Re: [asterisk-users] Reliable wireless SIP phones
I you have such a problems with siemens you should consider 8 voip port linksys gateway with dect bases, their gateway is rock solid and cheap. I'd recommend against buying new analog POTS gear myself. We've got a mix of SNOM 300 corded VOIP phones and generic DECT bases attached to Linksys SPA-whatever ATAs. The user experience is much more positive and direct using an actual integrated VOIP phone rather than going through the ATA. SNOM offer SIP DECT bases and I've got one on order now for a proof-of-concept to get the analog DECT bases and Linksys ATAs out of the mix. It's hard to put your finger exactly on all of the little differences that make the actual VOIP phone feel like better quality but I'll try: - can key as fast as you want and see your number and correct as you're dialing (Linksys ATA seems to get confused if you DTMF too quickly and no option to correct) - can hit the checkmark button to dial immediately. Usersfind this more intuitive than hitting # key, more like a mobile phone - callerid with alpha text such as other extensions works properly (although this could be a phone issue with our cheap DECT bases) - easy multi-line support The Linksys ATAs were quick to deploy, and low-cost and low-risk as they let us re-use existing POTS gear with easy fallback to analog, but I wouldn't choose that route for a new install. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] distinctive ring on sipura
Hi All, I'm trying to move some POTS phones from Zap to sipura. I've searched and read a several articles which suggest that something like this should work: exten = 600,1,Dial(SIP/cordless) exten = 600,n,Hangup() exten = 700,1,Set(ALERT_INFO=Bellcore-r2) exten = 700,n,Dial(SIP/cordless) exten = 700,n,Hangup() Unforutnately though 600 and 700 sound exactly the same -- normal US ring. I found this in the sipura config: Ring1 Cadence: 60(2/4) Ring2 Cadence: 60(.3/.2,1/.2,.3/4) Ring1 Name: Bellcore-r1 Ring2 Name: Bellcore-r2 So I think I'm on the right track, but just passing the variable wrong. Here's the console output during the call: -- Executing [EMAIL PROTECTED]:1] Set(SIP/office-081c5970, ALERT_INFO=Bellcore-r2) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/office-081c5970, SIP/cordless) in new stack -- Called cordless -- SIP/cordless-081cba38 is ringing This is all on 1.4.21.2. When the phone is plugged into the zap card, Zap/1r2 definitely rings differently on the phone. Thanks a bunch in advance for any help. Best regards, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
I didn't say because I wanted my original email to limit itself to facts I was sure of, but I think my SIP problems started with 1.4.20 as well. I'm fairly sure 1.4.19 was solid... going back today. It looks like someone at bugs.digium has found what it was, so a fix should be coming soon. PaulH I guess that since there was no mention of this fix in 1.4.21.2 that it's still an open issue? Can you reference the bud ID at Digium so I can follow along? I didn't see it, but might not have known what I was looking for. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
Interestingly enough, I've had my Grandstream suffering from the same problem since I upgraded to 1.4.20, although my config is static rather than realtime. I'd actually written it off to typical Grand-heap-of-$#!+-stream behaviour. :) I didn't say because I wanted my original email to limit itself to facts I was sure of, but I think my SIP problems started with 1.4.20 as well. I'm fairly sure 1.4.19 was solid... going back today. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
Hi David, It may be IAX2 bug, do you use IAX? In my case downgrading back to 1.4.19 did the job. No IAX for me. I don't recall ever having this issue on 1.4.19 so unless I hear any other suggestions as to how to troubleshoot this I'll go back to that version as well. Best regards, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.21.1 SIP failing, requiring reboot
Hi All, Experiencing frustrating SIP failures on 1.4.21.1. Everything works fine, and then randomly after a week or so of use SIP just completely fails to talk to outside providers and nothing short of a reboot fixes it. Symptoms: - both of my SIP providers at the same time become unreachable. Here is the most recent time it started from /var/log/asterisk/messages [Jul 16 05:15:35] NOTICE[27073] chan_sip.c: Peer 'voipplanet' is now UNREACHABLE! Last qualify: 15 [Jul 16 05:15:41] NOTICE[27073] chan_sip.c: Peer 'viatalk' is now UNREACHABLE! Last qualify: 118 [Jul 16 05:16:59] NOTICE[27073] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) [Jul 16 05:17:06] NOTICE[27073] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) and it goes on until I catch it... over 12 hours in this case (it's a very lightly used system) - sip show peers shows them both as unreachable - sip show registry shows them both as request sent - internet connection is fine, I can still ping both servers - reload sip does nothing - restart asterisk does nothing - reboot server and it's fine for another week or two This is 1.4.21.1 on kernel 2.6.25 on a fully updated gentoo system. The system is a proof-of-concept system, Asus board, core 2 duo, 2 GB RAM and extremely lightly loaded. The two SIP providers are totally unrelated and actually on different continents. Any ideas of what I can do to diagnose this problem further? Thanks in advance and best regards, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF Compatible Phones
One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Hi Dayton, It's even easier than that. With an asterisk PBX your receptionist shouldn't be picking up analog lines directly for either incoming or outgoing calls. S/he will be able to sit back and file their nails and answer the phone when it rings, and to dial out just let asterisk manage the lines for that as well. You don't have to worry about anyone managing the lines, let the software do the work. If you fill up all the lines with either incoming or outgoing calls you'll run into issues of course, but then it's a simple case of adding lines to meet demand. You will want to keep track of usage for this purpose, but it need not be actively managed by the receptionist unless you specifically want them to. A small script is more reliable and better for most cases. If you've already got the server set up at home, dig through ATFOT a little more and start configuring a line or two at home and you'll start to realize how the config works. Best regards, David [EMAIL PROTECTED] ___ -- 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] Best Linux distribution to use in Asterisk server
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? I don't know what's prompting you to leave Gentoo but it's gotten much better with respect to asterisk very recently. That's all I use and I have to say that after a very frustrating year they've come forward leaps and bounds in the last few months. They actually have 1.4, 1.6, and versions of zaptel that work with modern kernels now finally! If you haven't already, look into the voip overlay. They seem to work on it in spurts of activity and although not much has changed in it for several weeks it does contain a relatively recent 1.4 version that works just fine for me on my tiny production environment. Best regards, David [EMAIL PROTECTED] Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Do yourself a favor and upgrade a Asterisk 1.4 which has a proper implementation of DTMF. It's likely your SIP provider upgraded to something which does not recognize the DTMF tones from Asterisk 1.2. I've upgraded to 1.4.18 (along with zaptel 1.4.9.2) and still experiencing the same problem (not recognizing DTMF on SIP inbound calls) as well as new problems. The new problems are much more severe than the previous problems so I'm starting a new thread with a more descriptive subject. I've changed sip.conf to eliminate warnings for new syntax: insecure=port,invite dtmfmode=rfc2833; Choices are inband, rfc2833, or info Everything else is as-was in sip.conf, extensions.conf, iax.conf, rtp.conf, voicemail.conf, zapata.conf, zaptel.conf (although I looked through the new samples and didn't see anything glaring I needed to change). For the config files I had not changed I took the new sample files. Now in addition to not recognizing DTMF on SIP still, asterisk is now frequently dropping calls when I start to enter DTMF. On console I get lines such as: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/x-081ea720, incoming|s|1) in new stack -- Goto (incoming,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(SIP/x-081ea720, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/x-081ea720, /home/dnedved/hello) in new stack -- SIP/x-081ea720 Playing '/home/dnedved/hello' (language 'en') == Auto fallthrough, channel 'SIP/x-081ea720' status is 'UNKNOWN' It's also happening on zaptel channels (although not nearly so frequently), so it's not a SIP only problem: [Mar 27 10:42:07] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 18 (Ring Begin)... [Mar 27 10:42:08] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 2 (Ring/Answered)... [Mar 27 10:42:12] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 18 (Ring Begin)... -- Executing [EMAIL PROTECTED]:1] Goto(Zap/4-1, incoming|s|1) in new stack -- Goto (incoming,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, /home/dnedved/hello) in new stack -- Zap/4-1 Playing '/home/dnedved/hello' (language 'en') == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN' -- Hungup 'Zap/4-1' I don't know much about asterisk debugging since it has worked so flawlessly so far, but I would guess that the Auto fallthrough with status UNKNOWN means that the application that was running died, didn't set any return code, so asterisk dropped the call? I'm running in console mode with 5 v's of verbose mode, how do I find more information about why it's dropping these calls? Thanks and best regards, David [EMAIL PROTECTED] Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Try adding this line in the general section of extensions.conf autofallthrough=no The default behavior in 1.2 was no. In 1.4 it changed to yes. That will be your simplest fix (without seeing your dialplan). Asterisk is moving on to the next step in the dialplan before you enter your digits. You need to have it wait for the digits to be entered. Thanks for that. I did see that note in UPGRADE.txt but didn't realize the full importance of it changing the logic of the dialplan. I've got it set back to no and will read the new version of ATFOT to figure out how to restructure my dialplan. So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the most part but completely ignoring DTMF on incoming SIP calls. Best regards, David [EMAIL PROTECTED] Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] DTMF suddenly stopped working on SIP channel
--- Eric Wieling [EMAIL PROTECTED] wrote: Inband only works with the ulaw and alaw codecs. I think you might be onto something here. I don't have any explicit allow or disallow lines, just taking the defaults. I've got plenty of bandwidth and CPU, I'm much more concerned about calls going through. Without knowing what codecs my provider uses and not seeing anything specific in the logs, is there a setting that would be better than default for reliability? I had originally set to inband for outgoing calls because the default wasn't working for dialing into voicemail systems, etc. Switching to inband fixed the outgoing DTMF issue and incoming worked fine for months until earlier this week. Thanks for any suggestions. Best regards, David [EMAIL PROTECTED] Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] DTMF suddenly stopped working on SIP channel
Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them what they changed (fat load of luck getting that question answered anyway). Everything was working fine with dtmfmode=inband and relaxdtmf at the default, now I've tried all 6 valid combos of those two settings with no change. This is on asterisk 1.2.27 that's been working fine in production for about 3 months now. Here's the section from sip.conf (the way it had been working all along): [viatalk] type=peer secret=(yep it's right) username=(yep it's right) host=newyork-1.vtnoc.net canreinvite=no insecure=very qualify=yes context=incoming-viatalk dtmfmode=inband ; Choices are inband, rfc2833, or info ;relaxdtmf=yes ; Relax dtmf handling Thanks in advance for any help. I've got all incoming calls on Viatalk shunted to an extension in the meantime, not an elegant solution. Best regards, David [EMAIL PROTECTED] Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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
