Re: [asterisk-users] HP DL380 G5 with TE420
On Thu, Feb 26, 2009 at 11:25 AM, Hans Konings h...@konings.nu wrote: I've tried with a different server (HPdl320g5p) and the card is detected in this but the cards generate NMI errors on many bootups. Does anybody have this combination of hardware working? Or can anybody think of something I've missed? We have it working on a dl320, several ML350's, ml310, but never tried on a dl380 yet. We had serious issues in the past when iLO was enabled on a 350. disabling iLO on that machine helped us. (we had irq errors) You could try disabling iLO, just to make sure. regards, stoffell ___ -- 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 tones mid conversation
On Thu, Feb 26, 2009 at 6:08 PM, Simon Dixey simon_...@hotmail.co.ukwrote: I wonder if anyone is able to offer any [polite ;-)] words of wisdom?? I can be polite, I'm not sure about the wisdom .. :-) DTMF threshold in misdn-init look high doesn't it... I'm not entirely sure what it should be set to, to be honest.. (min-max values for tuning); have read max value is 100, but others suggest it'll go higher - but what exactly is it tuning the sensitivity value of specifically? (i.e. what is the threshold value). DTMF detection works well for *genuine* DTMF digits dialled over the ISDN trunk, but mISDN/Asterisk still recognises them incorrectly at times during calls (to/from cell phones). I'm using almost the same setup but I'm in Belgium. Also a b410p. The dtmf seems to get triggered more by some calls then other. It also depends on the voice. (higher tones trigger dtmf more easily) My dtmfthreshold is set to 100. Guess it's the (in)sane default ? :-) Oh, are you having random crashes on your mISDN setup too ? I've also seen other posts refer to settings in Dahdi.conf (such as relaxdtmf) - surely Dahdi doesn't have anything to do with this if I'm using chan_misdn?? Correct. Only has anything to do with it if you're using chan_dahdi. I can't reproduce the same behaviour on dahdi. Is anyone able to confirm exactly whether mISDN's hardware DSP and driver is responsible for detecting DTMF, or whether it's Asterisk analysing the inbound audio? Scanning the README.misdn (sourced separately) the chan_misdn driver readme comments a feature as DTMF Detection in HW+mISDNdsp (much better than asterisks internal!) - so surely DTMF is recognised and passed on by mISDN to Asterisk. The fact that the log messages prefixed by P[ 1] are mISDN - I think I've answered my own question there... Yes, it's mISDN that detects the dtmf. Prior to going down the mISDN route, I looked at Dahdi as the Dahdi configs mention native Dahdi B410P support. But, the conclusion I came to (although what I read didn't make it clear me) is that the readme was referring to Dahdi B410P support in Ast 1.6, not 1.4. That sound right? Dahdi readme: Right again. i've been experimenting with Dahdi's b410p for a while now, it's only available in asterisk 1.6, with the latest libpri and dahdi releases. It's much cleaner, imho, but I'm having issues with receiving faxes when using dahdi, so I'm stuck with mISDN for the moment :-) Enough reading.. if you're still awake! Any help would be very much appreciated. Nice to see someone else is using the same setup. I was beginning to think that people with BRI stopped using asterisk in Europe :-) Keep in touch or post to the mailing list if you have any further news/experiences.. cheers, stoffell ___ -- 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] asterisk 1.4.23.1 and mISDN 1.1.8 segfaults
hi all, I'm have a bit of a hard time with some segfaults on running 1.4.23.1 and mISDN 1.1.8. I already enabled DONT_OPTIMIZE and DEBUG_THREADS in asterisk so I can now generate a bt. I did that (following the instructions on voip-info) but I'm not sure how to read' the output now. By looking at the bt below, can one see if the problem is caused by something special ? I'm running debian etc with kernel 2.6.18-6-686. Core was generated by `asterisk -vg'. Program terminated with signal 11, Segmentation fault. #0 0xb7dfb5f7 in malloc_consolidate () from /lib/libc.so.6 (gdb) bt #0 0xb7dfb5f7 in malloc_consolidate () from /lib/libc.so.6 #1 0xb7dfd236 in _int_malloc () from /lib/libc.so.6 #2 0xb7dffd56 in calloc () from /lib/libc.so.6 #3 0x08118697 in _ast_calloc (num=1, len=1420, file=0x8157ed4 manager.c, lineno=2432, func=0x81597b9 accept_thread) at /srv/software/misdn-asterisk-20081107/asterisk-1.4.23.1/include/asterisk/utils.h:358 #4 0x080cb2ea in accept_thread (ignore=0x0) at manager.c:2432 #5 0x0811a025 in dummy_start (data=0x81acc70) at utils.c:856 #6 0xb7f31c51 in pthread_start_thread () from /lib/libpthread.so.0 #7 0xb7e537fa in clone () from /lib/libc.so.6 (gdb) q Thanks in advance for any pointers. cheers stoffell ___ -- 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 wcb4xxp and fax
Hi all, I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Our setup uses a digium b410p card with asterisk 1.6, latest libpri and dahdi, hylafax with iaxmodem, and all this on 1 machine. chan_dahdi.conf contains: faxdetect=both When receiving a fax call, hylafax (iaxmodem) answers the call after the obligatory wait of 3 seconds (fax detection) but to me it seems that echo cancellation is still being done. Any pointers on this or workarounds? We're back to our old misdn setup for now ;) Here's some output from dahdi show channel 1 (the one that had the fax connection going), i cut out some non-related stuff : *CLI dahdi show channel 4 Signalling Type: ISDN BRI Point to Point Owner: DAHDI/4-1 Real: DAHDI/4-1 Callwait: None Threeway: None Confno: -1 DSP: yes Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: yes Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently ON PRI Flags: Call PRI Logical Span: Implicit Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Regards, stoffell ___ -- 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 wcb4xxp and fax
On Wed, Feb 25, 2009 at 5:28 PM, Steve Underwood ste...@coppice.org wrote: Lee Howard wrote: Make sure that you're using the latest mISDN drivers. Even the latest mISDN gives variable results. Some people say its OK. Some people say its hopeless. It probably varies with the machine its running in. the whole point is I wanted to move away from mISDN (for other reasons) to the digium-way so I can use native digium (and only digium) software. :-) regards, stoffell ___ -- 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] Request for testing of new driver for B410P Quad-Port BRI
On Tue, Nov 11, 2008 at 11:54 PM, Shaun Ruffell [EMAIL PROTECTED] wrote: If you are a user of the B410P card, and are able, please test these release candidates in your environment. To test you will need version 1.4.4 or greater of libpri and version 1.6.0 or greater of Asterisk. Shaun, this is great news! I have been testing the B410P driver today on a test machine, running asterisk 1.4.22, dahdi from svn. Do I really need to run asterisk 1.6 ? It seems to work as it should.. As of now it seems to work well on this test server. (Debian Etch, 2.6.24-686 kernel, HP ML115) I will try to get it into a small production environment asap and keep you posted in case of issues. cheers ___ -- 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] bristuff qozap support for beronet cards
Hi all, In the changelog of bristuff, as of version 0.4.0test4(test5) the beronet cards should be supported. Can anyone confirm if the beronet 2,4 and 8 ports version are supported by qozap now? Regards, stoffell ___ -- 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] HP proliant and hpasm
On Feb 10, 2008 2:01 AM, Steven [EMAIL PROTECTED] wrote: Is anyone successfully running asterisk on an HP proliant while using their management software, hpasm? I have two DL360's and two TE220B's. The cards have their own IRQ's. No matter what combination of settings I use, the cards fail the patlooptest if hpasm (ver 7.9.1) is running. If I stop it the cards pass the test. Hi there, we do run the hpasm on the HP Proliant servers without any problem. We had some issues a while ago with ML350's that kept giving problems, IRQ misses, red alarms, dropped calls, etc.. Everything 'looked' fine (no irq sharing etc..) but the problem was related to iLO. Disabling iLO made it all work.. So if you have issues, try that for starters.. What distro and versions are you using? cheers, stoffell ___ -- 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] Hardware needed
On Feb 13, 2008 10:15 AM, voip crazy [EMAIL PROTECTED] wrote: Someone made a similar instalation? which hardware (server) did you use? Which was the processor type and the amount of memory used by the server? You will probably get some useful info on the list but also check out voip-info.org: http://www.voip-info.org/wiki/view/Asterisk+dimensioning http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations cheers, stoffell ___ -- 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] Asterisk and fax
I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). We use (at multiple sites, mostly BRI) iaxmodem and hylafax. Extra bonus: you get all the cool features and possibilities of hylafax! ;-) cheers, stoffell ___ -- 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] How to hookup to cell phone for outbound calls?
On Feb 5, 2008 9:10 PM, Ed W [EMAIL PROTECTED] wrote: I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. What's your budget? You could use voiceblue's SIP/GSM gateway (exists in 2 or 4 channels), it connects to your internal NIC of your small pbx (like in: laptop?) or by using a switch.. cheers, stoffell ___ -- 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] wireless VOIP phone recommendations?
So any recommendations for another wireless VOIP phone? As someone else pointed out, the Siemens C450 IP (and higher models) work great! Also, the snom m3 gets some good reviews and will be the next one I'll try out.. cheers, stoffell ___ -- 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] Asterisk 1.4 and ISDN-BRI support
Has anyone been able to get ISDN-BRI support to work reliably on Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, kernel, modules, versions, config files). Maybe your best bet is using bristuff, the bristuff-0.4.0 series are tests for asterisk 1.4, I haven't tested them out yet. ( http://junghanns.net/downloads/ ) mISDN however is the other option.. cheers ___ -- 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] Which IP Phone is really the best?
I'm in Europe (yeah, that does matter when choosing a good phone!) .. Some of my (and my customers') favorites: - Polycom (pretty much all of them) - Thomson ST2030 - Siemens Gigaset C450 IP dect (for wireless phones) cheers, stoffell --- http://www.electromarket.be ___ -- 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] Polycoms lose registration and won't re-register
Maybe you could do a test with : a; using the latest polycom administration guide (examples found on voip-info.org) to supply configs and firmware b; use latest firmware (2.1.0.something) c; if the issue doesn't go away, try ethereal between a 'misbehaving' phone and the switch to see what the phone is actually sending out.. (in the mean time use sip debug on the asterisk CLI) cheers.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
I also think you get what you pay for and I don't use hfc based isdn cards in production any more. Having said that, a small home installation isn't quite the same as a 30 user office environment. My home-pbx for example is quite happy reloading asterisk+zaphfc every night. Of course not something I'd accept in a production environment, but that's probably not what HFC-s cards are aimed at either, right? Imho, the bri cards based on hfc chipsets work very well.. I too have some problems with bristuff, but I don't think the card is to be blamed here.. The mISDN way looks like a better way, the BRI cards of digium also use it.. And they're not cheap.. So I guess it's time (for me at least) to reconsider the setup and try out the mISDN-way.. It's no problem to use mISDN with the quadbri of junghanns or beronet, they're all hfc-based.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bri Card for Asterisk ?
On 9/15/06, Noc Phibee [EMAIL PROTECTED] wrote: a small question: what is the best card for Asterisk for supply 2/4 BRI access to a old PABX ? A good bri card is the quadbri of Junghanns/Beronet or Digium (haven't tried the Digium one, but seems interesting because of the on-board echo can..). cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] non-technical, dealing with users giving feedback
Hi list, Any suggestions on how to deal correctly (socially and technically) with users complaining about features/issues? For instance, users complaining about echo; personally I ask the user(s) to give me all the details when reporting echo (like; using handset/speaker, internal/external call, volume, location, time, phone number, ..). This goes well the very first time, but the users (and I understand that) forget some details, or after a few times, don't give the details anymore.. Result, troubleshooting gets much harder. Is there a good way to deal with these situations without annoying the users or myself too much? Thanks for the tips.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] non-technical, dealing with users giving feedback
Search Daily Asterisk News for echo: Yes, that's for the issue with echo, but I was more or less meaning the social side, the communication with the users.. echo was an example.. :) (bad choice maybe? :)) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Firmware
On 9/13/06, Forum [EMAIL PROTECTED] wrote: Unfortunately they pointed me back to Polycom and I have not yet heard back from them. Can somebody post a link to download sip2.0.1? If they point you back, report that to Polycom, they'll contact the reseller (if it's an authorized reseller, that is) to give proper support. Or hang out on irc in #asterisk, I have been told sometimes links pass by... :) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Firmware
On 9/13/06, Forum Expansive [EMAIL PROTECTED] wrote: What is the latest polycom firmware and where can I get it? sip2.0.1, ask your reseller, they must give it to you. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can not hear the telco System Announcement
On 9/1/06, Xue Liangliang [EMAIL PROTECTED] wrote: Hi, all. I am from Singapore, we deployed a few PABX based on Asterisk. Here in Singapore there are two Teleco providing E1 pri service, we encountered a strange problem : when calling a number that is unavailible or line suspended, one of the E1 provider keep the call ongoing, because there are system announcement like The line currently I have something similar on a european E1. I do think this has something to do with the PBX.. (asterisk in this case) I have the same 'issue' on a BRI (ISDN) interface. The 'old' PBX (a classic PBX) did sent out the telco announcement. I have tried changing priindication, but this didn't help. I can see the hangup_cause and can play prompts according to the hangup_cause, but I would prefer using the telco announcement. cheers.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] telco error message on PRI and BRI
hello, Since a few days I'm looking for the 'best' way to get the telco error messages when dialing wrong/busy/non-existing numbers. I can't get it to work on E1 or ISDN BRI. An alternative option is to detect the hangup_cause (no problem here) and play our own voice prompts. I would like to avoid this to make sure the users experience the same behaviour as before. (with the 'traditional PBX') I have tried changing the priindication setting (tried inband,outofband and passthrough) but this didn't change anything. Does anyone have any idea as to how I can debug this? cheers, stoffell ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can not hear the telco System Announcement
On 9/5/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Have you tried progressinband=yes? As far as understand it, it forwards early RTP (that is, stuff that is received prior to the ANSWER), which might just do the trick. Hm, I have just added this in zapata.conf and sip.conf, and also tried the other values (no, never) but neither of one worked. I can find out the hangup-cause but the telco's message is not played back to me. (line gets dropped and hangupcaused is available, but that's it) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redundant Ethernet
On 7/20/06, shadowym [EMAIL PROTECTED] wrote: their Asterisk server? What I would like it to do is use both ethernet controllers on my motherboard so that if one fails the other one takes over. I don't see anyway to make it work seamlessly with 2 IP addresses it would here are some url's to look into: http://www.debian-administration.org/articles/350 http://www.debian-administration.org/articles/163 cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] qozap w/ 1.2.9.1
On 7/7/06, mike [EMAIL PROTECTED] wrote:
someone deployed Junghanns's 0.3.0-PRE-1q (* 1.2.9.1) with a quadBri
card on a production system ?
drawbacks ?
Not really... But be sure to test if you don't have the hangup bug.
(call your cell phone, don't pickup, just hangup as soon as its
ringing, and see if it keeps ringing or instantly stops ringing)
If it doesn't stop ringing, apply the patch:
Replace in libpri/q931.c the first block of text with the second
block. (just comment the first block)
if (c-peercallstate != c-sugcallstate) {
pri_error(pri, updating callstate, peercallstate %d to %d\n,
c-peercallstate, c-sugcallstate);
c-peercallstate = c-sugcallstate;
if (c-ourcallstate != c-sugcallstate) {
pri_error(pri, updating callstate, ourcallstate %d to %d\n,
c-ourcallstate, c-sugcallstate);
c-ourcallstate = c-sugcallstate;
cheers
___
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Re: [asterisk-users] Zap Channel not hanging up on Telco side
On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote: I'm having an issue where Asterisk hangs up a call (either party hangs up) but the telco side of the T1, both the local company and ATT, does not receive the hangup signal from Asterisk. Therefore Asterisk thinks the channel is available but it's still off-hook on the telco I have not experienced this on a standard asterisk yet, but I did on a bristuff version (the latest 0.3.0pre-1 series, being n, o, p and q). However, there's a patch to libpri for this in the mailing list. Are you using the bristuff'ed version? cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with Junghanns Quadbri
On 5/31/06, Jean-Louis curty [EMAIL PROTECTED] wrote: I does nothing special, no output, nor error, same.. .:-( you should at least get any output from ztcfg, but aside from that, like Tommaso said, you must also set the correct jumpers. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff hangup issue
hi, Just wanted to inform everyone, if you're using the latest bristuff's you might (depends on the country!) have hangup issues. The issue appears every time you dial an external number, and hangup after letting it ring for a few times. Then the remote party keeps ringing. In some situations (we only encountered this while dialing to other * servers) it keeps the line open on the telco-side. Meaning.. you pay for it! The cdr on the calling asterisk (with the bug) doesn't indicate a long connection time. However, the cdr on the called asterisk does.. (I've seen several durations of over 20 hours) A show channels doesn't indicate any active calls. A quick fix has been posted a while ago by Marcel van der Boom (in libpri/q931.c), this works. According to the release notes this should have been applied to the latest bristuff, but be careful, the problem still exists on bristuff-0.3.0-PRE-1q. I have emailed junghanns.net to let them know. Best regards, stoffell ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff hangup issue
On 6/29/06, Olivier [EMAIL PROTECTED] wrote: I have emailed junghanns.net to let them know. Did they acknowledge the issue ? I didn't get any reply yet. (but I'm used to that ;)) But yes, the -q release CHANGES file contained this: - libpri fix for P2P BRI in Belgium But the bug still exists (at least in Belgium on ISDN), before it also existed in The Netherlands if I'm correct. The issue appears every time you dial an external number, and hangup after letting it ring for a few times. Is it really every time ? Yes, on calling cell phones, other * servers, fixed PSTN's, .. Just want to inform and see if anyone else is 'infected' :) , we had an extra bill of 300€ last month due to this error. (in the beginning we didn't even knew the error was there..) Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suggested Phone
On 6/28/06, Forrest Beck [EMAIL PROTECTED] wrote: So far we have a Grandstream 2000 Cisco 7912 Very good phone but not so big display. Polycom SoundPoint IP What model? they recently released an alternative to the 501, being a 430. Looks promising. And we are looking at getting a Linksys SPA-942 My current price-wise favorite is the thomson st2030, good hardware quality for a decent price. Better then GXP-2000. (combine a plantronics headset with the st2030 or a polycom, that's all you'll ever need ;-)) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_sms not working anymore
Hi, I have been using app_sms for a few weeks now, since I recently upgrade to asterisk 1.2.9.1 (latest bristuff, -q) however, app_sms doesn't seem to work that well anymore.. On receiving an sms, I execute the app_sms script, and get this as output: -- Accepting voice call from '171701' to 'ournumber' on channel 0/1, span 2 -- Executing Goto(Zap/4-1, custom-smsrx|sms|1) in new stack -- Goto (custom-smsrx,sms,1) -- Executing SMS(Zap/4-1, asterisk-20020-1151573913.346|a) in new stack -- SMS TX 93 00 6D -- Hungup 'Zap/4-1' For some reason there's no SMS RX after the TX, can this be a bug, or is this telco related? cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_sms not working anymore
On 6/29/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I thought that this was me going mad. I'm trying to use SVN trunk and have exactly the same problems. So, I think it's a bug. Can you confirm sending out works fine? I send out an SMS without any problem, on receiving however, I have that error, and I also think the telco side thinks the delivery is okay. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
On 6/1/06, whois wes [EMAIL PROTECTED] wrote: we're running fedora core 4 with the stock kernel, but a move to either CentOS, Debian, or just running Pound Key might be in the works. i currently have the following options added: vga=normal nmi_watchdog=0 acpi=off i had also tried noapic and leaving the nmi_watchdog flag off entirely, didn't seem to make much difference. For what it's worth, we've been running a 2850 with a TE110P on Debian Sarge (with 2.6.15 kernel from etch) without any issues. (we did make sure the TE110p was in a PCI slot without IRQ sharing by disabling the 2nd on-board NIC) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
On 5/31/06, whois wes [EMAIL PROTECTED] wrote: random server lockups, we moved 4 of the servers to Sangoma A104d's last week. The problems we were having have basically disappeared. Keep in mind, this is after trying PCI based NIC's and ensuring there were NO IRQ's being shared, messing with APIC, ACPI, irq latency, and a host of nice to see your feedback! looks promising. however, would you like to share the *, libpri and zaptel versions you're running on these servers? cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recent debian packages?
On 5/30/06, Stefan Reuter [EMAIL PROTECTED] wrote: Are there any apt repositories which provide newer versions of the software? sure: http://pkg-voip.buildserver.net/debian Hi Stefan, very nice. A related question, is there any way you could share the process of how to create the asterisk debian packages? (or maybe even share it through http://wiki.debian.org/) Thanks in advance for any feedback! Best regards, Kristof. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
On 5/18/06, Remco Barende [EMAIL PROTECTED] wrote: Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy a dual port ethernet adapter which will use only one irq to free up an IRQ on another slot. This just totally sucks and irq sharing in a box with only 3 pci slots is totally unnecessary Because we only needed 1 NIC, we disabled the 2nd onboard NIC. That made 1 pci slot free of IRQ sharing, making the system stable and performing very well. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad BRI card
On 5/17/06, Hadley Rich [EMAIL PROTECTED] wrote: They do, but it isn't released yet. Put B410P into google and you will get a couple of hits. Digium's marketing page says it is available and the distributor I use had one on show the other day so they can't be too far away. Aside from being available.. What driver does it use? Will it be needing bristuff ? (that wouldn't work I guess) Or will the near future integrate BRI ( and hfc?) drivers in asterisk? And thus, making bristuff obsolete? (wich means, BRI users will be able to use cvs easily..) Just to make clear I'm very curious on this card. And yes I'm in europe ;) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems
On 5/17/06, Marcel van der Boom [EMAIL PROTECTED] wrote: We had the exact same problem. It started happening for us starting at the 'k' release of bristuff (i mailed a msg on it in february i think to junghanns). Marcel, thanks. This does seem to work indeed! I just tested this on our bristuff 0.3.0-pre1p, works perfect. Thanks a lot! I will also forward your mail to junghanns support. cheers! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic!
On 5/16/06, Edu [EMAIL PROTECTED] wrote: We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb RAM. It was working 24/7 without any for a month, but for not related causes I Just for your info, I have experienced the same issue (just once) on a Dell PE 2850 also. Same error (10 retries, and on and on..) but the machine didn't freeze. Asterisk hung by that time, and had to be killed (-s 9), before we could start it again. After that, it ran good. It's been a week or 2 now, haven't had any issues yet. (* 1.2.7, zaptel 1.2.5, libpri 1.2.2) If you get any feedback, please share with the list.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuffed Asterisk: Hangup problems
On 5/15/06, picciuX [EMAIL PROTECTED] wrote: have you tried EXPLICITLY disabling busydetect? It could cause confusion on digital (BRI PRI) lines... If you have busydetect=yes in previuos channel definitions, it will be inherited by your BRI channels also. hi, thanks for the tip, but busydetect is set to no already, for the same reason as you're suggesting. still, thanks for reminding. I have contacted junghanns.net, hope to get some solution soon now. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuffed Asterisk: Hangup problems
On 5/11/06, Tim Robinson [EMAIL PROTECTED] wrote: There is a lot of junk in your zapata.conf that you do not need, as it relates to analogue lines. This might be causing confusion? I have tried a similary config to yours, doesn't helps. I haven't got this problem on an E1, just on the newer bristuff'd packages. I have sent an email to junghanns.net about this, but haven't received an answer yet. If I do receive anything, I'll post it back to the thread. If there are any other things I might check, please let me know.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuffed Asterisk: Hangup problems
On 5/9/06, Jeroen Zwarts [EMAIL PROTECTED] wrote: I have a problem with the Bristuffed version of Asterisk. I have tried Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have the same problem it seems: Hi Jeroen, any progress made yet? I noticed I'm experiencing the same problem. I wonder if every bristuff user has this issue or it has something to do with the zapata configuration.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards, so disappointing !
On 5/2/06, Wayne Gemmell [EMAIL PROTECTED] wrote: Opened pseudo zap interface, measuring accuracy... This may be a stupid question but how did you do this? in your zaptel source dir (after making..): ./zttest -v or search for zttest on voip-info. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards, so disappointing !
On 4/30/06, Remco Barende [EMAIL PROTECTED] wrote: I e-mailed Dell support and asked them if it is possibel to assign a unique IRQ to one of the three PCI slots. Their reply was, not possible, you are ALWAYS sharing IRQ's, I guess this is the reason for the poor results I'm seeing. If you're using a 2850, you can disable the onboard NIC2, it's sharing its IRQ with 1 PCI slot. Just find out wich one. The other PCI slots share with 2 devices. By doing this I have also gotten very good results on a Dell 2850. cheers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuff 1.2.7.1?
On 4/29/06, Vidar [EMAIL PROTECTED] wrote: Has anyone managed to add the bristuff patch to 1.2.7.1 successfully? My attempts has ended up bad, so if anyone has a working patchfile for 1.2.7.1 I would be grateful to receive it. Have a look at this URL: http://www.junghanns.net/downloads/ You can download directly the most recent package, this is now bristuff-0.3.0-PRE-1o.tar.gz. (I believe it's 1.2.7.1) The homepage doesn't always reflect the most recent file in the downloads dir. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] replacing step-by-step giving echo
hi there, We just encountered the following.. a customer has a tradifional PBX that runs next to asterisk. Both PBX's have their own E1 line. Now 'some' numbers are forwarded from the traditional PBX to the new asterisk server. (both have different DID numbers assigned) When those numbers are called, basically the call arrives on the traditional PBX, gets forwarded to the new number, meaning it goes out the same E1 again, to the new E1 (different telecom operators). Those calls are encountering echo most of the time. When dialing into the E1 on the asterisk server directly, all goes well. (no echo) Should I look into asterisk, the traditional PBX, or even the telecom operator? cheers.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Restart and Dropped calls
On 4/24/06, Chris Gamble [EMAIL PROTECTED] wrote: We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully maybe this is related: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf (search for resetinterval) Please feedback if it 'is' related, i'm curious to know if it helps.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
On 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. James, 24 ports PoE on a Linksys switch occupied with Polycom IP 501's, no problem. The Polycom's use approx. 4W, so the 7.5W per port (if you use more then 12) is no problem at all. Agreed, a more expensive switch (Cisco,HP,Alcatel) might be better, but the Linksys is worth the money! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
On 4/11/06, Begumisa Gerald M [EMAIL PROTECTED] wrote: Again, if the IO-APIC is reporting that the card is on its own IRQ, it really, truly, honestly *IS* on its own IRQ. The reason that it is suggested to disable the IO-APIC is that on many low-end systems, Allow me to comment that Digium actually recommends turning off APIC and using lspci -vb to troubleshoot this kind of shared-interrupt problem. Interesting. Now 'why' do they suggest it, is it because older IO-APIC are 'broken' on some boards? I'm very curious as to 'why', because that would give everyone a better idea on what to look for when having this kind of problems. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
On 4/17/06, Alex Mosburger [EMAIL PROTECTED] wrote: -) * needs to listen to DTMF tones during the call (for transfers or any other features) Does this mean you cannot do any blind or attended transfer? or only the # transfer option (asterisk built-in, from features.conf) doesn't work? cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hardware for new office suggestion
On 4/17/06, Simone [EMAIL PROTECTED] wrote: look at the wiki and the phones suggested, we'd definitely like phones with internal ethernet switch and PoE capable, I'll try to get an idea of what could work for us. I just have a few suggestions on the phones.. First of all, try using 1 model for everybody. This makes life much easier in case of upgrades/configuration/central provisioning, etc.. Some phones I have used (with success) are: Polycom 501, Thomson ST2030, Cisco 7940/7660. cheers and good luck! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX in Production systems?
On 4/15/06, Min Hwan Chang [EMAIL PROTECTED] wrote: wondering if its stable enough to use. Currently I'm editing my own *.conf Using it at multiple sites (ranging from 10-50 extensions). scripts but it sure would be nice if there were some sort of web interface for other people to use. The only thing holding me back is the stability of the FreePBX package... Any comments on this? Thanks in advance. The stability ? freepbx is a web front-end to generate your config files. the config files it generates (most importantly, your dialplan) are well constructed. the setup is 'general', meaning you can use the freepbx-way to serve many different purposes. if you want a very slick dialplan suited for a specific environment, you might (!) be better of writing your own. (but that means, you're on your own ;)) If all you want to do is 'bring out' a webinterface to let someone edit the extensions, you can generate your own interface, or only use the extensions module of freepbx. choice is great, isn't it ? ;-) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 resource full problems ...
On 4/14/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get resource full on the phone when accessing the directory. No central directory was configured. All phones are flashed with the latest publically available sip and boot image. How weird, what is the exact SIP firmware you are using? 1.6.5 ? And how often does the problem occur? I'm willing to test it out on a bunch of phones also if you can share those details. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5
On 4/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: as mISDN neighter support fax-protocol nor beeing able two work in NT-Mode it's no alternative for bristuff :( I guess http://www.visdn.org/ is an alternative. I haven't used it yet, but will be looking into it for sure.. (use daily builds) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX on Debian
On 4/2/06, Christian Gröger [EMAIL PROTECTED] wrote: need that Zaptel stuff? It always prompted errors so i am now using mISDN -without errors, is there a module for freePBX for mISDN? to use mISDN with freepbx, you can Add custom trunk in the Trunks menu. Anyway, is there a good manual for installing FreePBX on debian? Something with typical debian-errors and stuff? That standard manual is so focussed on Suse :( Try the readme or INSTALL file, in the archive. It basically explains what you need to do. Or join #freepbx on irc. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc on an 'actual' asterisk?
On 3/29/06, Benoit Panizzon [EMAIL PROTECTED] wrote: Isn't there an acutal patch to get zaphfc support in *? You even have 3 possible ways out.. 1; you stay with the current bristuff (a somewhat older zaptel+asterisk, but is this really making a difference?) 2; you use a visdn snapshot (www.visdn.org) 3; you use mISDN (more info on beronet) Howver, zaptel has the most 'advanced' echo cancellation, so be sure to test it out! If you encounter any pro/contra's, don't hesitate to report back.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3Com Phones
On 3/25/06, Daniel Hazelbaker [EMAIL PROTECTED] wrote: We are looking at installing a VoIP system with Asterisk and are currently looking at the line of 3Com phones. Has anybody had success with using the following phones? We need to buy a lot and we don't want to end up with phones that don't work properly with asterisk. I didn't even know 3Com had VoIP phones, I'm also curious on these.. How many phones do you need and what is your budget and features wishlist? cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe freezes machine with Junghanns QuadBRI cards
On 3/23/06, Henning Holtschneider [EMAIL PROTECTED] wrote: I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using the qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running in TE mode, the other one in NT mode. Have you (or can you) tried it with 0.3.0-pre1k ? I will be trying the same tomorrow (I'm using 1 quadBRI) and let you know.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe freezes machine with Junghanns QuadBRI cards
On 3/23/06, Michiel van Baak [EMAIL PROTECTED] wrote: I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using the qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running in TE mode, the other one in NT mode. Hm, how weird. I'm not experiencing this. I'm running bristuff-0.3.0pre-1k (* 1.2.4) on debian sarge, 2.6.8 kernel. cheers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI
On 3/21/06, Dave Cotton [EMAIL PROTECTED] wrote: span=1,1,3,ccs,ami I was going mad with a Quadbri until I changed ccs,ami to ccs,hdb3 and it's been running 6 months now. Dave, nice to read on this, can you explain what was going wrong when you used ccs,ami? And how did you find out about placing hdb3 there? As a quadbri user, I'm curious about this :) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING
On 3/18/06, Watkins, Bradley [EMAIL PROTECTED] wrote: cluster (or clusters, in the case of one site). So there is no NAT, and it is an Asterisk-only solution (at least insofar as telephony software is concerned). I'm just barging in.. This all looks 'very' promising stuff, I'm looking forward to any drafts/further discussions on the list. In the meanwhile it looks like I have to build some test-boxes to start trying this.. :) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How hard to create Asterisk for Compact Flash?
On 3/2/06, mustardman29 [EMAIL PROTECTED] wrote: Flashybrid looks interesting. I remember reading about the Astlinux development environment but have not heard much about it lately. Could not find any links to it anywhere. Now that I have a link I will have to check that out as well. Also interesting, a project that just started: http://live.debian.net/ Aside from that, there's also bootcd for debian. (included in stable/testing/...) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hands-on experience with soft videophone
hello, Has anyone got some real life experience with a good software videophone for windows? (SIP) On looking in the wiki I think the best looks to be the eyeBeam phone (commercial), any other suggestions? (free/commercial) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How hard to create Asterisk for Compact Flash?
On 3/1/06, mustardman29 [EMAIL PROTECTED] wrote: I am aware of Astlinux and the other embedded Asterisk solutions out there? Astlinux is nice but the problem is that when I hit a snag and need to incorporate a patch and what not I cannot do that with Astlinux because I cannot compile my own version. I'm also looking for/testing out some things to do the same. (basically I want to be able to boot from cd, flash or ide-flash) I'm now testing out the functionality of dsslive, wich seems to be a cool framework just to do this kind of stuff.. Very nice is also that it uses unionfs. Have a look at their website (http://dsslive.org/) and if you have any questions or are interested, let me know.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How hard to create Asterisk for Compact Flash?
On 3/1/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Did you try the development environment: http://mirror.astlinux.org/astlinux-devel.tar.bz2 Kristian, this means I could create a bristuff'ed version of astlinux by using this one? Cool! cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BLF not working after reload
On 2/26/06, mustardman29 [EMAIL PROTECTED] wrote: Since I am using two completely different phones it must be my Asterisk configuration. I don't know about the aastra, but on the GXP-2000 this is a bug. (do you run latest firmware? maybe it's fixed in that one) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
On 2/25/06, Chris Bagnall [EMAIL PROTECTED] wrote: It's a fascinating thread, this. So, for all the criticism, I'll continue using cheap switches, recycled Chris, I mostly agree.. In Europe a 'small' business often only counts 2 - 5 persons. When the budget doesn't allow it, the only way one can keep a customer satisfied, is by trying to get the best of both worlds.. Wich isn't always easy, but sometimes necessary. (not all SMB's need a failover server, etc..) However.. When possible, I also try to use manageable switches, high-end hardware, and dedicated servers for important tasks. The idea of having an asterisk server that also handles 'other' tasks during the night (in case the company isn't open 24hrs..) is a good one and one to remember.. Cheers.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
On 2/22/06, Clint Sharp [EMAIL PROTECTED] wrote: I had to drop 1.0.1.12 because it has a serious handset volume issue that seems to cut the handset volume in half. Fix one bug, cause another. True, but the latest (beta, okay, but does that matter?) firmware fixes bot and some other. Please watch the voip-info wiki to check the current status, but it seems to be heading the good way.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] aastra v1.3.1 firmware
Hi there, Is it possible with the new aastra firmware to have distinctive ring support? (the wiki says: There doesn't seem to be any way to have the server request a distinctive ring.) The rest of the features make this sound like a good phone. (price/quality) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: aastra v1.3.1 firmware
On 2/17/06, Gareth Owen [EMAIL PROTECTED] wrote: No, distinctive ring isn't supported in 1.3.1. You only have the option of setting the ring-tone on a per-line basis. hm, okay. is it a feature that will be built-in in the future? or can you say for sure it will not? thanks, cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones. - Is it a good choice (budget limit of 100 Euro/phone is mandatory)? - Can be a profitable business the direct buying of 50 phones (to save other money) or is it a risk? if you've never tried a phone, it's always a risk. I'd advise against buying 'any' 15 phones without first trying at least 1.. However, the GXP-2000 is an okay phone. The Thomson ST2030 however is firmer (almost same price) but doesn't have the BLF and MWI. It depends on what features you need. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Performance differences 64-bit vs 32-bit
On 2/8/06, Morgan Gilroy [EMAIL PROTECTED] wrote: As far as I know there will be no difference. 32bit runs natively on AMD64 chips. The only advantage of 64bit is the extra address space and huge integers I agree, but.. I have recently installed debian with an em64t kernel (it was a Xeon 3.0), the thing we noticed was that booting with the em64t was much faster then with the 386 kernel.. (a difference of 10 seconds if I remember correctly!) Comparing the 'speed/performance' of the software running on it (a mailserver) was not part of our little test, so no idea on that.. cheers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] thomson speedtouch ST2030
On 2/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: linksys 942 doesnt look very competetive anyway. (2 10mbit ethernet ports? who is linksys kidding?) Ouch, indeed, that's really working against the other features this phone has. So the advantage over a 941 is also gone now.. :( Thanks for pointing this out.. cheers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug in bristuff?
On 2/6/06, Conrad Wood [EMAIL PROTECTED] wrote: Please note the spelling of uniqueid. I find the spelling in res_features.c - but only once I patched it with bristuff patches. Does anyone know whether that is a known problem with bristuff? If so is it fixed in a later version? What version of bristuff are you using? Then I can have a look in my bristuff to see if I have the same problem.. Where do I report a bug in bristuff? ;) Check this website to contact the author of bristuff, http://www.junghanns.net cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] thomson speedtouch ST2030
hi there, I saw a page on voip-info about the thomson ST2030 phone. There is not so much info on there, that's why I would like to raise a question here. Has anyone got hands-on experience with this phone? (with or without extension module) I am interested if it can be used (as SIP phone) in a good way with asterisk. (also, do all the functions behave like they should; like Supervision backlighted keys etc..) Seems like a nice alternative to other phones, here in Europe. (because linksys 942 is not easily available in europe, yet) All info on this more then welcome! cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware and network requirements
On 2/3/06, John Jensen [EMAIL PROTECTED] wrote: Can a normal server with Pentium 4 3.6 Ghz CPU Most likely. It'll do 40-50 concurrent 711 to 729 transcodings. Hm, interesting. In the case that you do PRI (or BRI) to G729. How do you calculate this number (40-50) ? Or do you write this number down because of your own experience? I assume when using PRI - G.711, that machine could handle 'much' more? Cheers, Kristof. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User web portal for Asterisk
On 2/4/06, Technical Support [EMAIL PROTECTED] wrote: Is there a web portal available for users to: destar configures you asterisk, but also has a user-login to change some user-settings. http://destar.berlios.de/ cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do we need a QOS switch ?
On 2/5/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We have 10 people on our network and each person will have a SIP phone connected to our Asterisk server. All phones, Asterisk, other servers and users workstations will be using the same network. The question is: would I need a QOS device to give SIP traffic a chance? Our internal network is 100M. We will have a ISDN30 for outgoing calls. No calls will be made over the internet. If you have a fairly decent 100Mbit switch, you'll be fine. I assume you will make up to 10 simultaneous calls, so you can calculate the bandwidth you'll be using. (http://www.voip-info.org/wiki/view/Bandwidth+consumption) When using the G.711 codec, it'll be about 1.5-2Mbps when doing 10 simultaneous calls. If you don't overload your internal network, you'll be fine.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI)
On 1/31/06, John Jensen [EMAIL PROTECTED] wrote: Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Only have experience with junghanns cards, but they are the same.. beronet doesn't use bristuff.. but you can also use junghanns cards the beronet-way.. have a look on voip-info.org, some usefull info on these BRI cards can be found there. cheers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Kirk IP600
On 1/31/06, Giordano Grandis [EMAIL PROTECTED] wrote: [chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_park_call Jan 31 10:31:15 WARNING[19727]: loader.c:391 load_modules: Loading module chan_sccp.so failed! check the chan_sccp homepage, make sure you 'clean up' your asterisk modules and include directories.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface
On 1/30/06, Strain Jer [EMAIL PROTECTED] wrote: I was curious which one is best suited for asterisks. Thanks The 'best' depends on your personal flavor I guess. However I'm impressed by voiceone (http://www.voiceone.it/), didn't use it yet, but will surely look into it sooner or later. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
On 10/27/05, Erick Baum [EMAIL PROTECTED] wrote: We're having a rather serious echo problemusing the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it every once in a while on outgoing calls through the PRI. It's not the speakerphone echo problem, we're running the 1.0.1.12 firmware that pretty much fixes that. It seems like most of the echo cancellation functions are for Erick, we're also using 1.0.1.12, having some echo problems, mostly with in/out going ZAP calls (on quadBRI, w/asterisk 1.0.9), the internal SIP calls seem to work fine. (but you have to make sure your volume isn't too high) Also the GXP-2000 has the annoyingfeature that calls get disturbed when you touch the wire (going from handset to phone). We are having some echo issues but that is due to echo cancellation (we think) on zap calls, I'm now starting a test (by using it for a while) with a cisco 7912 to see if it makes any difference. Please keep me posted, I'll inform you of my experiences.. Cheers. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Terrible echo with Te110P and Adit 600
On 10/24/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yes I did notice it immediately.I intend to tweak more, but for themoment it seems like echo is minimized to zero. I also encountered some echo problems and used (uncommented :)) following parameters in zconfig.h: #define ECHO_CAN_MARK3 (instead of MARK2) #define CONFIG_CALC_XLAW #define CONFIG_ZAPTEL_MMX Up untill now it seems to be much better.. It also 'sounds' much better during normal conversation. Oh, and in the Makefile, changed some flags: KFLAGS+=-DSTANDALONE_ZAPATA -march=pentium4 -O3 -fomit-frame-pointer CFLAGS+=-DSTANDALONE_ZAPATA -march=pentium4 -O3 -fomit-frame-pointer cheers. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Quad BRI with Fedora, anyone?
On 10/15/05, Lars Dybdahl [EMAIL PROTECTED] wrote: It seems that the bristuff source from junghanns.com wasn't written for gcc 4, which is the one included in FC4, and I have seen somedescriptions on making zaphfc compile, but there are more problemsthan just that one. Also, RPMs would reduce the amount of time spenton making this work significantly. What kernel are you using? What version of bristuff are you using? (latest is bristuff-0.2.0-RC8p.tar.gz) Where do you encounter errors, and wich errors do you get? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
