[Asterisk-Users] Buzzing X100P - call 650.210.9331 to hear it
Can anyone tell me what that strange buzzing sound is? My system is working fine except for this problem with the X100P. After what seems to be a random amount of time after system startup - sometimes hours, sometimes days, the card gets bolloxed and just does that buzzing sound when the line picks up. Restarting asterisk or re-running ztcfg does not make it go away. I have to reboot. Thanks, Sean ___ 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] cvs update and new x100p cards broke menu playback
After struggling with the carrier access channel bank for a few weeks, I finally gave up on it, and got myself three X100P cards instead, for my incoming lines. The plan is to use the channel bank just for internal lines. I installed the cards and at first they were mostly OK, except occasionally they'd "lock up" and stop accepting calls, requiring a full reboot. So I figured I'd update everything including the zaptel drivers (my prev installation of the drivers was about two months old - the asterisk version was about three days old). Now the system appears to be working, with one major flaw: none of the recorded messages (greetings, menus etc) will play back any more - not on the sip phones, and not even on the dial-in lines! The console indicates that asterisk is trying to play the right files, and there are no error messages at all. There's just no sound. However, simple tones will play, and calls between sip phones or through IAX work fine. I figured perhaps some new directive was needed in the conf files, so I diffed all of the sample files against my own but I didn't see anything. Aside from that, I don't even know where to begin troubleshooting this. Can anyone point me in the right direction? Thanks, Sean ___ 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] echo
Read the faq, checked the config files... can't find anything about an echo problem like this. Here's what I've got: 4 channel t1 card, span 2 going to channel bank with both fxs and fxo lines Polycom IP600 phones on same LAN with asterisk iax connection to voicepulse (T1 going out on another router) Asterisk on a 2.4GHz machine what I hear: FXO <-> FXSno echo at all FXO or FXS <-> voicepulse no echo at all IP600 <-> voicepulse no echo at all IP600 <-> FXO or FXS echo heard by IP600 caller, no echo heard by remote party IP600 <-> IP600 can't get it to work yet - SIP times out (separate issue I guess) I also did a few more tests: - I made an extension that just does Wait(). Called it from the ip600. No echo. - Used the built in Echo function. I only hear one echo. I measured the latency by recording it with a microphone. It is 100ms, which seems a bit excessive for ethernet. The fact that I hear only one echo when doing the Echo test, and no echo anywhere except in the IP600 <-> POTS path, would lead me to believe that the source of the echo is within asterisk (not in transmission or in the phones), and only when bridging SIP<->POTS. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Okay, I'm an idiot. The tones are picked up just fine by asterisk with no changes. It helps if you understand the syntax of zapata.conf. I thought busydetect=yes just had to be under the context line. I didn't realize how the "channels=" is actually the delimiter that includes the stuff above it (I had busydetect below that line). I should add that I find the asterisk config files to be very whacky in general. On Jan 2, 2004, at 12:34 PM, Martin Pycko wrote: If the on/off times are diffrent you need to edit Makefile and uncomment BUSYDETECT_TONES_ONLY flag or something like that ... and then you can change the MAX/MIN values in dsp.c too. That should help you with busycount=10 and busydetect=yes regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: Here's a recording: http://www.seanadams.com/hangup_tones.aif (sorry - recorded from speakerphone - skip to the end) The following numbers are not real precise, I just got this from visually looking at the spectrum on my computer: The tones appear to consist of 2600, 2440, 2000, and 1400 Hz. The timing is 120ms on, 80ms off. I'll take a look at dsp.c and see if I can make it work. Thanks for the pointers. On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote: busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf and measure the length of the tone .. should be equal the pause too. Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like this: your result - 100, your result + 100 [ms] regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes "if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc." I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Not having any luck with just tweaking those values. I'm a bit confused still as to how the different busy detection choices are supposed to work - I've uncommented a few of the #if 0 to see if it's doing anything, and I can't see any indiciation that it is. Don't the specific off-hook tones need to be in dsp.c, or is it intended that asterisk should match the signal just by the timing? Here's some information I found which confirms the tones I measured: http://www.hackfaq.org/telephony-27.shtml -- Receiver Off-Hook Tone This tone is used to cause off-hook customers to replace the receiver on-hook on a permanent signal call and to signal a non-PBX off-hook line when ringing key is operated by a switchboard operator. Receiver Off-Hook Tone is 1400 Hz, 2060 Hz, 2450 Hz and 2600 Hz at 0 dBm0/frequency on and off every .1 second. On some older space division switching systems Receiver Off-Hook was 1400 Hz, 2060 Hz, 2450 Hz and 2600 Hz at +5 VU on and off every .1 second. On a No. 5 ESS this continues for 30 seconds. On a No. 2/2B ESS this continues for 40 seconds. On some other AT&T switches there are two iterations of 50 seconds each. - On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote: busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf and measure the length of the tone .. should be equal the pause too. Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like this: your result - 100, your result + 100 [ms] regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes "if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc." I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Here's a recording: http://www.seanadams.com/hangup_tones.aif (sorry - recorded from speakerphone - skip to the end) The following numbers are not real precise, I just got this from visually looking at the spectrum on my computer: The tones appear to consist of 2600, 2440, 2000, and 1400 Hz. The timing is 120ms on, 80ms off. I'll take a look at dsp.c and see if I can make it work. Thanks for the pointers. On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote: busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf and measure the length of the tone .. should be equal the pause too. Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like this: your result - 100, your result + 100 [ms] regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes "if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc." I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Are the tones increasing in pitch? No, the beeps are the same pitch - sounds like it was deliberately designed to be a loud and awful sounding as possible through an off-hook phone, to get your attention to go hang it up. My ears tell me it's roughly 250ms on, 250ms off and so on. Are they the Special Information Tones (SIT) that are also on the message when you dial a number that has been disconnected? No, not like that at all. I'll make a recording. If so, then they are defined somewhere in the code, at least as part of app_zapateller since that is how it tries to get rid of telemarketers. You could then see about adding that to the dsp routines to detect the SIT tones and determine what to do at that time. Taking my first peek at the code now... BTW, which CAC channel bank did you buy? The ADIT 600 should do disconnect supervision, and I thought the AB1 did too. It's the AB1 with 8 fxo, 16xfs. Here's the page I was talking about: http://www.voip-info.org/wiki-Asterisk+hardware Also, others have reported this problem but I can't find a resolution: http://www.mail-archive.com/[EMAIL PROTECTED]/ msg18626.html Are you also sure you have that on your line so as to be detected? Your other option might be to switch to groundstart lines which detect hangup much easier. May be difficult to get unless you are a business though. I just have regular business lines without any special provisioning. I don't understand why a $20 answering machine can do this but an expensive channel bank can't. :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hangup detection
So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes "if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc." I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
As a new asterisk user myself, I would agree with you that the learning curve is steep, but that was my expectation coming into this. I took the time to browse the list archives before signing up so no surprises there. There are some real experts here and they obviously help those who ask interesting questions that aren't answered elsewhere. I would agree that this list would be better without the retarded flame wars, and furthermore, trolls the likes of you. If you don't want to read the information that's available, or if what you expect is total hand-holding - someone else to install and configure your phone system for you, then asterisk is a great choice but you need to hire someone to do that. Or you can go with a commercial phone system and pay thousands for a basic system with 1/10th the features. Regarding the stability problem you're having - clearly that's not the norm. I wouldn't suggest that anyone "expect" that behavior. I certainly haven't seen any crashes. On Dec 31, 2003, at 12:37 PM, Me wrote: As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone as intercom
Wow! Thanks John for the detailed information. This is such an awesome system... and great support here, too. On Dec 31, 2003, at 12:07 AM, John Baker wrote: Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: In this real-world example, whenever I set ALERT_INFO to "Sales" in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: se.rt.8.type="ring" tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer="11" tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"...> and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) The se.rt.4.timeout="1000" tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to "Ring Answer" whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: Hello, It's all dependant upon the firmware of the phone(nothing to do with the PBX or SIP currently). The documentation of the Polycom VOIP phones shows no way of doing this currently but it is really just a matter of Polycom adding this feature to their firmware in the future which we are pushing for. People have gotten this to work with Cisco and Snom phones. MATT--- -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 6:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phone as intercom (new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phone as intercom
(new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio format for announcements
Hi guys. First off, to the folks at Digium: outstanding work. The fact that Asterisk is open source puts you right at the cusp of what will be the most important telecom advance since the transatlantic cable. Anyway... a couple newbie questions concerning sound quality - I don't see any reason why the system should not use the best possible format for any given connection. 1) Is it possible to store the menu sounds in wav/aiff, and let asterisk compress them to gsm only as necessary? Eg for POTS lines, yes the lines are crap already, but why butcher the sound any further by running it through a speech codec? 2) For my internal SIP phones, I don't care about bandwidth usage. What settings will give the best sound quality? Does the protocol (or for that matter, any particular brand of phones) support uncompressed or very high bit rate audio for intra-pbx calls? Thanks, Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
