Re: [asterisk-users] asterisk for small home phone system
On Thu, Oct 25, 2012 at 11:33:06AM -0700, Matthew Hixson wrote: >Is there any reason a regular old voicemodem wouldn't work? IME the voice quality and reliability are pretty grotty. If you find one that works, great! R -- _ -- 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] asterisk for small home phone system
On Thu, Oct 25, 2012 at 11:09:01AM -0700, Matthew Hixson wrote: > - Is the Linksys SPA3102 a good piece of hardware for this type of setup or > is there something cheaper? Perhaps a card that can go right into the Linux > box? I'm using an OpenVox A400 (with an FXO module), which Asterisk can drive directly. > - Would we configure our SIP clients on our iphones to login directly to > Asterisk running on my home Linux box? I have 18MB/2.5MB internet service > with a static IP so this wouldn't be a problem. That would be the simplest approach (modulo firewalls). If you already have another SIP provider, you could configure your home asterisk to forward calls to that... R -- _ -- 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 to stop ringing when incoming PSTN call is answered externally?
On Tue, May 22, 2012 at 11:32:19PM -0400, ft...@mindspring.com wrote: >The calls are routed just fine, but when a call is answered at one of >the extensions or externally (by a home telephone) the asterisk >extensions continue to ring one more time. Is there a way to have >Asterisk drop an incoming PSTN call as soon as it's answered? I have the same problem, and earlier discussion here suggests it's insoluble. -- _ -- 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] using Wifi smartphones as SIP clients
On Mon, May 07, 2012 at 09:14:36PM +0200, Hans Witvliet wrote: >Hope that these are better that the utstar F1000: >Keep on re-chargibg as battery is empty in no-time, and security is >lousy; just wep, no wpa. WPA and WPA2. Battery lasts about a day in dual mode, much longer in 2G-only of course. And at UKP30 they may be worth a punt even if you end up upgrading to something else. -- _ -- 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] using Wifi smartphones as SIP clients
On Mon, May 07, 2012 at 12:03:17PM +0200, Bart Coninckx wrote: >What about phones like the Unidata WPU-7800 ( >http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have >experience with those? Would these also suffer from connection >losses? I've been using a UTStarcom GF-210 for the last year and more as my personal phone - dual-mode 2G GSM and SIP/802.11. Sound quality on SIP is slightly better than 2G, getting it to talk to Asterisk is no problem at all, but certainly if you're moving from one wifi device to another you will get dropped calls. If that's your use case, it's going to be that way whatever hardware you use - I haven't seen any implementations of 802.11F or 802.11r in the field. -- _ -- 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] Asterisk 1.8.12.0-rc1
On Thu, Apr 12, 2012 at 01:14:25PM -0700, motty.cruz wrote: >Can this be acomplish? I hope I explained better. Yes, no problem. First, get the two servers talking to each other (I like IAX for this, but SIP also works). If NAT is a concern, there are various ways round it (I like VPN tunnels). Then set up the dialplan on the public server to route the call to the other machine. -- _ -- 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] Asterisk 1.8.12.0-rc1
On Thu, Apr 12, 2012 at 11:04:22AM -0700, motty.cruz wrote: >Hello All, >Is it possible to have an Asterisk server connect to a 2nd Server using >Extension? >For instance I have an Asterisk Server with public IP address then I have a >2nd Asterisk server in the local network that I want to do intercom pagin >with this server can I connect this server as extension of the main >Asterisk Server? I'm not sure what you mean by "using Extension", but you can certainly route calls between the two with as much complexity as you desire. (For example, I run a public server to receive calls presented as SIP/IAX, and a private server connected to IP-phones and a POTS card, with a VPN tunnel between them.) -- _ -- 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] MessageSend, SIP, and call files
On Tue, Apr 10, 2012 at 11:50:40AM -0500, Danny Nicholas wrote: >This is what "core show applications" in 10.1.3 shows > SendDTMF: Sends arbitrary DTMF digits > SendFAX: Sends a specified TIFF/F file as a FAX. > SendImage: Sends an image file. > SendText: Send a Text Message. > SendURL: Send a URL. >You are using sendtext - you might want to use sendurl instead. Those are all about sending data in an existing channel, though - the trick is that I don't _have_ a channel, which is presumably why MessageSend exists. Is there a way to set up a channel without ringing the phone? -- _ -- 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] MessageSend, SIP, and call files
As I've occasionally posted here before, I have user terminals which can
accept SIP text messages to an SMS-like interface.
After upgrading to Asterisk 10, I do indeed have external processes
generating these messages. But it's a bit ugly. What I'd _like_ to do is
simply generate a callfile, and something like this almost works:
Channel: Local/8902
Application: MessageSend
Set: MESSAGE(body)=messagebody
Data: sip:glowworm
Data: sip:glowworm
but (a) I need that reserved local number to let the call work at all
(the number just does an Answer(), Wait(10), Hangup) and (b) I can't
seem to set the sender's name. That ought to be the second Data
parameter; actually the second one seems to determine where the message
goes, and whatever I set the first one to the sender name always comes
up as "asterisk". (Specifically, in the packet capture, I have
From: "asterisk"
.) Now, I _can_ achieve the desired result, but only by having _another_
local number that does
exten => 8901,n,SET(MESSAGE(body)=${msg_out_body})
exten => 8901,n,MessageSend(${msg_out_to},${msg_out_from})
and setting up the callfile with:
Extension: 8901
Set: msg_out_to=glowworm
Set: msg_out_from=
at which point the message will appear to originate from FROM (note that
if I put a display name component in the msg_out_from it gets ignored -
but that is the terminals' peculiarity). But that's ugly. Has anyone got
this working with a relatively straight callfile setup?
While I'm writing, does Asterisk 10 have any way to send a SIP message
that isn't text/plain?
Roger
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Re: [asterisk-users] Standard UIDs, especially for asterisk?
On Tue, Nov 15, 2011 at 04:42:05PM +, Tony Mountifield wrote: >But it sounds like it is distro-specific. No, it's system-specific. Debian for example will assign UIDs out of the relevant range based on the order in which packages are installed. Just use the textual UID/GID values, not the numeric ones. Roger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to asterisk IAX trunk
On Tue, Oct 11, 2011 at 02:53:26PM +0100, Jonathan Archer wrote:
>How can I get the 5 to stay where it is so that lookups work correctly?
>is it part of the outbound CID?
My trunking (prefix 9 to get trunk access from either side of the link)
includes things like:
exten => _9NX.,1,Set(CALLERID(num)=9${CALLERID(num)})
exten => _9NX.,n,Dial(IAX2/remoteserver/${EXTEN:1},,wW)
R
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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
On Mon, Sep 12, 2011 at 12:21:06PM -0600, linux guy wrote: >FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have >graphical tools. To add to what everyone else has said: if you _really_ need to run a graphical tool on the server, you can always ssh -X into it without having to have a full desktop installed there. (As for wireshark: tcpdump on site, then bring the capture file home to analyse with wireshark. Works for me...) Roger -- _ -- 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] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: >Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the "rename" syscall. Might be worth shelling out to your system's mv command. R -- _ -- 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] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
>shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
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Re: [asterisk-users] Answering machine answers after pickup a phone.
On Fri, Aug 05, 2011 at 10:59:03AM +0200, Jorge Barreiro wrote: >What I try to do is that, when there is an incoming call from the ouside, if >someone answers on a phone, then the PBX won't answer. I have a couple of VoIP phones fed through Asterisk, as well as analogue phones linked directly to the line. In this case, picking up the analogue phone stops the VoIP phones ringing (after ten seconds or so). I don't know whether this would be achievable with the Asterisk console and soundcard drivers... Roger -- _ -- 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] Re : Direct RTP with Asterisk
On Sun, Jun 19, 2011 at 01:40:31PM +0100, Sagbo Romaric wrote: >No, I can't, because, it's a different NAT. I try to simulate P2P with >asterisk. >What you suggest to me ? I like VPN tunnels. They give you a flat network topology and decent security. -- _ -- 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] click to call
On Fri, Jun 17, 2011 at 05:20:39PM +, salaheddine elharit wrote: >i want to use sip 223 in order to call phone number Is that meant to be the originator or the destination? Channel: gets the originator; Extension: gets the destination. Roger -- _ -- 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] Audio dropping
On Fri, May 27, 2011 at 10:31:57AM +0200, Mark Scholten wrote: >What could the reason be audio in 1 direction is dropping? (Normally from >the Asterisk server to the mentioned SIP clients.) No clear information is >in the logs (it is like the call ended normally) and not all calls are >having problem (most not, but it happens to often for us to start using VoIP >more at the moment). While the most usual problem is packet filtering / NAT, this generally manifests as no audio at all in one direction, not a drop in mid-call. But it's possible that one of the intermediate transit providers is doing something "clever". (Disabling ping, as you mention in your later email, is often a good indicator of a company with insufficient Clue.) Are you in a position to tunnel the traffic over a VPN or similarly flat and unfilterable network link? (This might be a good idea anyway.) Roger -- _ -- 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] UK English sounds packs
On Thu, May 26, 2011 at 02:09:21PM +0100, Ishfaq Malik wrote: >Does anyone know if there are any free UK accented English sounds packs? I use: http://www.enicomms.com/cutglassivr/ -- _ -- 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] Skype-like dialing from web page
On Tue, May 17, 2011 at 01:30:33PM -0400, Mike wrote: >Is there any softphone or TAPI plug-in that allows one to dial from a web >page? As you may know, Skype has a mechanism that converts phone numbers on >a web page to a click-to-dial application. I'd like to use this but on a >normal softphone (Bria, Xlite, other). Generate a callfile, setting Channel to point to the softphone (e.g. SIP/Xlitephone) and Extension to point to the number you want to dial. (You'll need to specify Context too.) When the callfile is processed, the softphone will ring; when it's picked up, it will dial the far end. Roger -- _ -- 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] Voicemail Configuration
On Mon, May 09, 2011 at 03:00:19PM -0600, John Marvin wrote: >However, I want to record what is "said" during that time and send it >to a third voicemail box once the caller hangs up without having >pressed 1 or 2. You could use Monitor to record the whole call, then use an AGI to do something with it on hangup if the other conditions haven't been satisfied...? -- _ -- 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] Best Scripting Language
On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote: >Can anyone suggest which is the best scripting language for Asterisk or any >telecom device? Depends on the other parameters. Perl is great for rapid development, but I wouldn't run it per-call on a box taking hundreds of calls per second. (Ditto Ruby and Python.) C will be much faster, but it's more effort to write and debug. -- _ -- 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] Dialplan help: hang up incoming call and call the number back
On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote: >Is there a better way of handling the post-hangup >processing? Callfiles? -- _ -- 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] Detect DTMF tone during call?
On Sat, Feb 26, 2011 at 03:08:02AM -0600, Dan Saul wrote: >I am attempting to create a intercom buzzer system using asterisk as a >back end. Most is figured out except the actual action of buzzing the >door. I need to detect whether a DTMF key was pressed by the the >called party (the resident). Is this possible to do using just a >dialplan? I can't see any options on the Dial command that would lead >to this, am I looking in the wrong place? I looked briefly through the >archive and I heard mentions of AGI, is this what must be used to >accomplish this? If you want it to be detected within a call, which is what I'd assume, you'll probably be looking at the applicationmap section within features.conf. http://www.voip-info.org/wiki/view/Asterisk+config+features.conf -- _ -- 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] Carrying context from one server to another?
The relevant part of my setup is something like: SIP phones -> local server -> remote server -> SIP-to-PSTN provider I want _some_ of the SIP phones on the local server to be able to get access to SIP-to-PSTN, but not all of them. The local-to-remote connection is IAX2 over VPN. Do I need to set up two separate IAX2 connections, one "privileged" and the other not, or can I somehow tag calls from some phones on the local server so that they're noted as privileged on the remote server? -- _ -- 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] Unknown calls
On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote: >Still last night there was a call to a customer. Plz help me figure out the >solution for this problem. Can you be sure that the call _is_ coming through your Asterisk server, rather than being the result of random scanning for your customers' phones? -- _ -- 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] On-Hold Music
On Tue, Feb 15, 2011 at 09:01:16AM -0600, Danny Nicholas wrote: >Thanks for the tip - got a "Norwegian translator" for "uhort.no"? Anything wrong with http://translate.google.com/translate?js=n&prev=_t&hl=en&ie=UTF-8&layout=2&eotf=1&sl=no&tl=en&u=uhort.no ? R -- _ -- 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] On-Hold Music
On Tue, Feb 15, 2011 at 08:39:57AM -0600, Danny Nicholas wrote: >Good suggestion, Roger, but this seems like a "slippery slope" path. >Today's podcaster could be tomorrows ASCAP/BMI member coming back for you? Doesn't matter if you use music that has been explicitly released as royalty-free (usually under a CC licence or similar). The URLs I gave are resources _for_ podcasters; sorry I didn't make that clearer. R -- _ -- 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] On-Hold Music
On Fri, Feb 11, 2011 at 04:37:49PM -0600, Danny Nicholas wrote: >In 500 words or less (if possible), please explain what is a >legal music-on-hold file? One source of explicitly royalty-free music is the podcasting community: http://uhort.no/ and http://www.podsafeaudio.com/ both have extensive libraries. -- _ -- 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] Using files .call or AMI
On Mon, Feb 14, 2011 at 04:06:10AM +, Edwin Quijada wrote: >How would be the dialplan for this context from-lan ??? This list is for non-commercial support. If you want someone to do the work for you, I suggest you go elsewhere and offer money. -- _ -- 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] Using files .call or AMI
On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: >This works for me.! but the agent has to dial the number ? >How could be the context for do this ? U can give an example ? I'm using this to place calls from local IP-phones over the PSTN. So my script will generate, say: Channel: SIP/lanphone Context: from-lan Extension: 08001234567 taking the 0800... from the list of customer details. SIP/lanphone is the ID of the "originating" phone. Extension is the sequence the agent would dial if he were placing the call himself. The "originating" phone rings; when it's picked up, the Asterisk server calls the "Extension" number and bridges the two calls, so the local agent hears ringing tones from the far end. All the agent has to do is pick up the phone when it rings and put it down when the call is over. -- _ -- 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] Using files .call or AMI
On Sat, Feb 12, 2011 at 04:23:00PM +, Edwin Quijada wrote: >I have a webpage with information about a customer so in this page the agent >click a phone number and asterisk do the call and transfer the call to agent >if this call is answered. Usually it's the other way round: the agent's phone rings, and when he picks it up the other end gets dialled. That's trivial with call files: Channel: (local channel ID for agent) Context: (context for calling local channel) Extension: (remote party's phone number) -- _ -- 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] meetme conference & playback of random sound file
On Thu, Feb 10, 2011 at 04:58:05PM -0800, John Jolly wrote: >i am trying to configure the meetme conference (asterisk 1.8) to play a * >random* sound file from a specific directory prior to it dropping the caller >into the conference itself. Absent an Asterisk-specific solution, how about a separate process which would link a random file into a fixed pathname? (Fired off from cron, perhaps.) Roger -- _ -- 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] SIP MESSAGE outside calls - state of the art?
I have a mobile phone (UTStarCom GF-210) that uses SIP MESSAGE to send
"SMS" messages over VoIP. My Asterisk 1.4 installation drops these
messages and returns a failure condition to the phone:
[Feb 9 10:17:22] WARNING[11960]: chan_sip.c:9859 receive_message: Received
message to from "Display Name"
;tag=87739132, dropped it...
Content-Type:text/plain; charset=UTF-8
Message: test message
(Packet trace shows a SIP MESSAGE, answered by a 405.)
...and apparently is unable to originate them either; SendText, which
looks as though it ought to be the right way to send them, produces (in
the context of a call, since I can't send the message outside one):
-- Executing [604@default:2] SendText("SIP/mob776-02ba6050", "test
message") in new stack
-- Incoming call: Got SIP response 405 "Method Not Allowed" back from
10.0.155.21
even though it's also making a SIP MESSAGE request.
The only documentation I can find talks about a patch and is pretty old:
http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
What I would like is to be able to send a textual message from the phone
into an AGI script (or for other processing), and to return results the
same way. Is anyone doing this with later versions of Asterisk, or
indeed anything else?
Roger
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Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote: >In the meantime, does anyone have a nice way to update a stable/stock lenny >installation with the updated glibc as well as the latest kernel At this point the easiest option will be to upgrade to squeeze. R -- _ -- 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] Failed SIP registration kicks registered device off?
On Wed, Jan 12, 2011 at 10:13:22AM -0600, Kevin P. Fleming wrote: >His point is valid though... A's registration should not have been >overwritten until B *successfully* registered. A failed attempt to >register should have no effect on the existing registration. Indeed, the avenue for a brute-force DoS (absent something like fail2ban) is fairly obvious. -- _ -- 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] Callback form to place on site for customers. Recomendation to achieve this.
On Sun, Jan 02, 2011 at 12:04:07AM +, JP CR wrote: >What I want is when a potential client submits his number... the PBX dials the >number makes an announcement and dials an extension (which is actually a >cellhopne dahdi member) and makes the connection. You might try something based on this: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out It's easy to generate a call file which dials the agent's phone, waits for a pickup, and then dials out. -- _ -- 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] incoming
On Mon, Jan 03, 2011 at 02:41:36AM +0400, Thomas Perron wrote: >Cool. So, one Asterisk machine handling up to 100 DID numbers, correct? As many as you like, modulo memory and CPU requirements. >I assume that the DID mumbers dialed would be the exaxt match needed >to start the respective context. Correct? Depends on how they're presented to you by the DID provider. -- _ -- 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 to initiate a two-party call from within Asterisk
On Mon, Nov 29, 2010 at 01:36:17PM -0600, Chris Gentle wrote: >This is "click-to-call". It can be done with the Asterisk Manager Interface >(AMI). See this site: Thanks to you and Tilghman for this, though as it turned out it was much simpler to avoid AMI completely and use the Extension: parameter to an outgoing call file. Roger -- _ -- 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] start services automatically
On Mon, Dec 20, 2010 at 02:34:23PM +, salaheddine elharit wrote: >the 0,1, and 6 are OFF just the 2,3,4,5 are ON , >and when i reboot the server i found that the service httpd is off with >command "service httpd status" and service asterisk status > >please advice This is just one of many problems you will encounter. You need to train or hire an actual Unix/Linux system administrator. -- _ -- 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] Specifying DID for outbound calls
On Sun, Dec 19, 2010 at 12:14:11AM -0500, Stephen Reese wrote: >Thanks for the heads up, I have been setting the caller-ID but the >trouble I'm running into is specifying the which number to call out >as. How can an extension specify a different number? See below for my >current extension.conf, thanks. I think I'd probably replace the two outgoing contexts with one, using a GotoIf to distinguish between the two phones (branching into your current code). Alternatively you could give them each a custom context (say phone1 and phone2); phone1 would include incoming and outgoing1, phone2 would include incoming and outgoing2. -- _ -- 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] Call sip:u...@domain.com?
On Fri, Dec 17, 2010 at 04:52:32PM +0100, Gilles wrote: >Thanks for the tip but I wanted to be able to call _any_ SIP number, >not just Ekiga, so needed a destination-agnostic solution. How would you _expect_ to be able to specify a destination server from a telephone keypad? -- _ -- 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] Configuring server to call SIP numbers on the Net?
On Mon, Dec 13, 2010 at 07:28:58PM +0100, Gilles wrote:
>This is a newbie question : With a simple Asterisk server on a private
>LAN, an FXO port to handle the PSTN, and an ADSL connection to the
>Net, ie. with no VOSP in the mix... how should I configure Asterisk so
>that SIP clients can dial SIP numbers on the Net, such as those below
>to perform an echo test?
If you want to dial a SIP number that's not on your local server, you
need to route it via dialplan logic. You could do this with a prefix
code if you want to be able to dial lots of numbers at the same server:
exten => _9NX.,1,Dial(SIP/user:p...@ekiga.net/${EXTEN:1})
or something more specific if you just want to connect to one:
exten => 602,1,Dial(SIP/user:p...@ekiga.net/*010600)
(Don't quote me on syntax; I don't have any SIP examples handy as I only
use it for local-network calls.)
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Re: [asterisk-users] Asterisk 1.6.2.10 video call
On Mon, Dec 06, 2010 at 03:23:42PM +0100, Jonas Kellens wrote: >I'm trying to set up a video call from my Ekiga client to a >Grandstream GXV3140 IP-phone. The call succeeds but there is no >video. Try restricting video codec to H.261. R -- _ -- 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] How to initiate a two-party call from within Asterisk
The desired result is that user A's phone rings; when he picks it up, user B is dialled, and user A's channel is connected to that. (This is to be a back-end for a web-based address book.) -- _ -- 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] Elementary question - accessing feature codes fromcell phone
On Fri, Nov 05, 2010 at 12:49:45PM -0400, John Regal wrote:
>Anyway, I think the idea of replicating the function into an extension will
>work. Any pointers on the best way to accomplish this? I created a new
>extension but am unsure what to do next. I thought about the FollowMe
>feature but I would have to hardcode the number and I want to be able to
>enter a forwarding phone number for the extension using my cell.
You could set up an extension match that triggers on
(feature ID)(access code)(extension)
as it might be, with an access code of 62889:
exten => _*7262889.,1,Set(FWDNUM=${EXTEN:8})
and then put FWDNUM into the astdb or however else you want to handle
it.
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Re: [asterisk-users] Asterisk, VoIP and Samsung iDCS100
On Wed, Nov 03, 2010 at 12:05:51PM +, Ronny Adsetts wrote: >What hardware would I need in the Asterisk so I could hook up some analogue >extensions? Am I right in thinking I need something like an FXO/FXS card? Yes, this ought to work. If you're plugging phones into the Samsung it's providing an FXS interface, so you'll need an FXO interface to talk to that; if you want to connect those analogue phones to Asterisk, you'll also need FXS interfaces (though as a short-term fix it would probably be easier to leave them plumbed directly into the Samsung box). Getting four modules (each of which can be FXO or FXS) on a single card is pretty easy (I use an OpenVox A400P from voipon, following recommendations on this list). You could then connect (some combination of) analogue channels to (some combination of) SIP phones, and vice versa to allow outward dialling. Once you build the Asterisk-only system, you can use the FXO modules to connect to analogue PSTN lines (assuming you have a use for this). Roger signature.asc Description: Digital signature -- _ -- 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] IAX or SIP - connecting two Asterisk servers together
On Tue, Nov 02, 2010 at 03:20:48PM -0400, Silver Thorne wrote: >I am not looking for someone to do this for me, I am just not really >sure how to get started. Perhaps some suggested reading, examples, >etc? The simplest approach would be to skip the answering and just dial through immediately, feeding back the destination's ring tone to the originator. Set up an IAX link between the two boxes (you could do it with SIP, but I found IAX less trouble), then set up an appropriate bit of dialplan logic on the American box, as it might be: exten => 4682,1,Dial(IAX2/usern...@eurobox/8873) Roger -- _ -- 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] Asterisk, VoIP and Samsung iDCS100
On Tue, Nov 02, 2010 at 05:54:27PM +, Ronny Adsetts wrote: >Would it be possible do you know to use the Samsung handsets with an Asterisk >system? Is it worth even trying to save money here? (I've no idea of the cost >of VoIP handsets for use with Asterisk). I've never heard of Samsung handsets in the context of Asterisk, so I'm guessing they're Samsung-only. If I were in your shoes I'd go for open standards all the way. The cheapest Grandstream SIP phone will run you about 40 pounds retail and _will_ work with Asterisk - or with anything else that speaks SIP. (And of course with an open platform you can give people softphones on their PCs if that's what they prefer - some laptop users do.) Roger signature.asc Description: Digital signature -- _ -- 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] Asterisk, VoIP and Samsung iDCS100
On Tue, Nov 02, 2010 at 04:13:01PM +, Ronny Adsetts wrote: >3. Other ways? It all rather depends on what your proprietary system has been set up to do. (If you didn't already have the Samsung box, you wouldn't need to buy one.) Dedicated telephony hardware tends to be restricted in all sorts of perverse ways to try to make you buy more from the same manufacturer; that'll be your biggest problem. Ideally you would be able to tell your iDCS100 "there are multiple VoIP phones at this IP address", and connect to the Asterisk server over the LAN. How you would go about that, I have no idea; I suspect "SIP IP Trunking" is what Samsung calls this feature. The more work you can shoft onto the Asterisk server, the cleaner this will all be. In this scenario, the Asterisk server just has a normal network card in it, and you shift all your VoIP traffic over the LAN and VPN. signature.asc Description: Digital signature -- _ -- 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] SIP Load Balancing
On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote: >I have a very simple setup with two SIP routes to my carrier. I need to have >every other phone call placed to that carrier go to a different address. I think what you need to do here is check/set a variable in the astdb. (If the variable is 1, set it to 2 and route via A; otherwise, set it to 1 and route via B.) Translation of this to dialplan logic is left as an exercise for the student. R -- _ -- 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] SIP - no audio behind nat problem
On Fri, Oct 15, 2010 at 06:22:07PM +0200, Zarko Zivanovic wrote: >We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this >natted network. The simplest solution will be to stick another Asterisk box inside the NAT and tunnel IAX or SIP over a VPN. R -- _ -- 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] sound file debug
You have two separate problems here: (1) >dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit, >mono 8000 Hz You should have generated this with 16-bit resolution, like all the others. (2) Not sure about the cents - sure it's coming out as 16-bit? Is the file in the right place? -- _ -- 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] Asterisk sharing a line with POTS handsets: how to interoperate cleanly?
I now have an OpenVox A400P and it is working well. Thanks to Ade Vickers for the recommendation, which I second. However, I need to make a slow transition between a conventional multiple-extension setup and a full VoIP network on these premises. So at the moment the Asterisk box shares the PSTN connection with several conventional analogue handsets. The desired result for an incoming call is that the Asterisk server will wait N seconds before answering (which I can arrange easily enough), and if the call has been answered on one of the handsets by that time the Asterisk server should ignore it completely. Otherwise it should start checking CLID, prompting for extensions, and other good stuff, which again I know how to do. What is a good approach to making sure the Asterisk server doesn't pick up a call that has been answered elsewhere? (Ideally in pure dialplan, but a perl AGI would also do.) R -- _ -- 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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
On Sat, Oct 02, 2010 at 04:09:33PM -0400, bruce bruce wrote: >Can't I in my ip tables just accept the pap2t.dyndns.org if that is bind to >the PAP2T? do you think the devices comes in with it's external IP rather >than the dyndns domain? Yes. An IP datagram carries only the source and destination IP addresses, not the DNS names associated with them. Your firewall _may_ be able to accept a DNS name to block or allow rather than an IP address, but most don't, and doing so makes you vulnerable to DNS spoofing attacks. To go further would be thoroughly off-topic for this list. Roger -- _ -- 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] minimum card for dahdi timing source ?
On Sat, Oct 02, 2010 at 06:24:24PM +0200, mancyb...@gmail.com wrote: >for a vicidial server which uses only voip, >which is the minimum telephony card which would provide the required clock >timing source for conferences to work properly ? Can't speak for vicidial, but MeetMe() works fine for me with asterisk 1.4 and ztdummy. I would assume 1.6 with dahdi works similarly... R -- _ -- 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] Problems compiling Asterisk on Debian
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote: >What should i do? aptitude install module-assistant m-a a-i dahdi -- _ -- 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] Unable to open vm-INBOXs
On Wed, Sep 22, 2010 at 12:32:21PM +0200, Jonas Kellens wrote: >[Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: >File vm-INBOXs does not exist in any format >[Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable >to open vm-INBOXs (format 0x8 (alaw)): No such file or directory >I do not find this particular soundfile on my system. How are you invoking it? That terminal "s" on the filename looks rather unexpected. R -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote: >I have setup an OpenVPN tunnel between Server A (running Asterisk) and >Server B suppling it's SIP Phones with DHCP pool of IPs. Have you considered running Asterisk on Server B as well, and using IAX to trunk between them? This is working well for me. Roger -- _ -- 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] Bug with Realtime?
On Mon, Sep 20, 2010 at 11:59:16AM -0400, Dan Journo wrote: >Can we not do pastebin any more? No, it's just one user with an excessively paranoid and chatty mailfilter. -- _ -- 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] Authentication best practice
I am working with a simple "follow-me"-style service: rather than have something that rings several phones in turn, the user dials a number (in the present implementation, unique to that user) to register his presence at a particular extension. What's the standard way to protect this from unauthorised use? Voicemail()-style, where the user has to enter a PIN once the connection is made? With a very long number, so that number and PIN can be integrated in the phone's contact list? With a single central number, where the each user has to enter his own unique identifier _and_ PIN? Roger -- _ -- 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] Correct queue agi syntax in 1.6.2.11
On Mon, Sep 13, 2010 at 08:15:34PM +0200, Jonas Kellens wrote: > [Sep 13 20:14:59] -- Launched AGI Script > /var/lib/asterisk/agi-bin/cleanpickup.agi > [Sep 13 20:14:59] opruimenpickup.agi: Failed to execute > '/var/lib/asterisk/agi-bin/cleanpickup.agi': Permission denied So check that /var/lib/asterisk/agi-bin/cleanpickup.agi is executable by the user under which asterisk is running. -- _ -- 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] Force ip disconnect after register?
On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote: >Is there a way to drop a ip connection to asterisk after a number of >register attempts. Consider writing a filter for fail2ban [http://www.fail2ban.org/] that works on the Asterisk logs? -- _ -- 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] A way to check against a list of numbers?
On Fri, Sep 10, 2010 at 03:51:01PM -0500, Hose wrote: >Does anyone have any suggestions as to >how to approach that, or if they have a entirely different way in mind? AGI script that can look directly at your master list of numbers/routes? R -- _ -- 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] What can make G.729a codec hostid change?
On Tue, Sep 07, 2010 at 10:58:18AM -0700, Dave Platt wrote: >Note that "ifconfig" will not necessarily show all of your >interfaces (hard- or soft-) - only the active, configured ones. ifconfig -a would help here. Kernel upgrades often seem to bring in new default interfaces. If this turns out to be the problem, rmmod or a custom kernel compilation may do the trick. (Of course if you've _lost_ an interface you were using under etch this may be more of a problem.) R -- _ -- 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] Wanted: UK-specific hardware recommendations (FXO and FXS)
I have a pair of Asterisk servers which are happily routeing VoIP calls. I want to hook one of them to the PSTN. Given that I am in the UK, what is a reasonably easily-available device to provide an FXO interface from a Linux box, with a minimum of faffing around with drivers? Just one line is needed, though in theory two might eventually be useful. My usual white-box hardware suppliers don't seem to play in this field. Also: I've heard good things about the PAP2T for getting analogue handsets to talk to a VoIP server. But all the ones I can see on eBay are PAP2T-NA models. Will these work with British handsets? (Obviously with a plug adaptor to put the BT jack into an RJ11 socket, but that's relatively easy to arrange.) Roger -- _ -- 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
