE-S44/dp/B000QYEN > 1W/ref=sr_1_10?ie=UTF8&qid=1426296706&sr=8-10&keywords=poe+8-port > > > Brian Franklin > NTG, Inc. - "Problem Solved" > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of thufir > Sent: Friday, February 20, 2015 1:58 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] [OT] switches > > Pardon, this might be off-topic. I'm reading: > > http://en.wikipedia.org/wiki/Network_switch > > For a setup of ~5 agents, would I be wrong in thinking that a generic 16 > port unmanaged switch would fit the bill? > > The first model to come up for me in an Amazon search is: > > http://support.netgear.com/product/fs116 > > > > Is this a reasonable choice? Would I be wrong in thinking that most any > Fast Ethernet switch would be fine for Asterisk? > > > > thanks, > > Thufir > > > -- > _ > -- 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 > > - > This email was processed through Xeams to filter junk messages. > If you feel this message has been tagged incorrectly, you can > change its category by clicking the link below. > Click here > http://spam.ntginc.net:5272/FrontController?operation=mbeu&f=1_-150_ > 20150220_1643960.eml&chkBayesian=1&pr=1&mt=1&ma=s to mark email as junk. > - > > -- > _ > -- 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] how asterisk detects silence?
On Mon, Mar 23, 2015 at 12:25 AM, Dmitry Melekhov wrote: > 19.03.2015 09:31, Dmitry Melekhov пишет: >> >> Hello! >> >> As I see there is dsp_drop_silence switch in confbridge. >> Could you tell me how asterisk detects silence? >> Is it possible to change silence level, >> so, let's say some not loud enough background noises will be recognized as >> silence >> and only loud enough human voice will be recognized as sound? >> >> Thank you! >> Asterisk passes received voice data, converted to signed linear, through a DSP. The DSP looks at the energy level in the data and, if it is above a certain value, categorizes the data as 'silence' or 'talking'. You can tweak the periods necessary for Asterisk to decide if someone is talking or silent using the 'dsp_silence_threshold' and 'dsp_talking_threshold' settings: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- 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] [OT] switches
On 22/03/15 03:03, thufir wrote: > On Fri, 13 Mar 2015 20:33:13 -0500, Brian Franklin wrote: > >> If your phones support PoE, >> >> I have had huge success with Zyxel: >> http://www.amazon.com/ZyXEL-ES1100-16P-16-Port-Ethernet-Unmanaged/dp/B00 >> 5GRETMM/ref=sr_1_3?ie=UTF8&qid=1426296572&sr=8-3&keywords=zyxel+poe >> >> If you want to go even cheaper, I have successfully used these as well: >> http://www.amazon.com/TRENDnet-8-Port-100Mbps-Switch-TPE-S44/dp/B000QYEN >> 1W/ref=sr_1_10?ie=UTF8&qid=1426296706&sr=8-10&keywords=poe+8-port >> >> >> Brian Franklin NTG, Inc. - "Problem Solved" > > > This is the router/modem gateway the ISP supplied: > > http://www.cisco.com/web/consumer/support/modem_DPC3825.html > > When I connect one of these switches to the router, that doesn't create a > double-NAT problem? > No, ethernet switch works at lower / physical / MAC layer, NAT is 'above' that; so as long as everything is OK with your TCP/IP settings everywhere, a switch is entirely transparent to TCP/IP (or generally, when it's encapsulated into MAC traffic). All that happens at a level totally transparent to the TCP/IP stack In a way, an Ethernet Switch is /the/ network near you, your cables are 'just' interconnects. HTH, el es > > thanks, > > Thufir > > -- _ -- 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
