[Asterisk-Users] IAXy's apparantly failing in the field
I am not sure if this is the place for Digium user-to-user discussion, but... We have deployed many (20+) IAXy's in the field. At a couple of locations, the IAXy's have just stopped working after 1 or 2 days use. No lights go on, no DHCP lease is renewed as far as we can tell, and of course no dialtone and no registration with the server. Has anyone else experienced high failure rate with these devices? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.conf qualify=yes not working?
Thanks, In these cases, the IAXy cannot be called by another extension. Yes they are behind a firewall router, but in two cases, just a cheapie Linksys WRT54G (or similar Linksys). I also use a WRT54G and do not have this problem with my IAXy. On Wed, 2005-01-12 at 08:15 +0100, Wilson Pickett wrote: In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because iax2 show peers shows the device as status UNKNOWN. However, when a user picks up the telephone plugged into the IAXy, they can place a call just fine within our Asterisk server. Are the IAXy behind firewalls or routers? It sounds like the message sent by asterisk is not geting through. The IAXy would still know how to call out, but can they be called? If UNKNOWN, I assume not. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax.conf qualify=yes not working?
We have many IAXy devices in the field now. In all cases, in iax.conf, we have qualify=yes, so that using iax2 show peers, we can see whether or not the device is currently online. In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because iax2 show peers shows the device as status UNKNOWN. However, when a user picks up the telephone plugged into the IAXy, they can place a call just fine within our Asterisk server. Can anyone tell me if there are any conditions which might affect the functioning of the qualify feature, while still allowing outbound calls to go through? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 (IAXy) and DTMF Question
In my particular configuration, by the time the IAXy gets DTMF, it's just audio (e.g. not out-of-band in any way). The SIP modems play the audio of DTMF quite nicely, while the IAXy plays it quite warby, thus my DTMF-driven application (which is plugged into the IAXy) can't decode them. Are there codec settings in the IAXy which might do a better job of rendering DTMF as audio? Thank you, Brent On Tue, 2005-01-04 at 01:56 -0500, [EMAIL PROTECTED] wrote: Looks like the IAXy at the originating end converts the audio DTMF into an IAX DTMF message (and strips the DTMF out in the process). Meanwhile the IAXy at the answering end doesn't convert the DTMF indication message back into tones. FYI, Mark implemented DTMF in the latest version of the IAXy firmware that is in CVS head. It will make it's way to the stable branch at some point. Russell ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 (IAXy) and DTMF Question
I am having trouble with a DTMF-based application on Asterisk 1.0.3. Specifically, when two IAX2-based devices are talking, when they send DTMF to eachother, the other side only hears clicks, and maybe a millisecond of DTMF tone, but not any real duration. Furthermore, when one IAXy device calls the Echo test program, we can hear our echo, but when we punch DTMF in, we get the same effect (can't hear it, or can only hear clicks). In contrast, when a SIP device calls the Echo test program, we can punch DTMF all day and hear it echoed back to us. Can anyone tell me if we're doing something wrong? Thank you, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 and DTMF
For efficiency reliability, when SIP transmits DTMF as non-audio data, it uses RFC2833 or INFO. My question is - (not knowing much about IAX2) - when IAX2 transmits DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it using some other IAX2-specific mechanism with its own name? Thank you, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Command-line dialer/recorder for asterisk?
I'm somewhat new to Asterisk and am tasked with having it perform some automated functions. Is there a way with the current system (and/or extra modules out there to:) 1. Launch something from a command line (on the Asterisk server) to: 2. Dial an extension 3. Issue some DTMF sequences, 4. Record the output to a WAV (or GSM) file, and 5. exit Any quick pointers would be greatly appreciated, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP dtmf=rfc2833 not working
We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that the IAXy is digitizing DTMF tones and sending just the pure data, rather than the audio representation, and that this explains why the IAXY's work flawlessly in this application. I am also under the impression that SIP modems should also support a mode like this.. We have tried: dtmfmode=rfc2833 in sip.conf, and we have also tried turning on DTMF Tx: to AVT on the Sipura, but this does not affect reliability at all. So my question is: 1) Are we doing anything wrong, or is there something more we should be doing, to enable DTMF translation (ala rfc2833) in Asterisk and/or our SIP modems? 2) Is there any kind of debugging mode in Asterisk which we can turn on, which will show once and for all whether or not we really have successfully enabled rfc2833? We are using Asterisk 1.0.3, by the way. Thank you very much in advance! Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Optimizing Sipura/Asterisk for DTMF?
We have an application which is primarily DTMF driven (automated on both sides), which we are trying to deploy over VOIP and Asterisk (using some Sipuras and some IAXY's). We are finding that in around half the cases, the Asterisk server can't decode the DTMF digits from the field office (or at least some of them). Though, when we place voice calls for testing, we can hear eachother quite well. I was wondering if there are any settings in Asterisk and/or in SIP clients such as the Sipuras, which will optimize the connections for DTMF rather than voice? Thank you in advance, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to tell Who's Online?
On Thu, 2004-12-16 at 17:41 +0100, Wilson Pickett wrote: if I'm missing something obvious, but I couldn't find any console command to show users online. sip show peers iax2 show peers Thank you, Do you know, if an IAXy device (or anything else speaking IAX2) disappears, how long will it be (minutes, hours?) before Asterisk notices they are offline, and iax2 show peers will reflect the change of online status? Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to tell Who's Online?
We have an Asterisk server online, with many SIP clients (some Sipuras, some laptops), and we're also using some IAXy's. I've been trying to find a simple way to check who's online, meaning who is reachable at the moment, without actually going through and dialing everybody. Is there a way to do this with Asterisk? I am sorry if I'm missing something obvious, but I couldn't find any console command to show users online. Thank you, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
