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] 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] 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 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] 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] 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
[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 sip:asterisk@[ipaddr]:5060
.) 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=sip:FROM@DOMAIN
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] 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
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 12:23:22PM -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] .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] 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
[asterisk-users] DTMF digits received, but not completely forwareded
Hello, we are running an Eicon Diva Server card with chan_capi and Asterisk-1.4.8. When we put in capi.conf softdtmf=off, the local command read() is recognizing dtmf digits from cell phone and from ISDN phones and from VoIP phones (via PSTN) very well, and asterisk is forwarding those digits correctly and completely to other switches via SIP. However, when an old analogue telephone is sending the DTMF digits, they are recognized approx 60% only. Even more strange: When we put softdtmf=on (relaxdtmf=off), the local command read is correctly recognizing _almost_any_ incoming DTMF digits from the Diva card. This is the desired behaviour, which we would like to keep. However, when the read() is running on another asterisk box, and the call is switched to that box using Dial(SIP/), only 80% of the digits are arriving at the other box. We tried with RFC2833 and with INFO (on both sides same). No difference. Can anyone please give me a hint, why not every digit, which the first box would recognize with read(), is forwarded by SIP to the other box? Why that difference? 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] 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
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
[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] 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] 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 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=nprev=_thl=enie=UTF-8layout=2eotf=1sl=notl=enu=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] 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 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] 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] 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 sip:2...@firedrake.org from Display Name sip:mob...@firedrake.org;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 -- _ -- 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 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] Vitelity offline?
Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis -- _ -- 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
Re: [asterisk-users] Remote-Party-ID party=called
Hello, escape the semicolons with a backslash! At least in astersik-1.6.X this works fine. I.e. replace in the SIP-Header-command all ; by \; Regards, Roger. Jonas Kellens schrieb: Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten = 10,1,NoOp() exten = 10,n,SIPAddHeader(Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) exten = 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12 14:56:19] -- Executing [...@from-test:1] NoOp(SIP/test6-0094, ) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-0094, Remote-Party-ID: eric sip:1...@192.168.1.150) in new stack [Jul 12 14:56:19] -- Executing [...@from-test:3] Dial(SIP/test6-0094, SIP/test2) in new stack/ SIP debug : /asterisk*CLI sip set debug peer test6 SIP Debugging Enabled for IP: 192.168.1.104:5063 [Jul 12 15:02:42] --- SIP read from 192.168.1.104:5063 --- INVITE sip:1...@192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5063;branch=z9hG4bK-fe158095 From: test 6 sip:te...@192.168.1.150;tag=adbbedf0959298ddo3 To: sip:1...@192.168.1.150 *Remote-Party-ID: test 6 sip:te...@192.168.1.150;screen=yes;party=calling* Call-ID: fb31bee7-94a6a...@192.168.1.104 CSeq: 101 INVITE Max-Forwards: 70 Contact: test 6 sip:te...@192.168.1.104:5063 Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp/ In all the other SIP-messages there is no trace of the Remote-Party-ID header... Shouldn't there be a /*Remote-Party-ID: eric sip:1...@192.168.1.150;party=called */somewhere ?? Jonas. -- Roger Schreiter Spindelberg 11 D-74354 Besigheim Tel.: +49 7143 36476 -- _ -- 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] Remote-Party-ID party=called
Hello, the SIP header now should be sent. What the remote device is doing with this header, or whether the syntax of the header is as the remote device expects it, is another question. You can check with sip set debug on whether the header is now sent as you expect! If it does, I cannot tell you, why your Cisco device is not displaying it. Regards, Roger. Jonas Kellens schrieb: Roger, your answer did resolve something : /[Jul 12 15:51:24] -- Executing [...@from-test:2] SIPAddHeader(SIP/test6-009a, Remote-Party-ID: eric sip:1...@192.168.1.150;party=called ) in new stack/ However this SIP-header is never send as a SIP-message to the phone from where I'm placing the call. The name eric is not displayed on the screen. This is a Cisco SPA 941 and supports the Remote-Party-ID. -- _ -- 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] Help with IP Routing
Hello, if the remote side (the public IP side) is capable to do something like asterisk's nat=yes (in sip.conf), than a mascerading router (like every cheap DSL router) would do enough NAT do let SIP work. If the remote side does not support that nat-hack (which is not SIP standard), than you will need a NATing router also doing a lot of SIP header rewriting. Maybe the most easy thing will be to install asterisk on the NATing machine and operating regular SIP links on both sides. Roger. Nivin Kumar schrieb: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- 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] Help with IP Routing
Nivin Kumar schrieb: Is there a tool that will allow me to automatically change sip headers in realtime? Hi, imho changing the SIP headers will not be sufficient, since the old IP addresses are now private IP addresses (only in your network, outside, there are still public, but pointing not to your equipment). You will need a gateway, which does both: NAT 1:1, old IP addresses - new IP addresses and rewriting or all SIP headers, including those headers concerning the RTP endpoints. Maybe, you can do this with OpenSIPS. But I'm not sure about the SIP-headers for RTP. For H.323, it is imho less complicate, since it is robust for NAT and has no headers including IP addresses. Regards, 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] Early audio problem in chan_dahdi
... Hi, I've found the solution. I remembered, that with IAX2 - DAHDI everything is fine. Only SIP - DAHDI showed the problem. It seems, that chan_sip does not open ealry audio, if progressinband=yes in sip.conf. progressinband=no is needed for early audio. Strange! Anyway, that's ok for me now. 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] Early audio problem in chan_dahdi
Hello, if have a problem since I switched to asterisk-1.6: When making an outgoing call through chan_dahdi, I cannot hear anymore early audio, the asterisk generated sound (as defined in indications.conf) is played. Thus, I cannot hear announcements by the operator, and when the line is busy, sometimes I can hear first the ringing indication by asterisk, and some moments later the busy. I already tried both in chan_dahdi.conf: callprogress = yes and callprogress = no No difference. What I'm doing wrong? 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] BRI vs. PRI?
Ken D'Ambrosio schrieb: ... pretty pricey. Is there any reason that a BRI can't do exactly the same stuff, but on 2B+D instead of 23B+D? Hello, this depends on your operator and the telcom regulation in your country. In Germany, the main difference (besides the number of channels) is the numbering plan. With a BRI line, you'll get up to 10 single numbers, maybe consecutive, but without a mean to add or manage extensions. With at least two BRIs or with a PRI, in Germany you generally get a range of numbers, and you may manage the extensions on your own. E.g.: If you get the range -00 .. -29: Typically small business use than as numbering plan: -0 -10 .. -29 Maybe in your country the situation is similar. Roger. -- _ -- 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] Virtual ISDN device /dev/XYZ
Hello, I do remember having read some weeks ago something about a virtual device provided by asterisk, behaving like an ISDN device, i.e. like /dev/isdn0. I know iaxmodem, but iaxmodem imho unfortunately does not transport raw ISDN data (HDLC frames), but only voice. Do I remember right, and there is an aseterisk application, providing such a device, which other linux executables can use, which expect a common ISDN device? Roger. -- _ -- 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] Hangs up after 16 minutes on a call.
Hello, I had a similar problem with asterisk-1.2 long time ago, when I used the S(...) parameter in the dial command. Even if I used S(8), which is approx one day, asterisk hang up after exactly 64 seconds. When I erased the S() parameter completely, the problem was gone. Imho, this problem does not occour in asterisk 1.6. Anyway, if you are using the S paramter, try without and check, whether it helps! Roger. William Kenworthy schrieb: Hi, Ive just upgraded my home asterisk (on gentoo) from 1.4 to 1.6 and have an odd problem. After about 16 minutes on a call, it hangs up. Is -- Roger Schreiter Spindelberg 11 D-74354 Besigheim Tel.: +49 7143 36476 ___ -- 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] Sangoma card reports HDLC errors
Hello, I've recently bought a new Sangoma A104d PCI-card. WANPIPE release is 3.4.7. Machine is a dual Xeon with Debian 5.0.3. Asterisk is 1.6.11 with recent libpri and dahdi. When I boot the machine (including hardware and wanpipe and dahdi drivers) and start asterisk, everthing runs fine for almost 5 minutes. Then every few seconds HDLC errors occour. I also tried booting - and starting asterisk later: The HDLC errors start almost 5 minutes after booting (not after starting asterisk). I cannot find any other job, starting after almost 5 minutes and consuming resources, top also show almost 100% idle time. I also tried to produce high load during the first 4 minutes, tar.gzing large directories. Anyway, no HDLC errors during the first 4 minutes, just after almost 5 minutes. For me, it seems, that there is anything within the sangoma drivers or the dahdi software overrunnig after appprox 4 minutes and 50 seconds, causing those HDLC errors. Any idea, who to find the cause of the error or how to solve it? Roger. ___ -- 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] Sangoma card reports HDLC errors
Hi, mea maxima culpa. I've found the cause, however Sangoma card and driver are working excellent. Just for the archive, to help someone else, maybe having the same problem: In order to test my system before bringing it to the data center, I plugged in a loop cable. Therefore I had to switch the E1 clock mode to master instead of normal. Of course I knew, I have to switch it back to normal, before connecting to the real carrier, but I forgot. Anyway, the dependence of the 4:50 minutes remains funny. Maybe a function of the divergence of the two clocks (my one the the carrier's one). Thank you answering anyway! Roger. ___ -- 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] dahdi_tool shows no alarms, but no line connected
Tzafrir Cohen schrieb: ... head -n1 /proc/dahdi/* # head -n1 /proc/dahdi/* == /proc/dahdi/1 == Span 1: WPE1/0 wanpipe1 card 0 (MASTER) HDB3/CCS/CRC4 == /proc/dahdi/2 == Span 2: WPE1/1 wanpipe2 card 1 HDB3/CCS/CRC4 == /proc/dahdi/3 == Span 3: WPE1/2 wanpipe3 card 2 HDB3/CCS/CRC4 == /proc/dahdi/4 == Span 4: WPE1/3 wanpipe4 card 3 HDB3/CCS/CRC4 If there are no alarms there, the wanpipe driver probably did not report them to DAHDI. Probably. But why? (When I turn of wanpipes, I can see them disapear with dahdi_tool.) How can I investigate the reason? Roger. -- Roger Schreiter Spindelberg 11 D-74354 Besigheim Tel.: +49 7143 36476 ___ -- 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] dahdi_tool shows no alarms, but no line connected
Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name | Protocol | Station | Status| wanpipe1| AFT TE1 | N/A | Connected | wanpipe2| AFT TE1 | N/A | Disconnected | wanpipe3| AFT TE1 | N/A | Disconnected | wanpipe4| AFT TE1 | N/A | Disconnected | However: # dahdi_scan 2 [2] active=yes alarms=OK description=wanpipe2 card 1 name=WPE1/1 manufacturer= devicetype= location= basechan=1 totchans=31 irq=0 type=digital- syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS Why are the dahdi tools not reflecting the values by wanrouter? Roger. ___ -- 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] Monitor
if you haven't exectued the queue cmd you cannot know who will took that call.
You cannot know this before the agent took it because there are many agents who
can do it.
You can know it via cdr or manager interface, but only when the call is tooked
or finished.
On Mon, Jul 06, 2009 at 03:23:29PM +0530, Sriram wrote:
Hi All
am using trixbox with call queues..I've set setinterfacevars=yes in
queues.conf and following is dial plan :
[test]
exten = s,1,Answer()
exten =
s,2,Set(FILE_NAME=${CALLERID(num)}-${MEMBERINTERFACE}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,3,Monitor(wav,${FILE_NAME},m)
exten = s,4,queue(55365)
exten = s,5,Hangup()
but MEMBERINTERFACE is always empty - i basically want to add the member who
took that call in that monitor file..i tried in trixbox too bt problem
persists...can anyone throw some light ?
rgds
Sriram
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Re: [asterisk-users] Multiple Outgoing Lines: extensions.conf
hello,
you can define a variable in sip.conf in each extension like:
[201]
...
setvar=LINE=89859716
...
then in extensions when user 201 calls you have a the var defined and you can
use it with ${LINE}.
On Thu, Jun 18, 2009 at 08:19:27PM +1000, Clara Chan wrote:
Dear all,
I am currently trying to configure a PBX make use of a multiple of outgoing
lines, currently my extensions.conf looks something like below
; extensions.conf
; 20th October 2008
[globals]
sip1=201
sip2=202
sip3=203
sip4=204
[general]
autofallthrough=yes
[default]
[incoming_calls]
exten = _89859715,1,Dial(SIP/201)
exten = _89859716,1,Dial(SIP/202)
[macro-sipmail]
exten = s,1,Verbose(1,Extension ${ARG1}) ;line req to pick up ext if it's
not
reg.
exten = s,n,Dial(SIP/${ARG1},30)
exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?busy:unavail)
exten = s,n(unavail),Voicemail(${ar...@default,u)
exten = s,n,Hangup()
exten = s,n(busy),VoiceMail(${ar...@default,b)
exten = s,n,Hangup()
[macro-conference]
exten = s,1,Playback(conf-theatre)
exten = s,n,MeetMe(${ARG1},i)
[internal]
include = outbound
;Voicemail
exten = 8,1,VoiceMailMain()
;Conference Rooms
exten = 600,1,Macro(conference,600)
exten = 601,1,Macro(conference,601)
exten = 602,1,Macro(conference,602)
exten = 603,1,Macro(conference,603)
exten = 604,1,Macro(conference,604)
exten = 605,1,Macro(conference,605)
;Extensions
exten = 201,1,Macro(sipmail,201)
exten = 202,1,Macro(sipmail,202)
exten = 203,1,Macro(sipmail,203)
exten = 204,1,Macro(sipmail,204)
exten = 205,1,Macro(sipmail,205)
exten = 206,1,Macro(sipmail,206)
exten = 207,1,Macro(sipmail,207)
exten = 208,1,Macro(sipmail,208)
;Digium card Channels
exten = 301,1,Dial(Zap/1-1)
exten = 302,1,Dial(Zap/1-2)
[outbound]
exten = _9.,1,Dial(SIP/${EXTEN:1...@61289859715,30,tr)
exten = _9.,n,Hangup()
exten = 000,1,Dial(SIP/0...@61289859715)
exten = _7.,1,Dial(SIP/${EXTEN:1...@61289859716,30,tr)
exten = _7.,n,Hangup()
exten = 000,1,Dial(SIP/0...@61289859716)
[phones]
include = internal
include = incoming_calls
include = outbound
Each extension has its own incoming and outgoing account, I know how to route
the incoming number to each particular extension, but how does one route
outgoing calls from a particular phone to use a specific line, ie, from phone
no. 89859715 an outgoing call will use caller id 89859715 and line 89859715?
Or
for phone no. 89859716 to use the 89859716 line?
I have sixteen outgoing lines I need to configure, so that each individual
phone can send its own caller id; any suggestions?
Thanks for your thoughts.
Rgds,
Clara
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Re: [asterisk-users] ITSP's no longer supporting IAX?
Steve Totaro wrote: I understand you are a developer and you want IAX2 to be great. That is your job, but the fact is that it is not and has caused audio and security problems for YEARS in EVERY release. It should bug you and everyone at Digium that waves the IAX2 flag. Can you elaborate on these audio and security problems Steve? Looking at the two protocol specs I cannot see a basis for your claim. IAX doesn't embed the local IP address in the packet data but that's surely no substantive security. It does separate data and signaling at the application-level, but again, that's no basis for such a claim. Protocols must be looked at separately from their implementations. From the various responses it appears that Asterisk 1.4's implementation of IAX has flaws. These do not necessarily reflect on the protocol. OTOH, there are a lot of engineers with SIP skill and experience who, naturally, are concerned with their investment in time, education, and experience. While this may or may not apply to Sonicwall engineering, it's also true that any streaming protocol will be better handled by devices that process packets in ASICs (high-end firewalls) rather than CPUs (PCs and low-end firewalls). FWIW (2 data points) I get uniformly better service from our IAX trunk provider than our SIP trunk provider. No idea whether that's protocol, implementation (1.4 on my side), or provider-related though I suspect the later. Roger Marquis ___ -- 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] ITSP's no longer supporting IAX?
Jon Pounder wrote: This sounds like a bunch of gobbledegook spewed out by those very high end firewall vendors. Call it what you want but anything that processes packets in any way and makes a decision on what to do is by definition a CPU. You won't find much support for that opinion in network engineering circles. The processing advantage of ASICs is easily measured and widely documented. ASICs are particularly critical to latency-sensitive protocols and those using small packet sizes with correspondingly high packet counts. According to Praveen Kumar (Founder/CEO of Packet Island) the ASIC differential is even more noticeable with interactive streaming video than streaming audio. Roger Marquis ___ -- 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] 1.6
Jeff LaCoursiere schrieb: Is it ready for prime time? He Jeff, at least version 1.6.0-beta9 was not yet very stable. We are also used to handle serveral Mmin/month with asterisk 1.4, but in our test environment, our asterisk 1.6.0-beta9 consumed file handles without releasing, and even a previous ulimit -n 9 could not prevent the system from causing network busies ... . Maybe, the current 1.6.0.3-rc1 has been improved. We also would like to merge the stability of current 1.4 with the new features of 1.6. Regards, Roger. ___ -- 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] Packet size limit for HDLC?
Hi, I figured out, that app_pppd suffered from overruns under high out traffic. (ping -s 600 destip was already high in this context.) I've just made a quick and dirty hack to fix it. If interested, just download the original package by Sirrix (as mentioned on VoIP-Wiki) and the replace their app_ppp.c by: http://planinternet.net/download/voip/asterisk/app_pppd.c Maybe I will later find the time to bundle a complete package, like the one by Sirrix. Regards, Roger. ___ -- 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] Packet size limit for HDLC?
Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with the analogue modems. Then, everything went fine (for a while). Unfortunately, PPP via ISDN is typically using a MTU and a MRU of 1500, and I found, that some commercial ISDN routers do not allow negotiating MTU and MRU. They insist to use a size of 1500. Since, using CAPI or ISDN4Linux (not via asterisk), pppd is working well with the MTU/MRU value of 1500, I assume, there is some packet size limitation in the asterisk part (including app_pppd). I tried to find any too small buffer or similar, but successless. May I ask you, where do you think, the limitation does come from: - from app_pppd (I don't think so) - from libpri - from chan_dahdi - from the dahdi kernel modules - from the asterisk kernel Any hint is welcome! Regards, Roger. ___ -- 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] Packet size limit for HDLC?
Eric \ManxPower\ Wieling schrieb: ICMP is used to determine maximim packet size. If you or the other end are blocking all ICMP then MTU Path Discovery will not work. It's a Hi, the problem is, the other side (ISDN-router) does not negotiate the MTU while setting up PPP. I can see this in the log file: Our side is proposing 296, but the other answers with NACK and tells 296. I think, my side is doing something according RFC, when proposing a smaller MTU than usual, but this does not solve my problem, because: More info: http://www.znep.com/~marcs/mtu/ A MTU of 1500 is typical for PPP over HDLC, and when my solution does not do, what is typical, it is not compatible enough. Now I want bring asterisk and app_pppd also to work with a MTU of 1500 (like native linux ippp also does). I want to understand, why PPP via asterisk is failing, when MTU is 1500. Regards, Roger. ___ -- 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] IAX2 client for 'eee pc 1000'
Rob Hillis wrote: The solution for the problem of an IAX client is a SIP client. That's not a particularly good solution if you have a NAT between your client and Asterisk. IAX is still *much* easier to get working through a firewall. It's working fine here (Twinkle/Ubuntu over NAT/Netscreen). Didn't have to change any settings on the firewall either. Roger Marquis ___ -- 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] change codec mid-call
hello, I would like to know if it's possible to change the codec of a call in the middle of the call. I have an asterisk without g729 codecs and I recieve an incoming call. The codec is negotiated in ulaw althought who is calling have a g729 codec. My * plays and announcements and call and extension to pass the call to it. The extension have a g729 codec too. It is possible to change the codec of incoming call from ulaw to g729 and then asterisk bridge the calls? thanks for the help. Roger -- Roger Casaponsa - Adam Telefonía IP email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www: http://www.adamvozip.es http://www.adamvozip.es/ tlf: 934465010 / 933968021 ___ -- 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] QOS for outgoing SIP calls
Chris Mason wrote: QOS can only be on outgoing, you can't set the priority of a packet after you receive it. The only other solution would be the cooperation of the ISP to provide QOS upstream of you. Good luck. QOS is probably not the most precise term as it's normally associated with RSVP, MPLS, packet headers, etc. But you can, in Netscreens at least, define a Guaranteed Bandwidth. We do this for SIP/IAX IPs, in both outgoing and incoming policies, and it works both ways. Audio quality is good and there are no chan_sip.c: Peer is now (UNREACHABLE|Lagged) messages even during long DVD or Bitorrent xfers. The reason it works outbound is a no-brainer, but inbound bandwidth is also effectively guaranteed. Sure there's no way to control external devices that ignore ICMP source-quench or break TCP congestion control but those flows are typically limited to nefarious sources which would not be responsive to other types of QOS anyhow (BGP being one potential exception). Roger Marquis ___ -- 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] Unicall CRN 32769 - far disconnected cause=Switching equipment congestion [42]
Dears, After weeks trying to contact support of my telecom about 'Seize Ack' because that is not returned, was a lock for make calls on my E1s. Now I receive back de Ack and get ready to make calls, but the technical support reports to me that my attempts to call do not send any digits to the oder site (telecom station). 8 seconds after start 'Unicall event Dialing' the line is disconnected, like when you take up the line and hold without press any digits, after some seconds you got the congestion signal. Just for consideration I receive call without any problems, provided that performed the first setup. I have use http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 to do my configuration, sources are: http://www.moythreads.com/astunicall/files/astunicall-1.4.9-0.1.tar.gz zaptel-1.4.4-6 asterisk-1.4.9 libsupertone-0.0.2-1 spandsp-0.0.4-1 libunicall-0.0.3-1 libmfcr2-0.0.3-1 The only difference is I have use the sources to make a SRPM - RPM files on CentOS 5. Here is my config files: zaptel.conf loadzone= br defaultzone = br span=1,1,0,cas,hdb3 span=2,2,0,cas,hdb3 span=3,3,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 cas=32-46:1101 cas=48-62:1101 cas=63-77:1101 cas=79-93:1101 unicall.conf [channels] loglevel=255 language=pt_BR context=from-pstn usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both protocolclass=mfcr2 ;protocolvariant=br,20,4,x,max-seize-wait-ack=1 protocolvariant=br,20,4 protocolend=cpe group=1 callerid=asreceived channel=1-15 channel=17-31 channel=32-46 channel=48-62 channel=63-77 channel=79-93 protocolclass=mfcr2 Here is the LOGS when I try do make calls [Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Executing [ [EMAIL PROTECTED]:32] Dial(SIP/4805-0935d828, UniCall/g1|300|) in new stack [Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call called - 'g1' [Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call caller id - '4805' [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Call control(1) [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Make call [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Creating a new call with CRN 32769 [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 0001 - [1/DIALING /Seize /Idle ] [Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Called g1 [Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Unicall/1 event Dialing [Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Exception on 15, channel 1 [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - [1/DIALING /Seize /Idle ] [Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g on - [2/DIALING /Group I /DNIS ] [Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - 4 on [2/DIALING /Group I /DNIS ] [Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g off - [2/DIALING /Group I /DNIS ] [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 - 4 off [2/DIALING /Group I /DNIS ] [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Far end disconnected(cause=Switching equipment congestion [42]) - state 0x40 [Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: Unicall/1 event Far end disconnected [Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: CRN 32769 - far disconnected cause=Switching equipment congestion [42] [Jan 30 16:41:26] VERBOSE[10717] logger.c: -- Channel 0 got hangup [Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: needcongestion [Jan 30 16:41:26] VERBOSE[10717] logger.c: -- UniCall/1-1 is circuit-busy [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Channel gains [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Channel switching [Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1 [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Call control(7) [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Clearing fwd [Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 1001 - [2/FAR DISC/Clear fwd B /Idle ] [Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Jan 30 16:41:26] VERBOSE[10717] logger.c: -- Hungup 'UniCall/1-1' [Jan 30 16:41:26] VERBOSE[10717] logger.c: == Everyone is busy/congested at this time (1:0/1/0) If someone can help me I would be very grateful. Best Regards, -- Roger C. Beraldi Martins Fone: 55 41-8828-7068 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] Unicall CRN 32769 - far disconnected cause=Switching equipment congestion [42]
Man I am a little embarrassed now...
Actually dial plans and PBX rules is where I have less knowledge of
everything that involves the asterisk, because of this I am using freePBX
and this was my problem. I make the setup for outbound trunk to UniCall
using the freePBX and in this case has a bug causing this behavior:
http://freepbx.org/trac/ticket/634
But anyway, this mistake was very clear ... I should have seen !
Thank you Moises, now everything is working !
Best Regards.
2008/1/30, Moises Silva [EMAIL PROTECTED]:
Well, that's simple, the telco is not getting any digits because YOU
are not sending any digits!
From the logs, I see you are dialing like this:
Dial(UniCall/g1|300|)
Where is the number you want to reach?
I'd expect to see
Dial(Unicall/g1/1234567890|300)
To reach number 1234567890
- Moisés Silva
On Jan 30, 2008 1:21 PM, Roger C. Beraldi Martins
[EMAIL PROTECTED] wrote:
Dears,
After weeks trying to contact support of my telecom about 'Seize Ack'
because that is not returned, was a lock for make calls on my E1s.
Now I receive back de Ack and get ready to make calls, but the technical
support reports to me that my attempts to call do not send any digits
to
the oder site (telecom station). 8 seconds after start 'Unicall event
Dialing' the line is disconnected, like when you take up the line and
hold
without press any digits, after some seconds you got the congestion
signal.
Just for consideration I receive call without any problems, provided
that
performed the first setup.
I have use http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 to do my
configuration, sources are:
http://www.moythreads.com/astunicall/files/astunicall-1.4.9-0.1.tar.gz
zaptel-1.4.4-6
asterisk-1.4.9
libsupertone-0.0.2-1
spandsp-0.0.4-1
libunicall-0.0.3-1
libmfcr2-0.0.3-1
The only difference is I have use the sources to make a SRPM - RPM
files on
CentOS 5.
Here is my config files:
zaptel.conf
loadzone= br
defaultzone = br
span=1,1,0,cas,hdb3
span=2,2,0,cas,hdb3
span=3,3,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
cas=32-46:1101
cas=48-62:1101
cas=63-77:1101
cas=79-93:1101
unicall.conf
[channels]
loglevel=255
language=pt_BR
context=from-pstn
usecallerid=yes
hidecallerid=no
immediate=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
faxdetect=both
protocolclass=mfcr2
;protocolvariant=br,20,4,x,max-seize-wait-ack=1
protocolvariant=br,20,4
protocolend=cpe
group=1
callerid=asreceived
channel=1-15
channel=17-31
channel=32-46
channel=48-62
channel=63-77
channel=79-93
protocolclass=mfcr2
Here is the LOGS when I try do make calls
[Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Executing
[EMAIL PROTECTED]:32] Dial(SIP/4805-0935d828, UniCall/g1|300|)
in
new stack
[Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call called -
'g1'
[Jan 30 16:41:17] DEBUG[10717] chan_unicall.c: unicall_call caller id -
'4805'
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Call
control(1)
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Make
call
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1
Creating a
new call with CRN 32769
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1
0001 -
[1/DIALING /Seize /Idle ]
[Jan 30 16:41:17] VERBOSE[10717] logger.c: -- Called g1
[Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Unicall/1 event Dialing
[Jan 30 16:41:17] NOTICE[10717] chan_unicall.c: Exception on 15, channel
1
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2
UniCall/1 -
[1/DIALING /Seize /Idle ]
[Jan 30 16:41:17] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g
on -
[2/DIALING /Group I /DNIS ]
[Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2
UniCall/1 - 4
on [2/DIALING /Group I /DNIS ]
[Jan 30 16:41:25] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 g off
-
[2/DIALING /Group I /DNIS ]
[Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2
UniCall/1 - 4
off [2/DIALING /Group I /DNIS ]
[Jan 30 16:41:26] WARNING[10717] chan_unicall.c: MFC/R2 UniCall/1 Far
end
disconnected(cause=Switching equipment congestion [42]) - state 0x40
[Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: Unicall/1 event Far end
disconnected
[Jan 30 16:41:26] NOTICE[10717] chan_unicall.c: CRN 32769 - far
disconnected cause=Switching equipment congestion [42]
[Jan 30 16:41:26] VERBOSE[10717] logger.c: -- Channel 0 got hangup
[Jan 30 16:41:26] DEBUG[10717] chan_unicall.c: needcongestion
[Jan 30 16:41:26] VERBOSE[10717] logger.c: -- UniCall/1-1 is
circuit-busy
[Jan 30 16
[asterisk-users] txfax_exec: Transmission loop error
Hi, I just installed Antonio Gallo's agx-ast-addons package in order to use app_txfax with asterisk-1.4. Compiling according to docs went well. However, I'm getting an error after the first page of fax: /usr/src/agx-ast-addons/app_txfax.c:438 txfax_exec: Transmission loop error The (very first) page is transferred perfect anyway. Then app_txfax unfortunetly stops the transmission. Any hints? Regards, Roger. ___ -- 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] SIP call interrupted after 64 seconds
Jaswinder Singh schrieb:
Can you post the part of your dialplan which causes this behaviour
Hi,
I've found, what's causing the problem:
My dialcommands are always of the type:
Dial(IAX2/user:[EMAIL
PROTECTED]/12345678,120,gS(${maxduration})M(connect^${some_params}))
or
Dial(SIP/[EMAIL PROTECTED],120,gS(${maxduration})M(connect^${some_params}))
${maxduration} is set to 86400 in most cases, sometimes to 3600
or 7200 (but never to 64). I checked this from within
the console.
When I leave the S() parameter away, there is no call, stopping after
64 secs. When I have the S() parameter, about every 10th call stops
after exactly 64 secs.
Thus, I assume a bug with the S() parameter in asterisk-1.4.x.
Can maybe someone check this on his machine, before I open a bug
report!
Roger.
___
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[asterisk-users] SIP call interrupted after 64 seconds
Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to asterisk-1.4.8, and do encounter the same problem. I have other asterisk machines running, using the same dialplan, without this problem. Did anyone else observe this strange behaviour of calls ending after 64 secondes of uptime? My os is Suse-Linux 10.2. Thanks for any hints! Roger. ___ --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] Unicall protocol error. Cause 32776
Dears, Here is the logs when I put loglevel=255 on unicall.conf, I have use max-wait = 1 [Dec 14 09:53:42] DEBUG[28143] chan_unicall.c: unicall_call called - 'g1' [Dec 14 09:53:42] DEBUG[28143] chan_unicall.c: unicall_call caller id - '3007' [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Call control(1) [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Make call [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Creating a new call with CRN 32769 [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 0001 - [1/DIALING /Seize /Idle ] [Dec 14 09:53:42] VERBOSE[28143] logger.c: -- Called g1 [Dec 14 09:53:42] NOTICE[28143] chan_unicall.c: Unicall/1 event Dialing [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 seize_ack_wait_expired [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 R2 prot. err. [1/DIALING /Seize /Idle ] cau se 32776 - Seize ack timed out [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 1001 - [1/IDLE/Idle /Idle ] [Dec 14 09:53:53] NOTICE[28143] chan_unicall.c: Unicall/1 event Protocol failure [Dec 14 09:53:53] ERROR[28143] chan_unicall.c: Unicall/1 protocol error. Cause 32776 [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Channel echo cancel [Dec 14 09:53:53] DEBUG[28143] chan_unicall.c: disabled echo cancellation on channel 1 [Dec 14 09:53:53] WARNING[28143] app_dial.c: Unable to forward voice or dtmf [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Channel gains [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Channel switching [Dec 14 09:53:53] DEBUG[28143] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 10, callwait = -1, thirdcall = -1 [Dec 14 09:53:53] DEBUG[28143] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Dec 14 09:53:53] VERBOSE[28143] logger.c: -- Hungup 'UniCall/1-1' [Dec 14 09:53:53] VERBOSE[28143] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Dec 14 09:53:53] DEBUG[28143] app_macro.c: Executed application: Dial Best Regards, -- Atenciosamente, Roger C. Beraldi Martins Fone: 41-8828-7068 ___ --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] Unicall protocol error. Cause 32776
Moises, I was reading about your first reply and you said in the 2nd step: 2. libmfcr2 will set the ABCD bits to 0x0 (000) ( normally the ABCD bits are in Idle 1001 ). Setting the ABCD bits to 0x0 is our way to tell the far end ( the telco ) that we want to start a call, this is known as the Seize. If I understood correctly the libmfcr2 must put bits (Size) to indicate it's will dial. But at this time on log libunicall put 0001 to Seize: [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Creating a new call with CRN 32769 [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 0001 - [1/DIALING /Seize /Idle ] [Dec 14 09:53:42] VERBOSE[28143] logger.c: -- Called g1 After this libunicall set the IDLE (1001) state again: [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 1001 - [1/IDLE/Idle /Idle ] Are this correctly ? Regards, 2007/12/14, Roger C. Beraldi Martins [EMAIL PROTECTED]: Dears, Here is the logs when I put loglevel=255 on unicall.conf, I have use max-wait = 1 [Dec 14 09:53:42] DEBUG[28143] chan_unicall.c: unicall_call called - 'g1' [Dec 14 09:53:42] DEBUG[28143] chan_unicall.c: unicall_call caller id - '3007' [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Call control(1) [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Make call [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Creating a new call with CRN 32769 [Dec 14 09:53:42] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 0001 - [1/DIALING /Seize /Idle ] [Dec 14 09:53:42] VERBOSE[28143] logger.c: -- Called g1 [Dec 14 09:53:42] NOTICE[28143] chan_unicall.c: Unicall/1 event Dialing [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 seize_ack_wait_expired [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 R2 prot. err. [1/DIALING /Seize /Idle ] cau se 32776 - Seize ack timed out [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 1001 - [1/IDLE/Idle /Idle ] [Dec 14 09:53:53] NOTICE[28143] chan_unicall.c: Unicall/1 event Protocol failure [Dec 14 09:53:53] ERROR[28143] chan_unicall.c: Unicall/1 protocol error. Cause 32776 [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Channel echo cancel [Dec 14 09:53:53] DEBUG[28143] chan_unicall.c: disabled echo cancellation on channel 1 [Dec 14 09:53:53] WARNING[28143] app_dial.c: Unable to forward voice or dtmf [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Channel gains [Dec 14 09:53:53] WARNING[28143] chan_unicall.c: MFC/R2 UniCall/1 Channel switching [Dec 14 09:53:53] DEBUG[28143] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 10, callwait = -1, thirdcall = -1 [Dec 14 09:53:53] DEBUG[28143] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Dec 14 09:53:53] VERBOSE[28143] logger.c: -- Hungup 'UniCall/1-1' [Dec 14 09:53:53] VERBOSE[28143] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Dec 14 09:53:53] DEBUG[28143] app_macro.c: Executed application: Dial Best Regards, -- Atenciosamente, Roger C. Beraldi Martins Fone: 41-8828-7068 -- Atenciosamente, Roger C. Beraldi Martins Fone: 41-8828-7068 ___ --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] Unicall protocol error. Cause 32776
Moises, I try put the line exactly like you send me, saw the time wait getting longer with the parameter you describe to increment. But the error is the same as you can see in logs. Has other way to solve this problem, may I question to my telephony service de time it's need to send back the ACK ? May the libmfcr2 does not receive the expected bit pattern for the ACK ? FULL LOG: without max-seize-wait-ack [Dec 13 08:32:09] VERBOSE[3798] logger.c: -- Called g1 [Dec 13 08:32:09] NOTICE[3798] chan_unicall.c: Unicall/1 event Dialing [Dec 13 08:32:11] NOTICE[3798] chan_unicall.c: Unicall/1 event Protocol failure [Dec 13 08:32:11] ERROR[3798] chan_unicall.c: Unicall/1 protocol error. Cause 32776 max-seize-wait-ack = 5000 [Dec 13 08:43:54] DEBUG[4845] chan_unicall.c: unicall_call called - 'g1' [Dec 13 08:43:54] NOTICE[4845] chan_unicall.c: Unicall/1 event Dialing [Dec 13 08:43:59] NOTICE[4845] chan_unicall.c: Unicall/1 event Protocolfailure [Dec 13 08:43:59] ERROR[4845] chan_unicall.c: Unicall/1 protocol error. Cause 32776 max-seize-wait-ack = 1 [Dec 13 08:39:41] VERBOSE[4494] logger.c: -- Called g1 [Dec 13 08:39:41] NOTICE[4494] chan_unicall.c: Unicall/1 event Dialing [Dec 13 08:39:51] NOTICE[4494] chan_unicall.c: Unicall/1 event Protocol failure [Dec 13 08:39:51] ERROR[4494] chan_unicall.c: Unicall/1 protocol error. Cause 32776 max-seize-wait-ack = 2 [Dec 13 08:36:18] VERBOSE[4145] logger.c: -- Called g1 [Dec 13 08:36:18] NOTICE[4145] chan_unicall.c: Unicall/1 event Dialing [Dec 13 08:36:38] NOTICE[4145] chan_unicall.c: Unicall/1 event Protocol failure [Dec 13 08:36:38] ERROR[4145] chan_unicall.c: Unicall/1 protocol error. Cause 32776 max-seize-wait-ack = 5 ... ... ... ... Best Regards, 2007/12/11, Moises Silva [EMAIL PROTECTED]: Roger, You can try to pass the protocolvariant like this: protocolvariant=br,20,4,x,max-seize-wait-ack=3000 This deserves a little bit of more explanation. br = Brazil 20 = ANI digits 4 = DNIS digits x = this is just a hack to be able to work with defaults and specify the next value. protocolvariant expect here a mask of values ( an integer ), passing NOT an integer but a character x will cause the defaults to remain. max-seize-wait-ack = Number of milliseconds to wait for the ACK. Try incrementing that number to see if works. If does, please post back results here. Regards, On Dec 11, 2007 10:52 AM, Roger C. Beraldi Martins [EMAIL PROTECTED] wrote: Moises, Thank you for your reply and the lesson of MFC/R2 ! My configs for the unicall.conf is: [channels] language=br context=from-pstn usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both loglevel=0 protocolclass=mfcr2 protocolvariant=br,20,4 protocolend=cpe group=1 callerid=asreceived channel=1-15 channel=17-31 channel=32-46 channel=48-62 channel=63-77 channel=79-93 protocolclass=mfcr2 The teleco who provides the links E1s is Brasil Telecom, I use the protocolvariant as shown in voip-info.org: Brasil Telecom protocolvariant=br,20,4 But I have a question in relation to variable: protocolend=co I was using =co and others configs I saw are using =cpe. I have change it, but don't seams to have effect to me. I read something on the internet which suggested changes in the file mfcr2.c to correct variables of timing. I believe that that should be the way to solution, but I do not feel safe to do this changes. Some research later, I saw information that in future versions of libunicall would not be necessary to rebuild lib to change parameters of timing, but I believe that's not implemented yet. How I can set a time of increased response of Seize ACK ? Thank you ! 2007/12/11, Moises Silva [EMAIL PROTECTED]: Roger, The seize ack timeout problem is because libmfcr2 is expecting a response ( an ACK ) from the far end and it does not arrive in a R2 variant dependant amount of time. Which protocolvariant do you have configured in unicall.conf? This is how the process to start a call goes: 1. When you Dial(Unicall/blah), Asterisk will ask chan_unicall.c to initiate the call. chan_unicall will ask libunicall to start the call, and libunicall will ask libmfcr2 to start the call. 2. libmfcr2 will set the ABCD bits to 0x0 (000) ( normally the ABCD bits are in Idle 1001 ). Setting the ABCD bits to 0x0 is our way to tell the far end ( the telco ) that we want to start a call, this is known as the Seize. 3. The far end should detect this bit pattern change and answer with a Seize ACK ( ABCD bits in 0xC ), in this case, libmfcr2 does not receive the expected ACK in 2000ms unless you are in Kuwait ( 4000ms ) or Nigeria (1ms ). So, let
[asterisk-users] Unicall protocol error. Cause 32776
Dears, After having set up the board Digium TE420 to receive 3 E1s, I can receive calls without difficulties. As you can see in the log below: -- Executing [EMAIL PROTECTED]:1] NoOp(UniCall/14-1, Catch-All DID Match - Found 5908 - You probably want a DID for this.) in new stack -- Executing [EMAIL PROTECTED]:2] Goto(UniCall/14-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing [EMAIL PROTECTED]:1] Set(UniCall/14-1, __FROM_DID=s) in new stack -- Executing [EMAIL PROTECTED]:2] GotoIf(UniCall/14-1, 0 ?cidok) in new stack -- Executing [EMAIL PROTECTED]:3] Set(UniCall/14-1, CALLERID(name)=4133602900) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(UniCall/14-1, CallerID is 4133602900 4133602900) in new stack -- Executing [EMAIL PROTECTED]:5] Goto(UniCall/14-1, ivr-3|s|1) in new stack -- Goto (ivr-3,s,1) *snip* -- Executing [EMAIL PROTECTED]:10] BackGround(UniCall/14-1, custom/celia) in new stack -- UniCall/14-1 Playing 'custom/celia' (language 'br') -- Executing [EMAIL PROTECTED]:1] Hangup(UniCall/14-1, ) in new stack -- Hungup 'UniCall/14-1' -- Unicall/14 released Now I am having problems to make calls using the libunicall. The problem is clear in this line of the full log: [Dec 11 10:03:54] ERROR[12935] chan_unicall.c: Unicall/1 protocol error. Cause 32776 Searching for the error I discovered it is Seize ack timed out, but I do not know exactly of what it means or how to fix it. Here is de version of softwares/libs I have use ( http://www.voip-info.org/wiki/view/Asterisk+MFC+R2). asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 zaptel-1.4.4 I already try using asterisk 1.4.10 but the comportment is the same. I don't believe the problem is in asterisk. I think my configs are correctly but not sure. Attached in text file follow the tests I have done using testunicall, config files from zaptel.conf and unicall.conf I using on this solution. More logs is in the same file. This can be caused by a problem with signaling between my settings and the standard of telephony service ? I'm using FreePBX with a Custon Trunk (Custon String Dial: UniCall/g1), my extensions_aditional has the OUT_3 = AMP:UniCall/g1 and OUTMAXCHANS_3 = 10. Someone has already gone through a problem like this ? I would be grateful if received suggestions to correct it. Log Full: [Dec 11 10:03:51] VERBOSE[12935] logger.c: -- Executing [ [EMAIL PROTECTED]:32] Dial(SIP/2290-09b18a68, UniCall/g1|300|) in new stack [Dec 11 10:03:51] DEBUG[12935] chan_unicall.c: unicall_call called - 'g1' [Dec 11 10:03:51] DEBUG[12935] chan_unicall.c: unicall_call caller id - '2290' [Dec 11 10:03:51] VERBOSE[12935] logger.c: -- Called g1 [Dec 11 10:03:51] NOTICE[12935] chan_unicall.c: Unicall/1 event Dialing [Dec 11 10:03:54] NOTICE[12935] chan_unicall.c: Unicall/1 event Protocol failure [Dec 11 10:03:54] ERROR[12935] chan_unicall.c: Unicall/1 protocol error. Cause 32776 [Dec 11 10:03:54] DEBUG[12935] chan_unicall.c: disabled echo cancellation on channel 1 [Dec 11 10:03:54] WARNING[12935] app_dial.c: Unable to forward voice or dtmf [Dec 11 10:03:54] DEBUG[12935] chan_unicall.c: Hangup: channel: 1 index = 0, normal = 10, callwait = -1, thirdcall = -1 [Dec 11 10:03:54] DEBUG[12935] chan_unicall.c: Updated conferencing on 1, with 0 conference users [Dec 11 10:03:54] VERBOSE[12935] logger.c: -- Hungup 'UniCall/1-1' [Dec 11 10:03:54] VERBOSE[12935] logger.c: == Everyone is busy/congested at this time (1:0/0/1) -- Atenciosamente, Roger C. Beraldi Martins Fone: 41-8828-7068 [EMAIL PROTECTED] libunicall-0.0.3]# cat testcall.conf destination-no 4132185900 protocol-class mfcr2 protocol-variant br,20,4 protocol-end co on-offered accept circuits 1-10 [EMAIL PROTECTED] libunicall-0.0.3]# ./testcall Chan 1, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185900' Chan 2, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185901' Chan 3, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185902' Chan 4, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185903' Chan 5, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185904' Chan 6, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185905' Chan 7, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185906' Chan 8, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185907' Chan 9, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185908' Chan 10, class 'mfcr2', variant 'br,20,4', end 2, caller 0, from '' to '4132185909' Loading protocol mfcr2 Thread for channel 0 Thread for channel 1 Thread for channel 2 Thread for channel 3 Thread for channel 4 MFC/R2 Chan 1: Call control(9) MFC/R2 Chan 1: Unblock MFC/R2 Chan 1: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 2: Call control(9) MFC/R2 Chan 2: Unblock MFC/R2 Chan 2: 1001 - [1
Re: [asterisk-users] Unicall protocol error. Cause 32776
Moises, Thank you for your reply and the lesson of MFC/R2 ! My configs for the unicall.conf is: [channels] language=br context=from-pstn usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both loglevel=0 protocolclass=mfcr2 protocolvariant=br,20,4 protocolend=cpe group=1 callerid=asreceived channel=1-15 channel=17-31 channel=32-46 channel=48-62 channel=63-77 channel=79-93 protocolclass=mfcr2 The teleco who provides the links E1s is Brasil Telecom, I use the protocolvariant as shown in voip-info.org: Brasil Telecomprotocolvariant=br,20,4 But I have a question in relation to variable: protocolend=co I was using =co and others configs I saw are using =cpe. I have change it, but don't seams to have effect to me. I read something on the internet which suggested changes in the file mfcr2.cto correct variables of timing. I believe that that should be the way to solution, but I do not feel safe to do this changes. Some research later, I saw information that in future versions of libunicall would not be necessary to rebuild lib to change parameters of timing, but I believe that's not implemented yet. How I can set a time of increased response of Seize ACK ? Thank you ! 2007/12/11, Moises Silva [EMAIL PROTECTED]: Roger, The seize ack timeout problem is because libmfcr2 is expecting a response ( an ACK ) from the far end and it does not arrive in a R2 variant dependant amount of time. Which protocolvariant do you have configured in unicall.conf? This is how the process to start a call goes: 1. When you Dial(Unicall/blah), Asterisk will ask chan_unicall.c to initiate the call. chan_unicall will ask libunicall to start the call, and libunicall will ask libmfcr2 to start the call. 2. libmfcr2 will set the ABCD bits to 0x0 (000) ( normally the ABCD bits are in Idle 1001 ). Setting the ABCD bits to 0x0 is our way to tell the far end ( the telco ) that we want to start a call, this is known as the Seize. 3. The far end should detect this bit pattern change and answer with a Seize ACK ( ABCD bits in 0xC ), in this case, libmfcr2 does not receive the expected ACK in 2000ms unless you are in Kuwait ( 4000ms ) or Nigeria (1ms ). So, let us know your R2 variant, probably your country require more time to wait for the Seize ACK. Regards, Moisés Silva On Dec 11, 2007 7:03 AM, Roger C. Beraldi Martins [EMAIL PROTECTED] wrote: Dears, After having set up the board Digium TE420 to receive 3 E1s, I can receive calls without difficulties. As you can see in the log below: -- Executing [EMAIL PROTECTED]:1] NoOp(UniCall/14-1, Catch-All DID Match - Found 5908 - You probably want a DID for this.) in new stack -- Executing [EMAIL PROTECTED]:2] Goto(UniCall/14-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing [EMAIL PROTECTED]:1] Set(UniCall/14-1, __FROM_DID=s) in new stack -- Executing [EMAIL PROTECTED]:2] GotoIf(UniCall/14-1, 0 ?cidok) in new stack -- Executing [EMAIL PROTECTED]:3] Set(UniCall/14-1, CALLERID(name)=4133602900) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(UniCall/14-1, CallerID is 4133602900 4133602900) in new stack -- Executing [EMAIL PROTECTED]:5] Goto(UniCall/14-1, ivr-3|s|1) in new stack -- Goto (ivr-3,s,1) *snip* -- Executing [EMAIL PROTECTED]:10] BackGround(UniCall/14-1, custom/celia) in new stack -- UniCall/14-1 Playing 'custom/celia' (language 'br') -- Executing [EMAIL PROTECTED]:1] Hangup(UniCall/14-1, ) in new stack -- Hungup 'UniCall/14-1' -- Unicall/14 released Now I am having problems to make calls using the libunicall. The problem is clear in this line of the full log: [Dec 11 10:03:54] ERROR[12935] chan_unicall.c: Unicall/1 protocol error. Cause 32776 Searching for the error I discovered it is Seize ack timed out, but I do not know exactly of what it means or how to fix it. Here is de version of softwares/libs I have use ( http://www.voip-info.org/wiki/view/Asterisk+MFC+R2). asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 zaptel-1.4.4 I already try using asterisk 1.4.10 but the comportment is the same. I don't believe the problem is in asterisk. I think my configs are correctly but not sure. Attached in text file follow the tests I have done using testunicall, config files from zaptel.conf and unicall.conf I using on this solution. More logs is in the same file. This can be caused by a problem with signaling between my settings and the standard of telephony service ? I'm using FreePBX with a Custon Trunk (Custon String Dial: UniCall/g1), my extensions_aditional has the OUT_3 = AMP:UniCall/g1 and OUTMAXCHANS_3 = 10. Someone has already gone through a problem like
