Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sat, 28 Aug 2004, Michael George wrote: So even with X11 eliminated the sound is still bad to Digium. I tried another's 1700 number, and it sounded the same, so it's not something unique to digium and me. Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work with my ISP only giving me 1/2 duplex service? If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. To do that, in /etc/asterisk/logger.conf edit the debug line to be: debug = notice,warning,error,debug,verbose Then run asterisk like so: /usr/sbin/asterisk -vv -g -dd -c Then go iax2 debug at the CLI prompt. Do a test call, then send me the resulting /var/log/asterisk/debug file. THanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sat, 28 Aug 2004, Andrew Kohlsmith wrote: Please note that it seems impossible to disable jitter buffer between 20040806 CVS HEAD endpoints. The jitterbuffer numbers in iax2 show channels look live. The numbers look right (jitbuf 0ms) between 20040806 and RC1 (Nufone). I haven't upgraded since then. The numbers get reported still in the older version, but the buffer IS turned off. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disconnection From IAXTel
I'm using the firefly third-party softphone. However, the same thing happened when I used IAXphone 2.0. On 29 Aug 2004 at 7:13, you wrote: On Sat, 2004-08-28 at 14:01 -0700, Muiz Motani wrote: How do I go about disallowing transfers when I am running an IAX soft phone. Is that setting not at the * server? Obviously, I don't have control over the configuration of the IAXTel * server. Which softphone are you using? -- Dave Cotton [EMAIL PROTECTED] ___ 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 -- Muiz Motani Intelligent Distribution 72-6800 Lynas Lane, Richmond, B.C. V7C 5E2 email: [EMAIL PROTECTED] phone: +1 604 448 9293 fax: +1 604 448 9296 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disconnection From IAXTel
On Sat, 2004-08-28 at 23:36 -0700, Muiz Motani wrote: I'm using the firefly third-party softphone. However, the same thing happened when I used IAXphone 2.0. I can't offer any real solution because I was only testing the connection with Firefly, but I got exactly the same symptoms whenever the Firefly softphones tried to communicate, with or without the notransfer= setting. I blamed it on the Firefly but you say the same thing happens with IAXphone. Once hardphones where in place that problem went away. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disconnection From IAXTel
Dave Cotton wrote: On Sat, 2004-08-28 at 23:36 -0700, Muiz Motani wrote: I'm using the firefly third-party softphone. However, the same thing happened when I used IAXphone 2.0. I can't offer any real solution because I was only testing the connection with Firefly, but I got exactly the same symptoms whenever the Firefly softphones tried to communicate, with or without the notransfer= setting. I blamed it on the Firefly but you say the same thing happens with IAXphone. Once hardphones where in place that problem went away. I believe this was discussed awhile ago, the solution was to set qualify=no - if anyone knows why it's happening, I'm happily fix it in firefly -Adam ___ 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] Mobile phone integration via bluetooth
Has anybody tried to integrate amobile phonevia blutooth in asterisk PBX? I believe the most things needed are just existing in open source. I found a "kbthandfree" (http://docs.kde.org/en/HEAD/kdeextragear-3/kdebluetooth/components.handsfree.html)wich allows to control amobile phonevia an application and use your computer as "headset". You may dial und receive calls. Using a way like that it would be possible to integrate a mobile hone to astrisk (as channel - control and voice data are transported via bluetooth). So it would be possible to receive GSM (mobe phone)calls in the private branch, and even least cost routing using the GSM network und POTS would be possible. Did anyone start an integration like that? Heiko ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXy Power in Australia?
Jeremy Don't bother looking for a PSU with a lower current rating, The IAXy like any electrical device is designed to run at a particular voltage and consumes a certain amount of power (W) at that voltage. In simple terms this means that if the voltage is constant and the design parameters of the device don't change, the device will only draw the amount of current it needs to operate and no more. So a device that draws 500mA at 9V to operate, will only draw 500mA regardless of whether the PSU is capable of supplying 500mA (0.5A) or 5000mA (5A). This is why the electrical supply to your house may be capable of hundreds of amps (A) but your living room light with a 60W bulb only draws the 0.25A it needs to operate. Obviously an underrated PSU will not be able to provide enough current at the rated voltage and so the voltage will drop and the PSU will overheat, resulting in failure of the PSU and possibly the device connected to it, depending on the protection devices in the PSU. An overrated PSU (current wise) will just run cooler and be more capable of delivering the current the attached device needs. This is also why it is vital to have a fully regulated or switched PSU because there is no regulation circuit in the IAXy itself and if the voltage to the IAXy circuit changes so do its power and current requirements, making it unstable. OHM's LAW is a wonderful thing :) Cheers! Roy... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jeremy Bogan Sent: 28 August 2004 22:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXy Power in Australia? Have you tried feeding it less amps at all? Not yet, but i'll see if I can find a power supply with less amps. -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie
I am not sure about vonage but if you go with an IAX provider you can have multiple simultaneous calls to your DID. - Original Message - From: Michael Di Martino [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 28, 2004 6:46 PM Subject: [Asterisk-Users] Newbie I am interested in setting up an Asterisk server as my home phone system. I ultimately want one 10 digit phone number, three extensions, and an auto attendant My current phone service provider is Vonage, I have one line with call waiting. My concern is will I need to add additional lines if I want the auto attendant handle multilple calls. For example a call comes in and the auto attendant sends the call to ext 1. Now while the person on ext 1 Is still conversating can another call be handled by the auto attendant? Regards, Michael Di Martino Director of MIS The Telx Group Office: 212 480 3300 X2022 Cell: 646 207 6603 [EMAIL PROTECTED] -- Sent from my BlackBerry Wireless Handheld ___ 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 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] Termination in Holland.
Hi all, Can you guys recomend a good terminiation partner in Holland ? /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disconnection From IAXTel
So how do I get IAXTel to set qualify=no for me? On 29 Aug 2004 at 17:46, you wrote: I believe this was discussed awhile ago, the solution was to set qualify=no - if anyone knows why it's happening, I'm happily fix it in firefly -- Muiz Motani Intelligent Distribution 72-6800 Lynas Lane, Richmond, B.C. V7C 5E2 email: [EMAIL PROTECTED] phone: +1 604 448 9293 fax: +1 604 448 9296 ___ 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] IAXy died
While I'm waiting to hear from Digium, Has anyone ever experienced this? The IAXy now refuses to ask for an ip. I've plugged it into three different routers with the same result: the RJ11 small LED lights and the internal green one flashes, but no ip. Doesn't seem to be asking DHCP for one. Is there some kind of reset possible? I didn't see anything about the reset button - which of course I tried a few times. tai ___ 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] Asterisk Assistants for Linux or Windoze???
RE: [Asterisk-Users] Are there any graphic designers on (Bthis list? (B (BMark Paterson wrote: (B (BIs there an Asterisk Assistant for linux or windows? (B (BIt wouldn't make sense to create Asterisk Assistants for (BWindoze (it'd probably be called Asterisk Wizards, btw) (Bbecause Asterisk doesn't run on Windoze. (B (BAs for Linux, we are currently looking into that. (B (BThe Assistants are built using the Cocoa framework which (Bis a further development of NeXTStep/OpenStep. I am (Bcurrently playing with a tool called the (BPython-Objectice-C Bridge or in short PyObjC. (B (BThis is primarily intended to make it possible to use the (BCocoa framework from Python and to integrate with (BInterface Builder and Xcode on MacOSX. However, PyObjC (Balso supports OpenStep on Linux. (B (BThis means in principle it is possible to use PyObjC to (Bwrite a cross-platform GUI tool that would run under (BMacOSX and OpenStep on Linux. Nicolas Gudino (the author (Bof the Flash Operator Panel for Asterisk) and I have (Bdiscussed this possibility and we will be looking further (Binto this. (B (BI don't want to make any promises yet, but with a bit of (Bluck and perhaps a bit of help from others, we may be able (Bto make the existing Asterisk Assistants for OSX cross (Bplatform so that they can run under OpenStep. (B (Brgds (Bbenjk (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disconnection From IAXTel
Muiz Motani wrote (B (B How do I go about disallowing transfers when I am (B running an IAX soft phone. (B (BYou'll have to ask the author(s) of the softphone. I can (Bonly tell you about my observations with peering Asterisk (Bservers which suggest that disabling transfer might solve (Byour problem. (B (Brgds (Bbenjk (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: where can I find spandsp?
On Sun, Aug 29, 2004 at 12:20:49AM -0600, Rich Adamson wrote: Seems the opencall.org site has basically been unavailable for days/weeks. Is there another location to obtain the current code? http://sremington.zapto.org/downloads/asterisk/spandsp/ Just one week ago Seth Remington did send this information to the list. Also, will spandsp install against the current * cvs? Yes, but make shure that libtiff version 3.5.7 or 3.6.0 is used. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy Power in Australia?
Roy Eddleston wrote: Jeremy Don't bother looking for a PSU with a lower current rating, The IAXy like any electrical device is designed to run at a particular voltage and consumes a certain amount of power (W) at that voltage. My comment was based on opinions of others that have posted to this list stating the PSU that shipped was 900mA or was it only 800... in any case most devices are rated with a maximum amount of amp needed, not the minimum, devices usually don't draw a constant amount of amp, it varies depending if it's actually doing work or not, and since most of the time this device will most likely be idle (ie not in a phone call) it's unlikely to draw the maximum current most of the time... On the other hand if the device design has changed and the requirement for current has also change then it's possible later models won't require the same amount of amps, this can be due to refinement of design to be more efficient and require less power or any other number of reasons... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than myself. ___ 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] Caller ID problem
Dear all
I am using ISDN modem by OTE in Athens, the modem has 2 FXS interface which
is connected to X100P cards. The problem is we are not getting the caller id.
And also console print blank or garbage character in incoming call during
execution of NoOP,${CALLERIDNUM}. See the sample
WARNING[28693]: File chan_zap.c, Line 4056 (ss_thread):
CallerID returned with error on channel 'Zap/2-1'
-- Starting simple switch on 'Zap/2-1'
-- Executing Wait(Zap/2-1,
2) in new stack
-- Executing Answer(Zap/2-1,
) in new stack
-- Executing NoOp(Zap/2-1,
üüü) in new stack
Any help would be appreciated. Thanks in advance.
Regards
WAHID
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[Asterisk-Users] Extens and number converting so that i can dial following one standaard.
For asterisk I am using more than one sip providers. The provider in Holland would like to have the international calls like 00 31 20 1234567 but the provider in the US likes it like 0011 31 20 1234567 Can I make a rule in asterisk so that I can dail 00 31 20 1234567 and asterisk dails 0011 31 20 1234567 to the US provider? Greetings Han ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Termination in Holland.
Mike, Sometime ago there was a message from rits they can help you out. I think it is one of the biggest Voip companies in Holland From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Sikkema Sent: woensdag 18 augustus 2004 6:17 To: [EMAIL PROTECTED] Subject: [OT] RE: [Asterisk-Users] SIP / IAX provider in the Netherlands. [EMAIL PROTECTED] wrote: Can you reccomend a SIP / IAX provider in the Netherlands ? I need a few Numbers, and of course cheap rates :) We van provide SIP termination, send an email to [EMAIL PROTECTED] about your needs. -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: zondag 29 augustus 2004 6:11 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Termination in Holland. Hi all, Can you guys recomend a good terminiation partner in Holland ? /Mike ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sunday 29 August 2004 02:06, [EMAIL PROTECTED] wrote: On Sat, 28 Aug 2004, Andrew Kohlsmith wrote: Please note that it seems impossible to disable jitter buffer between 20040806 CVS HEAD endpoints. The jitterbuffer numbers in iax2 show channels look live. The numbers look right (jitbuf 0ms) between 20040806 and RC1 (Nufone). I haven't upgraded since then. The numbers get reported still in the older version, but the buffer IS turned off. Ok so the disparity between iax2 show channels between two 20040806 (looks live) and 20040806 and RC1 (shows 0s) is expected? Just making sure, as between the two 'new' versions it is live, but between the new and old, it looks dead, whereas your reply said the numbers are still reported in the older version and that's not what I'm seeing. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote: If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. To do that, in /etc/asterisk/logger.conf edit the debug line to be: debug = notice,warning,error,debug,verbose Then run asterisk like so: /usr/sbin/asterisk -vv -g -dd -c Then go iax2 debug at the CLI prompt. Do a test call, then send me the resulting /var/log/asterisk/debug file. Is there any way to do this 'live'? I get it intermittently and capturing debug for days before the problem is manifest is probably not the best way to do it. I've tried leaving the debug line in and not invoking any kind of -d in the asterisk startup but the debug log still grows. I can't comment out the debug line in logger.conf because a logger reload or reload will NOT create the debug file, only a restart will. Ideally some way to create the debug file but write very litte to it until I connect with asterisk -rc or something would be best I imagine. Also, is are logs of problem conversations already in progress any use to you? You nailed down the dead audio after 65535ms problem but every now and again (very very rare) we will have a conversation where the incoming audio goes totally dead for about 2-4 seconds and then continues just fine. This occurs usually several minutes into the conversation, and I've never seen it occur twice in a conversation. Obviously this is next to impossible to catch. :-( I haven't heard a complaint about it since turning off jitter buffer to nufone. As always, thank you for your knowledge and input. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mobile phone integration via bluetooth
On Aug 29, 2004, at 7:51 AM, [EMAIL PROTECTED] wrote: Has anybody tried to integrate a mobile phone via blutooth in asterisk PBX? I believe the most things needed are just existing in open source. I found a kbthandfree (http://docs.kde.org/en/HEAD/kdeextragear-3/kdebluetooth/ components.handsfree.html) wich allows to control a mobile phone via an application and use your computer as headset. You may dial und receive calls. Using a way like that it would be possible to integrate a mobile hone to astrisk (as channel - control and voice data are transported via bluetooth). So it would be possible to receive GSM (mobe phone) calls in the private branch, and even least cost routing using the GSM network und POTS would be possible. It comes up every now and then, but no one has actually done anything with it. Feel free to try. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote: If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
Those wild times especially occur before any audio is sent. (e.g. while ringing or pre ringing). At 17:10 29/08/2004, you wrote: On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote: If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW ___ 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 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] Re: Distinctive ring detection problem
Sure, this one works. You need a dringX definitions of the distinctive rings. Put in each one the output you get in the log for the call pattern when the phone gets answered. [channels] switchtype=national signalling=fxs_ks usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=256 echocancelwhenbridged=yes echotraining=yes rxgain=-5.0 txgain=-5.5 group=1 callgroup=1 pickupgroup=1 immediate=no dring1=0,0,0 ---from the log output when phone answered. dring1context=advan-mainline dring2=326,0,0 ---from the log output dring2context=advan-fax dring3=93,0,0 ---from the log output dring3context=distring3 dring4=94,0,0 ---from the log output dring4context=distring4 Quoting Paul Budden [EMAIL PROTECTED] I am trying to get distinctive ring to work on my PSTN with no luck. I can get 2 different ring codes but it skips the context assigned... here is my complete zapata.conf: [channels] signalling=fxs_ks usecallerid=yes rxgain=1.0 txgain=1.0 language=en context=default usedistinctiveringdetection=yes dring1=134,0,0 dring2=137,0,0 dring1context=internal2 dring2context=default channel = 1 -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote: On Saturday 28 August 2004 23:01, Michael George wrote: It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not running and the Framebuffer has been turned off in /boot/grum/menu.lst. I have disabled all the servers except for sshd. I have the latest source from CVS HEAD as of about 30min ago. Should be fine. I ran * on a P90 for a while; it did everything I needed except iLBC. :-) Okay, that's a good assurance. Unfortunately, I have discovered that either the HDD or the ide controller in that system is bad because it cannot stay up overnight. When I stress it with a YaST update, it will die much more reliably. Until I can address that issue, I will have to work on my main system. I'll just have to take it down to init 3 and stop many of the other server processes that will still be running. There is no Zap card in this sytem. The only phone on it is a SIP phone. With it I dial in to digium's 1-700 number. The audio is better, but still choppy and unacceptible. Is your SIP phone doing any kind of silence suppression? It must be turned off because asterisk takes its timing from the RTP stream and if the phone's not transmitting frames continuously you'll get shitty audio. Good suggestion and I have double checked it. I am and was not doing that. I think I'd read about it in a Granstream-* page Note that latest CVS HEAD looks like they're making provisions for self-timing but without a stable clock source it's unlikely to help you. There are ztdummy modules which use the RTC or certain brands of USB controller to provide adequate timing but ideally you want some kind of Zaptel hardware in there providing a 1000Hz interrupt. Hmm, I thought that the timing source was only needed for trunking. I don't have on on the little box, but I do have a TDM400 (which seems to have faults, also, but Digium suggested moving the FXO to socket 4, we'll see if that helps) in the main box, so that should be all set for a timing source. Also -- make sure your uplink is acceptable. First test: make sure there is nothing plugged into your upstream except for your asterisk box and the phone. Some routers are known to play silly bugger with your packets which naturally wreaks havoc with asterisk. :-) The only things on the net when I run the next test will be my main server. Since I have to test on that with X turned off, I don't even need the SIP phone active. In case it might be relevant (there are SO many pieces to this puzzle that I want to mention all I can think of in case they ring a trouble-bell in someone's mind...) my router is a Netgear FVS318 acting as a NAT to my ISP. So even with X11 eliminated the sound is still bad to Digium. I tried another's 1700 number, and it sounded the same, so it's not something unique to digium and me. Perhaps something to do with your upstream or connection to IAXtel. That's why I'm recommending having nothing but asterisk and the phone on the connection, at least until we nail down what the poor audio's being caused by. That's possible. I've checked with my ISP and he said that the connection is surely half-duplex, but you say that you have 1/2 also and it works fine for you, so that's not it. I'm also inquiring about other filters they might have in place. I've heard them mention before that they had some cool router software that could detect traffic patterns usually associated with software and music piracy and then throttle that traffic into a small part of The Pipe. I haven't yet heard back, and I'm hoping that isn't the case. However, if it *is*, a VPN between offices might help. IAXtel would be shot, though. Hoever, if that *is* the case, I can probably convince them to tell their software to leave me alone on a couple specified ports. Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work with my ISP only giving me 1/2 duplex service? It has nothing to do with IAX or GSM. Stop blaming them. My upstream is half duplex as well (pretty much anyone on DSL or cable is on a half duplex connection whether they realize it or not). There are many, many people using asterisk every day for long distance and in environments where audio quality is crucial. Let's stop blaming asterisk and take a good hard look at what's happenning, shall we? My apologies. I'm not trying to blame anyone, I love * and except for a couple glitches that we're working on (with all your gracious help), I'm very impressed. My one glitch may be with the hardware, so that's a separate issue, but the other is trying to figure out this issue with IAX/GSM. When I ask about sensitivity, I don't mean to be accusatory. IAX is open and freely available and GSM is freely usable, and I'm glad. Sometimes OSS has its limitations and I am willing to work with them. So I do not intend any condescention(sp?),
Re: [Asterisk-Users] iaxtel and jitterbuffer
At 17:10 29/08/2004, you wrote: I notice that the timing measurements are still showing wild values at times - here is a partial grab of an iax2 show channels: Lag Jitter JitBuf Format 00020ms 6291456ms ms ALAW 00012ms 6291440ms ms ALAW 00017ms 0004ms ms ALAW 00012ms 286523393ms ms ALAW 00012ms 0025ms ms ALAW -978714621ms 6293280ms ms ALAW Those wild times especially occur before any audio is sent. (e.g. while ringing or pre ringing). That maybe true, but the examples above appeared to be established calls! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sun, Aug 29, 2004 at 07:59:20AM +0200, [EMAIL PROTECTED] wrote: On Sat, 28 Aug 2004, Michael George wrote: So even with X11 eliminated the sound is still bad to Digium. I tried another's 1700 number, and it sounded the same, so it's not something unique to digium and me. Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work with my ISP only giving me 1/2 duplex service? If you think that the jitter buffer isn't working right and should fix this, then please capture debug from the buffer and send over to me. I'm not sure what the problem is. What I am hearing does sound like the descriptions I've read w.r.t. the jitter buffer, but making jitter buffer changes haven't really changed the effect. That gives 2 possibilities: 1. That the jitter buffer isn't working and it *should* fix the problem. 2. That the problem is completely independent of the JB so there is nothing the JB can do to fix it. To do that, in /etc/asterisk/logger.conf edit the debug line to be: debug = notice,warning,error,debug,verbose Then run asterisk like so: /usr/sbin/asterisk -vv -g -dd -c Then go iax2 debug at the CLI prompt. Do a test call, then send me the resulting /var/log/asterisk/debug file. I will do that. Hopefully that will help us isolate the problem and perhaps eliminate the jitterbuffer from the equasion. :) I will try to run this test today and report back my findings. Also, on Thursday I will be going into the main office. I will take my little * box and try the IAXtel test there. That should help determine if it's my local office net connection that is the problem. Thank you! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
[EMAIL PROTECTED] wrote: Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss? How much work would be required to change this? I guess it couldn't really be an option because of the totally different structure... Would it be possible for one person to make those changes or would it require the authors of all modules to recode? I haven't even completed by soft fax machine, and you are trying to be it completely useless. :-) Think about that. What you are suggesting is not really a satisfactory solution to anything, but certainly breaks things. :-\ Regards, Steve ___ 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] Empty Queues
Hi, Is there a way to detect if the caller will be entering an agentless queue? Id like to be able to redirect any caller who tried to join a queue with no logged in agents, to be redirected to the groups voicemail. Is this possible? I know I could create a menu and an announcement for voicemail (should the user wish to drop from the queue); but they wouldnt know no one was taking calls :/ Any help much appreciated. Regards, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Telephony with Asterisk book
Joseph Shi wrote: Does anyone know if there are any reseller for the book VoIP Telephony with Asterisk in Hong Kong/Asia region? I'm interested in purchasing the book but the shipping charge to Hong Kong is expensive. Thanks. Joseph Just wait for the simplified Chinese version to appear in Shenzhen's Book City. :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how does one get the very best quality output?
Clayton Smith wrote: Hi, i'm trying to send some songs over via asterisk, so i'm trying to get the very best quality possible i've been using gsm, using sox with a rate of 8000, single channel, resampled q1, and got some good results, but i'm wondering if there is at all a better way I'm using voicepulse, which supports * GSM * G.711ulaw * G.711alaw * ADPCM * ILBC * SPEEX any of those better to send music through G.711 is a lossless codec, so either G.711 would be better than a lossy codec like GSM for sending music. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
On 30 Aug 2004 at 0:26, Steve Underwood wrote: [EMAIL PROTECTED] wrote: Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss? How much work would be required to change this? I guess it couldn't really be an option because of the totally different structure... Would it be possible for one person to make those changes or would it require the authors of all modules to recode? I haven't even completed by soft fax machine, and you are trying to be it completely useless. :-) Think about that. What you are suggesting is not really a satisfactory solution to anything, but certainly breaks things. :-\ Is this English?! my soft fax? make it completely useless? Okay, I think I understand you now... This surely wouldn't concern your code unless your code does it's transmission via IAX, SIP, OpenH.323 etc? And unless I'm gravely mistaken fax won't work over IP anyway... Matt Riddell ___ 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] Sip device not login or register calls to that device go to busy voicemail not un-available
I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? Ariel Batista Kasi International - Computer NetworkingPh: 305-574-6721Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip device not login or register calls to that device go to busy voicemail not un-available
I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? That's the way its always been. Lots of folks believe its not the 'correct' way, but I've since forgotten what the logic was for leaving it the way it is now. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Sip device not login or register calls to that device go to busy voicemail not un-available
give a piece of you extensions.conf where it is configured On Sun, 29 Aug 2004 13:46:37 -0600, Rich Adamson [EMAIL PROTECTED] wrote: I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? That's the way its always been. Lots of folks believe its not the 'correct' way, but I've since forgotten what the logic was for leaving it the way it is now. ___ 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 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] Servers
Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but Im looking for a little more input on Askterisk Servers that host other network services as well. Thank you, Steve Maroney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Servers
I'm using 2 Dell Poweredge 2650 servers with a Wildcard TE410P in each and a custom linux installation. Works great and even picks up the dual xeons as quad processors. Duane Cox - Original Message - From: Steve Maroney [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 29, 2004 2:20 PM Subject: [Asterisk-Users] Servers Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but Im looking for a little more input on Askterisk Servers that host other network services as well. Thank you, Steve Maroney ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Servers
On Sunday 29 August 2004 15:20, Steve Maroney wrote: Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Typically speaking you do *not* want to do that. Asterisk is a latency-sensitive application and serving files, web pages, etc. is just asking for trouble. Certainly it can be done but you're asking for weird and inconsistent little problems. It has nothing to do with the processor load; it has everything to do with I/O and interrupt latency. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote: Also, is are logs of problem conversations already in progress any use to you? You nailed down the dead audio after 65535ms problem but every now and again (very very rare) we will have a conversation where the incoming audio goes totally dead for about 2-4 seconds and then continues just fine. This occurs usually several minutes into the conversation, and I've never seen it occur twice in a conversation. Logs of parts of a call are fine. The jitter buffer makes all its decisions about dejittering based on the timestamps of incoming frames. There a fundamental expectation that the sending side is correctly stamping each frame - 20msec, 40msec etc etc. The problem is that the sending side doesn't always do that. Sometimes for one reason or another the stamps jump. The receiver has no way of telling that the sender mangled the timestamps, and assumes that the packets with the new stamps have been delayed, or arrived early, or whatever. Either way, the jitter buffer does its thing and unknowingly makes things worse. Unfortunately, this is why you can still be better off without it - but the problem really needs to be fixed by fixing the timestamp generation on the sender. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sun, 29 Aug 2004, joachim wrote: Those wild times especially occur before any audio is sent. (e.g. while ringing or pre ringing). Yeah - because the sender does weird things to the timestamps it generates. This is the problem that needs to be resolved; the jitter buffer just shows up the issue. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sunday 29 August 2004 15:52, [EMAIL PROTECTED] wrote: The jitter buffer makes all its decisions about dejittering based on the timestamps of incoming frames. There a fundamental expectation that the sending side is correctly stamping each frame - 20msec, 40msec etc etc. Right, this makes sense. :-) The problem is that the sending side doesn't always do that. Sometimes for one reason or another the stamps jump. The receiver has no way of telling that the sender mangled the timestamps, and assumes that the packets with the new stamps have been delayed, or arrived early, or whatever. Either way, the jitter buffer does its thing and unknowingly makes things worse. Unfortunately, this is why you can still be better off without it - but the problem really needs to be fixed by fixing the timestamp generation on the sender. Hmm... I think next CVS update I'm gonna add a bit of code in chan_iax2 that tries to verify that timestamps aren't getting sent incorrectly. Fun fun fun. :-) -A. ___ 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] Jitter buffer
Hi, I thought I'd repost this to the -users list for some background on the jitter buffer and its workings and remaining issue.s I'll also pu a little executive summary here at the top: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) The jitter buffer on this machine is not used. Any jitter/jitterbuffer measurement shown is left over from the start of the call. Steve -- Forwarded message -- Date: Thu, 12 Aug 2004 00:02:26 +0200 (SAST) From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Dev] CVS HEAD (20040807) jitter buffer questions On Wed, 11 Aug 2004, Steve Kann wrote: The LAG measurement is pretty meaningless in the present implementation, if the clocks skew between both sides.Unless both ends of the connection are using ntp (for a while, and have stabilized), you can't trust it. [even then, I don't remember if ntp is accurate enough for this]. Andrew Kohlsmith wrote: I've been keeping an eye on the jitter buffer ever since upgrading and using Steve's patch to fix the 65s dead air problem but I've been noticing something... ... Asterisk frequently gets lag and jitter mixed up. This directly affects the jitter buffer, as the jitter buffer only grows when jitter grows. Hi Andrew, Please send logs to me. Or put them somewhere where I can get them. You need to understand that lag and jitter are measured completely independently, and need to be interpreted with care... First lag: Lag is measured by sending a LAGRQ frame from the one (final) end to the other (final) end. This frame has our timestamp time at which it was sent. When it arrives at the other (final) end it is immediately sent back. When it arrives back at the starting point, the echoed timestamp is compared with now and a lag is derived. Notice that the compared timestamp is our own - so I don't agree with Steve's claim that the two ends need synchronised clocks. Each end of the IAX call send LAGRQs every 10 seconds, so the measurements are just a snapshot and not some sort of super accurate smoothed figure or anything. Its important to understand about the impact of native bridging (see chan_iax2.c, forward_packet). If an IAX2 call goes over multiple hops. (eg callerA - server1 - callerB), Asterisk servers in the middle of the path just blindly forward frames, and don't themselves process the LAGRQ packets. (There is a little tweak done to the timestamp so they have the right 0-point for each leg, but that should have a nett zero effect) So you will find lag as 000msec on machines in the middle of the call (eg server1 here), and you also need to understand that the lag reported on callerA and callerB machines is actually the lag for the whole path. So this is most likely the explanation for the 0ms lag. Perhaps we should change the iax2 show channels to highlight this situation in a special way. It is possible I suspect for a lag measurement to slip in before the call is completely established and the native bridging kicks in. So I suppose you might see an initial measurement left over. The -1ms is probably some sort of rounding error which should be looked at. The same principle applies to the jitter buffer. Asterisk servers in the middle of a call do not do de-jittering, they leave it all to the end machines. Or, at least, that is so once native-bridging kicks in. Again, the jitter buffer on an intermediate machine will be used whilst the call is establishing but will stop being used once the call is established. So I wouldn't expect meaningful jitter figures reported for calls being native bridged on this box. Perhaps again we should hide them and mark N/A or something? Jitter is measured quite separately to lag. It is done by comparing the delay that arriving frames have experienced. You see (more of less) the biggest variation in delay seen over the last 2 seconds. Its quite possible for the jitter to show as more than the lag. The lag is a snapshot as seen by one frame going there and back. The jitter is measured by looking at 100 frames. And is, more or less again, the delay seen by the most delayed frame minus the delay seen by the least delayed frame. The jitter buffer only has a relative view of things and doesn't know the absolute delay - because it only has the other side's timestamps to look at. So you can see that you could get a 20msec lag and even a 1msec jitter. I'd just comment that the jitter buffer only needs to account for jitter. It matters nowt to the jitter buffer if the packets have been 2 hours in transit as long as they arrive at a steady pace. Sometimes timestamps on an iax call do jump rather than incrementing steadily. This can happen for example where voice starts coming from a new source. The jitter buffer sees
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote: Hmm... I think next CVS update I'm gonna add a bit of code in chan_iax2 that tries to verify that timestamps aren't getting sent incorrectly. Fun fun fun. :-) Its not that the generation is broken. Its that various optimisations and things have been added over time. The result is that sometimes the source of the timestamps changes - and suddenly. Like - we're playing locally generated Playback() audio down the line, then the dialplan rings another IAX2/ address. Then the other end answers. First the timestamps come from the Playback, then the ring generator, then from the remote IAX2/ system... So the discontinuities get in. There is also effort in the sending IAX2code to lock the timestamps to exact intervals (20msec), but sometimes it gives up and lets it jump to get back into step... Steve ___ 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] Python and AGI
Anyone using Python to write their AGI applications? I find very little info on it. The wiki has a link to http://sourceforge.net/projects/pyst but it seems like a dead project. I posted some questions to their mailing list a week ago and have not seen a reply or other posting. Is there some other python module that people are using? -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgp42OXsSKHNP.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel and jitterbuffer
On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote: On Saturday 28 August 2004 23:01, Michael George wrote: It has nothing to do with IAX or GSM. Stop blaming them. My upstream is half duplex as well (pretty much anyone on DSL or cable is on a half duplex connection whether they realize it or not). There are many, many people using asterisk every day for long distance and in environments where audio quality is crucial. Let's stop blaming asterisk and take a good hard look at what's happenning, shall we? Someone suggested that perhaps the machine is too slow. If someone who uses IAX2 between offices wouldn't mind, could you please indicate how heavy a system you are using for Zap -- IAX/GSM -- VOIP. Perhaps I am underestimating the HP required for the voice coding... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()
Robert, Thanks for the reply. I tried that initially and it did not work. To verify I went back and tried again. It answers and still no sound is heard. From the CLI I can see it answer and ask for conf-getconfno three times before executing the hangup... But no sound. Yet if I point the DID to a SIP extension it rings, upon answer there is 2-way speech path. Any other ideas? -- Accepting call from '8541' to '2688' on channel 0/2, span 1 -- Executing Wait(Zap/2-1, 3) in new stack -- Executing Answer(Zap/2-1, ) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing MeetMe(Zap/2-1, |Mps) in new stack -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (nec_pri, 2688, 5) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Jackson Sent: Friday, August 27, 2004 11:31 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe() -Original Message- From: Larry Shields [mailto:[EMAIL PROTECTED] Sent: Friday, August 27, 2004 12:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() If I assign the DID to ring extension SIP/2000 and then after time-out send it to MeetMe() or Playback() it works and the caller hears the .gsm file. Any assistance in solving this problem is appreciated. [nec_pri] ; Digital PRI from the NEAX2400 exten = 2688,1,Wait,3 exten = 2688,2,MeetMe,|Mps exten = 2688,3,Hangup I had a similar problem with my system, and I was able to fix the problem by executing Answer before I entered any other applications. Using your previous example: exten = 2688,1,Answer exten = 2688,2,Wait,3 exten = 2688,3,MeetMe,|Mps exten = 2688,4,Hangup Hope this helps, Robert Jackson ___ 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 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] not getting ringing/busy/answer feedback on my PRI
I posteda problem earlier thinking it was due to a lack of sound card.Several members stated that you do not need a sound card to play audio to a PRI channel. I did some further testing and discovered that there is a problem with call progress tones or signalingon my PRI. Ithink that the reasonI am not hearing audio from the MeetMe() or Playback() apps. is because the the calling side of the PRI (NEC IPX), is not seeing the Answer signal. I believe itis waitingfor a ring and/or answercondition evenafter Asterisk has executed an Answer() and Playback(). The only other problem that I am having with my setup is that the CONSOLE/DSP is not functional... I am not sure if the two problems are related. Any help is appreciated. Please see my two examples below: Unless my incoming DID(2000), is pointed to a SIP station that is registered and functional, I do not receive call progress toneson inbound calls. If I point the DID to an application like: [inbound_pri]; PRI from the NEAX2400 exten = 2000,1,Wait,3exten = 2000,2,Answerexten = 2000,3,MeetMe,|Mpsexten = 2000,4,Hangup I will not hear any initial ringback,and once answeredthere will be no audioon the channel. If I point the DID to a registered SIP station like: [inbound_pri]; PRI from the NEAX2400 exten = 2000,1,Wait,3exten = 2000,2,Dial,SIP/2001,15,Trexten = 2000,Hangup Itwill provide ringback tone to the calling channel on the PRI, andwhen theringing SIP phone answers there willbe 2-way speech path. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] not getting ringing/busy/answer feedback on my PRI
This is my PRI Debug
info for those interested in this problem:
PMDBRIDGE*CLI Protocol Discriminator:
Q.931 (8) len=39 Call Ref: len= 2 (reference 115/0x73)
(Originator) Message type: SETUP (5) [04 03 90 90 a2]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: 3.1kHz audio
(16)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83
83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan:
0
ChanSel:
Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 3 ] [1e 02 81 83] Progress Indicator (len=
4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
(1)
Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ]
[6c 0b a1 39 37 32 33 31 35 38 35 34 31] Calling Number (len=13) [ Ext:
1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
(E.164/E.163)
(1)
Presentation: Presentation permitted, user number not screened (0) '8541'
] [70 05 a1 32 36 38 38] Called Number (len= 7) [ Ext: 1
TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163)
(1) '2688' ]-- Making new call for cr 115-- Processing Q.931 Call
Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24
(cs0, Channel Identification)-- Processing IE 30 (cs0, Progress
Indicator)-- Processing IE 108 (cs0, Calling Party Number)-- Processing
IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931
(8) len=14 Call Ref: len= 2 (reference 32883/0x8073)
(Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 03 a9 83
83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan:
0
ChanSel:
Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 3 ] [1e 02 81 82] Progress Indicator (len=
4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
(1)
Ext: 1 Progress Description: Called equipment is non-ISDN. (2)
] -- Accepting call from '8541' to '2688' on channel 0/3,
span 1 -- Executing Wait("Zap/3-1", "2") in new
stack Protocol Discriminator: Q.931 (8) len=13 Call Ref:
len= 2 (reference 115/0x73) (Originator) Message type: STATUS
(125) [08 03 80 e1 0d] Cause (len= 5) [ Ext: 1 Coding:
CCITT (ITU) standard (0) 0: 0 Location: User
(0)
Ext: 1 Cause: Message type nonexist. (97), class = Protocol Error (6)
]
Cause data 1: 0d (13) [14 01 01]I Call State (len= 3) [ Ext:
0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1)--
Processing IE 8 (cs0, Cause)-- Processing IE 20 (cs0, Call
State) -- Executing Answer("Zap/3-1", "") in new
stack Protocol Discriminator: Q.931 (8) len=14 Call Ref:
len= 2 (reference 32883/0x8073) (Terminator) Message type: CONNECT
(7) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID:
Implicit, PRI Spare: 0, Exclusive Dchan:
0
ChanSel:
Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 3 ] [1e 02 81 82] Progress Indicator (len=
4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
(1)
Ext: 1 Progress Description: Called equipment is non-ISDN. (2)
] -- Executing MeetMe("Zap/3-1", "|Mps") in new
stack -- Playing 'conf-getconfno' (language
'en')PMDBRIDGE*CLI pridebug intense
no showPMDBRIDGE*CLI pri no debug
span 1Disabled debugging on span 1 -- Playing
'conf-getconfno' (language 'en')
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Larry
ShieldsSent: Sunday, August 29, 2004 3:42 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] not getting
ringing/busy/answer feedback on my PRI
I posteda
problem earlier thinking it was due to a lack of sound card.Several
members stated that you do not need a sound card to play audio to a PRI
channel. I did some further testing and discovered that there is a problem
with call progress tones or signalingon my PRI. Ithink that
the reasonI am not hearing audio from the MeetMe() or Playback() apps. is
because the the calling side of the PRI (NEC IPX), is not seeing the Answer
signal. I believe itis waitingfor a ring and/or
answercondition evenafter Asterisk has executed an Answer() and
Playback().
The only other
problem that I am having with my setup is that the CONSOLE/DSP is not
functional... I am not sure if the two problems are related. Any help is
appreciated. Please see my two examples
below:
Unless my incoming
DID(2000), is pointed to a SIP station that is registered and
functional, I do not receive call progress toneson inbound
calls.
If I point the DID
to an application like:
[inbound_pri]; PRI from the
NEAX2400
exten =
2000,1,Wait,3exten = 2000,2,Answerexten =
2000,3,MeetMe,|Mpsexten = 2000,4,Hangup
I will not hear any
initial ringback,and once answeredthere will be no audioon the
channel.
If I point the DID
to a registered SIP station like:
[inbound_pri];
PRI from the NEAX2400
exten =
Re: [Asterisk-Users] Jitter buffer
On Sunday 29 August 2004 16:07, [EMAIL PROTECTED] wrote: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) The jitter buffer on this machine is not used. Any jitter/jitterbuffer measurement shown is left over from the start of the call. When I next update CVS HEAD would a patch which zeroes these values under these conditions be accepted? Also if my understanding is correct, the only time IAX2 jitter buffer is ever used is where IAX2 -- something else (SIP, Zap, OSS, MGCP, etc.) is performed. Is this the case? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXy Power in Australia?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Duane Sent: 29 August 2004 11:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXy Power in Australia? Roy Eddleston wrote: Jeremy Don't bother looking for a PSU with a lower current rating, The IAXy like any electrical device is designed to run at a particular voltage and consumes a certain amount of power (W) at that voltage. My comment was based on opinions of others that have posted to this list stating the PSU that shipped was 900mA or was it only 800... in any case most devices are rated with a maximum amount of amp needed, not the minimum, devices usually don't draw a constant amount of amp, it varies depending if it's actually doing work or not, and since most of the time this device will most likely be idle (ie not in a phone call) it's unlikely to draw the maximum current most of the time... On the other hand if the device design has changed and the requirement for current has also change then it's possible later models won't require the same amount of amps, this can be due to refinement of design to be more efficient and require less power or any other number of reasons... Duane My post was directed at Jeremy to save him wasting his time replacing his 1500mA PSU with one of a lower current rating, but since you've replied quoting one paragraph out of context it warrants a reply so there is no confusion. I agree that devices are generally rated with the maximum current they require, plus a margin for safety, I never said they didn't. I also agree that the current drawn by the IAXy will vary (potentially up to it's maximum rating) depending on what it is doing at the time and whether an attached phone is off hook, ringing etc, again I never said it wouldn't. If you read the whole text of my post it is based on electrical/electronic principals for devices in general not just the IAXy although it makes no difference. You're also correct that if the design of the IAXy has changed it may have a lower maximum current REQUIREMENT or indeed a higher one, I said that in the second paragraph of my post. However your comment Have you tried feeding it less amps at all is impossible to achieve unless the PSU concerned is underrated and incapable of supplying sufficient current to the device at the rated voltage, in which case the PSU would then overheat and its output voltage drop therefore resulting in potentially early failure of the PSU and instability or failure in the device connected to it. This misled Jeremy into thinking purchasing a PSU with a lower current rating would achieve anything, his PSU is a 1500mA unit which is rated correctly according to the manufactures specification and will provide whatever current is needed up to and including 1500mA depending on whether that 1500mA rating is a constant current rating or based on a specific duty cycle, but that's another ballgame that I am not going to discuss here. You can not feed a device less current, a device will DRAW whatever current it needs depending on the supplied voltage and the resistance of its internal circuits or other circuits attached to it, the only thing that matters is that the PSU or device supplying that current is capable of supplying the maximum current the attached device will draw. So as I stated in my previous post, if the IAXy or any other device requires a current of say 800mA for example, it does not matter whether the PSU or device supplying that current is capable of supplying 800mA or 100,000A, the device will still only draw the 800mA it needs and no more, you can not feed it less. OHM's LAW is still a wonderful thing ;) Cheers! Roy... ___ 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] SMS and asterisk
Hi all! I am intrested in the following scheme My mobile phone - SMS to SOMEONE - Redirect to FWD number - FDW redirect to my * There are companies like calluk.com that provide DIDs for free, but they do not support SMS. In http://www.voip-info.org/wiki-Asterisk+cmd+Sms they say Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK Can you please advise how to get SMS into * ? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Servers
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote: On Sunday 29 August 2004 15:20, Steve Maroney wrote: Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Typically speaking you do *not* want to do that. Asterisk is a latency-sensitive application and serving files, web pages, etc. is just asking for trouble. Certainly it can be done but you're asking for weird and inconsistent little problems. It has nothing to do with the processor load; it has everything to do with I/O and interrupt latency. -A. Niceley said. Thank you for the advice. Steve Maroney ___ 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] System freezes when using Festival with usecache
I am using Festival to synthesize some menu Interaction with a caller
and am having a problem.
What I am working on is a remote callback where I can remotely call in
to an extension, and enter a callback number (or use the CALLERID info)
and a second outbound dialing number to connect to.
Things work O.K. until I set usecache=yes in festival.conf. After
doing this, things run well for the first two or three Festival
Commands, then the call to locks up. Here is a copy of my festival.conf
file that causes this behavior.
The dialplan logic works OK otherwise (ultimately I will probably
convert this to AGI).Also I have included a copy of what I am doing
in my extensions.conf to possibly give some insight. Any help is
appreciated.
festival.conf
[general]
host=localhost
port=1314
usecache=yes
cachedir=/var/lib/asterisk/festivalcache/
festivalcommand=(tts_textasterisk %s 'file)(quit)\n
from extensions.conf
exten = 4998,1,Answer
exten = 4998,2,SetVar(YourNumber=1${CALLERIDNUM})
exten = 4998,3,Goto(6)
exten = 4998,4,Festival(Caller please enter the callback number
followed by the pound key)
exten = 4998,5,Read(YourNumber)
exten = 4998,6,Wait(1)
exten = 4998,7,Festival(Caller please verify your callback number)
exten = 4998,8,SayDigits(${YourNumber})
exten = 4998,9,Wait(1)
exten = 4998,10,Festival(Caller press 1 to use this number press 2 to
repeat the number press 3 to enter a new number)
exten = 4998,11,Read(Result,,1)
exten = 4998,12,GotoIf($[${Result} = 1]?15:13)
exten = 4998,13,GotoIf($[${Result} = 2]?7:14)
exten = 4998,14,GotoIf($[${Result} = 3]?4:10)
exten = 4998,15,Festival(Caller please enter the outbound number
followed by the pound key)
exten = 4998,16,Read(TheirNumber)
exten = 4998,17,Wait(1)
exten = 4998,18,Festival(Caller you have entered)
exten = 4998,19,SayDigits(${TheirNumber})
exten = 4998,20,Wait(1)
exten = 4998,21,Festival(Caller if this is correct press 1 otherwise
press 2)
exten = 4998,22,Read(Result,,1)
exten = 4998,23,GotoIf($[${Result} = 1]?25:24)
exten = 4998,24,GotoIf($[${Result} = 2]?15:21)
exten = 4998,25,Wait(1)
;place a call in asterisk/outgoing before hangingup
exten = 4998,26,System(/bin/nohup /usr/local/bin/connect ${YourNumber}
${TheirNumber} 30)
exten = 4998,27,Wait(1)
exten = 4998,28,Festival(Caller this session is completed please hanup now)
exten = 4998,29,Wait(1)
exten = 4998,30,Hangup
___
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[Asterisk-Users] Revert to dial tone?
I am wondering if it is possible for an extension that is served by a zaptel device to revert to dial tone once a call disconnects. For instance, if I make a call to another extension, talk with them, and THEY hang up, can I then be presented with a new dial tone rather than a congestion tone? Further, can an extension be set up so that, once the call goes back to dial tone, if the user does NOT dial any digits within a timeout period, + the PBX will return 30 seconds of congestion tone, and then + the PBX will return 60 seconds of howler tone, and then + the extension is 'locked out.' ? Thanks. ___ 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] Which Zaptel release goes with Asterisk-1.0-RC2 ???
Hi (B (BI am trying to build an Asterisk-1.0-RC2 server with (BZaptel support and two X100P cards. (B (BI was wondering which Zaptel release I should check out to (Bgo together with 1.0-RC2. I tried the CVS from 18 August (Bwhich worked fine on another machine with the exact same (BOS and kernel and this resulted in a rather nasty internal (Bcompiler error. (B (Btor2.c:636: internal compiler error: segmentation fault (Bplease submit a full bug report, (Bwith preprocessed source if appropriate (B (BNow, this doesn't look like it's due to a mismatch between (BAsterisk and Zaptel and I don't really want to mess around (Bwith this. All I want is a working Zaptel driver for the (BRC2 installation. (B (BI'd appreciate to hear any suggestions. (B (Bregards (Bbenjk (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS Asterisk
Hi all! I am intrested in the following scheme My mobile phone - SMS to SOMETHING - Redirect to FWD number - FDW redirect to my * - My * doing smtg There are companies like calluk.com that provide DIDs for free, but they do not support SMS. In http://www.voip-info.org/wiki-Asterisk+cmd+Sms they say Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK Can you please advise how to get SMS into * ? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting
if you have anyone questions about your service you can contact us at the support 978-418-7300 James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Ben Wern Sent: Sat 8/28/2004 4:34 PM To: Asterisk Users Subject: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting Can anyone who is using Asterisk with Broadvoice tell of their experiences with 3-way calling and call waiting? I can't get Broadvoice to respond to my question, but I understand that there is a per minute fee (3.9 c/minute?) if you go over your use allowances. My question is, how are 3 way and call waiting calls handled? Because Asterisk would just handle them as two different channels/calls -- does Broadvoice allow BYOD customers to have two active lines and then start charging for a third? If so, does anyone have any configuration examples of limiting the number of sessions to a single provider? Ben Wern ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice problem
you are correct. if you don't use sip.broadvoice.com it mess up you sip uri so the server will reject it. Also you should enable srvlookup it will help things run better. James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Ed Brady Sent: Sat 8/28/2004 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Broadvoice problem Marty Mastera wrote: I had the same problem. To fix it, I had to do two things First: I had to update to CVS head, this was as per broadvoice support. Second: After updating, I had to change my sip.conf. Originally my sip.conf used hard coded ip addresses for broadvoice's IP servers, so I had to change the following lines as such: register = [mynumber]:[EMAIL PROTECTED] to read register = [mynumber]:[EMAIL PROTECTED] Ed, Weird things...I took your advice but executed it in stages...just like you, I was registering with 147.135.8.129, hardcoded ip. My CVS-HEAD is 7/14/04. The only thing I changed so far is to replace the 147.135.8.129 with sip.broadvoice.com. I didn't update from CVS, I also don't have SRV lookups enabled (yet anyway). It now registers and I can receive inbound calls. Does it make sense that BV may have implemented a change that would allow registrations from a FQDN but not from a hardcoded ip? Just a thought Marty ___ 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 Marty, Yeah, I agree it is pretty weird that Broadvoice would have made this change. When I called support they said that they had made some changes to coverup up some kind of security loop hole, however I am not clear how this would relate to this FQDN change. If nothing else, it caused me to (finally) update my system. BTW, does the latest CVS code have better support for SRV lookups? Ed ___ 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] Static Problem (t100p - Channel Bank)
Hello, We keep having a really bad static problem on phone calls completed using a Adtran TA750 and T100P card. The phones are Polycom IP 500 phones and, it occurs across all phones. Not just one. Everything appears to be on it's own interrupt. I noticed the last time we did this, we rewired everything to make sure it wasn't an electrical issue. Well about a month and a half later it has resurfaced. Has anyone seen this before? Thanks - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxtel and jitterbuffer
Is timestamp information calculated purely from the relative timestamps of
each frame of the current incoming stream or is there some degree of RTC
synchronization expected between the two endpoints?
Similarly, are jitter calculations made seperately for each discrete channel
(ie. the IAX level) or are they based on an aggregate of all channels
between each pair of two endpoints (ie. the TCP/IP level)?
k.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: August 29, 2004 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iaxtel and jitterbuffer
{clip}
The jitter buffer makes all its decisions about dejittering based on the
timestamps of incoming frames. There a fundamental expectation that the
sending side is correctly stamping each frame - 20msec, 40msec etc etc.
The problem is that the sending side doesn't always do that. Sometimes
for one reason or another the stamps jump. The receiver has no way of
telling that the sender mangled the timestamps, and assumes that the
packets with the new stamps have been delayed, or arrived early, or
whatever. Either way, the jitter buffer does its thing and unknowingly
makes things worse.
Unfortunately, this is why you can still be better off without it - but
the problem really needs to be fixed by fixing the timestamp generation on
the sender.
Steve
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[Asterisk-Users] Asterisk H.323 channel...
Hi all,
I am trying to use a "Siemens optiPoint 300"
IPPhone (H.323 only) with Asterisk (1.0-RC2).
So far I have been using the H.323 channel
included in the tarball (Nufone ?).
I encountered a strange behaviour when I try
to make a call from the IPPhone to my Asterisk box :
= here is the H.323
configuration for the incoming calls (192.168.1.50 is the IP of the Siemens
phone) :
[Damien]type=userhost=192.168.1.50context=incoming
= the incoming context
has a single extension :
[incoming]
exten = 666,1,Playback(demo-echotest)exten
=
666,2,Echo
exten = 666,3,Playback(demo-echodone)
= the IPPhone is configured to use
the Asterisk box as a H.323 gateway (system type = "Gateway" / IP address of the
gateway = IP address of the Asterisk box)
= when I dial "666" on the IPPhone
Asterisk seems to answer the call then 2 seconds later it hangs up
:
* DEBUG MESSAGES DURING THE CALL
*
== New H.323 Connection
created. -- Received SETUP
message -- Setting up Call
-- Call token:
[ip$192.168.1.50:1257/5625] --
Calling party name: [] --
Calling party number: [987654321]
-- Called party name:
[666] -- Called party
number: [666]Urgent handlerAug 30 11:48:32 DEBUG[56142768]:
pbx.c:1255 pbx_extension_helper: Launching 'Playback'Aug 30 11:48:32
DEBUG[56142768]: channel.c:1666 ast_set_write_format: Set channel
H323/ip$192.168.1.50:1257/5625 to write format
GSM -- Received RELEASE COMPLETE
message... -- Sending RELEASE
COMPLETE
1:32.765
H245:8a174a0
h323.cxx(3195) H245 Read error: Interrupted system
call
1:32.781
H323 Cleaner
h323.cxx(1542) H323 Connection ip$192.168.1.50:1257/5625
terminated.-- 987654321, 987654321 [192.168.1.50] has cleared the
callAug 30 11:48:35 DEBUG[56142768]: channel.c:1666 ast_set_write_format:
Set channel H323/ip$192.168.1.50:1257/5625 to write format ALAWAug 30
11:48:35 DEBUG[56142768]: pbx.c:1827 ast_pbx_run: Spawn extension
(incoming,666,1) exited non-zero on 'H323/ip$192.168.1.50:1257/5625'Aug 30
11:48:35 DEBUG[56142768]: channel.c:733 ast_hangup: Hanging up channel
'H323/ip$192.168.1.50:1257/5625'Aug 30 11:48:35 DEBUG[56142768]:
chan_h323.c:531 oh323_hangup:
oh323_hangup(H323/ip$192.168.1.50:1257/5625)
== H.323 Connection deleted.
= if I had a "Wait 1" in front the
extension it works :
[incoming]exten = 666,1,Wait,1exten
= 666,2,Playback(demo-echotest)exten = 666,3,Echoexten =
666,4,Playback(demo-echodone)
* DEBUG MESSAGES DURING THE CALL
*
== New H.323 Connection
created. -- Received SETUP
messageUrgent handler -- Setting up
Call -- Call token:
[ip$192.168.1.50:1260/5626] --
Calling party name: [] --
Calling party number: [987654321]
-- Called party name:
[666] -- Called party
number: [666]Urgent handlerAug 30 11:53:34 DEBUG[114731952]:
pbx.c:1255 pbx_extension_helper: Launching 'Wait'Aug 30 11:53:35
DEBUG[114731952]: pbx.c:1255 pbx_extension_helper: Launching
'Playback' =*= In
CreateRealTimeLogicalChannel for call
5626
-- externalIpAddress:
192.168.1.201
-- externalPort:
15508
-- SessionID:
1
-- Direction: IsTransmitter
-- Started logical channel: sending
G.711-ALaw-64k{sw}
-- channelsOpen = 1 -- Connection
Established with "987654321, 987654321 [192.168.1.50]"Aug 30 11:53:35
DEBUG[114731952]: channel.c:1666 ast_set_write_format: Set channel
H323/ip$192.168.1.50:1260/5626 to write format
GSM =*= In
CreateRealTimeLogicalChannel for call
5626
-- externalIpAddress:
192.168.1.201
-- externalPort:
15508
-- SessionID:
1
-- Direction: IsReceiver --
Started logical channel: receiving
G.711-ALaw-64k{sw}
-- channelsOpen = 2Aug 30 11:53:36 DEBUG[114731952]: rtp.c:1156
ast_rtp_write: Ooh, format changed from UNKN to ALAW
Any idea about this "H245
Read error: Interrupted system call" that appears in the debug
messages???
Thanks, Damien.
BTW, the H.323 channel has been compiled with the
recommended PWLib 1.5.2 and OpenH323 1.12.2.
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Re: [Asterisk-Users] Sip device not login or register calls to that device go to busy voicemail not un-available
From my observation, if a call cannot be successfully placed then execution
goes to n+101. So for example if a phone is busy then the call can't be
placed (channel can't be created) and you jump tp n+101 which is typically
voicemail busy. In the case of a phone being offline then the call cannot
be placed and hence the jump to n+101 and again the busy message.
What I did in this situation is have some logic at n+101 that checks
$DIALSTATUS and then takes action as appropriate (In this case forward to
reception, but easy to send to unanswered voicemail instead - specify
unanswered voicemail at line 106)
exten = _123495XX,1,Macro(SetCID,${CALLERID})
exten = _123495XX,2,Dial(SIP/${EXTEN:4},16,tr)
exten = _123495XX,3,Voicemail,u${EXTEN:4}
exten = _123495XX,4,Hangup
exten = _123495XX,103,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?106:104)
exten = _123495XX,104,Voicemail,b${EXTEN:4}
exten = _123495XX,105,Hangup
exten = _123495XX,106,Dial(SIP/9500,20,tr)
exten = _123495XX,107,Voicemail,u9500
exten = _123495XX,108,Hangup
exten = _123495XX,207,Voicemail,b9500
exten = _123495XX,208,Hangup
Craig
- Original Message -
From: Ariel's Hotmail [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 30, 2004 2:11 AM
Subject: [Asterisk-Users] Sip device not login or register calls to that
device go to busy voicemail not un-available
I feel this is in error some place. If I call a sip device that is not
registered or not connected at the time. Asterisk will send that call to
voicemail to busy not unavailable. Is there a way to correct this?
Ariel Batista
Kasi International - Computer Networking
Ph: 305-574-6721
Fx: 305-574-0212
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[Asterisk-Users] Bridging audio in cmd_dial() before connect completes?
Is it possible to make cmd_dial() bridge the audio going out to the network back to the calling party as soon as dial() starts? Put another way, is it possible to have the caller hear the outside dialtone and subsequent DTMF digits? I notice that there is an option 'r' to dial(), thus: r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Which implies that the caller should normally be able to hear the network side ringing/busy etc., but is ambiguous on the actual dial sequence. I ask because I'm using EM tie lines from a Norstar, via Asterisk and I get no audio at all after dial() and before the connect status is reached. I'm using in-band signalling at the moment and the 5 to 6 seconds of 'dead line' during dialing is confusing my users. I tried on hold music during connect (option 'm'), but that confused them even more... For now they have grudgingly accepted an 'outside transfer' playback before the silence period. I have tried including an Answer() before the dial to patch the audio, but with no change. Obviously opening the channel to two-way audio before the dialing sequence is complete would be a security problem so, any suggestions? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ 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] Asterisk and codecs?
Hello, How many different codecs support Asterisk? Where can I find more detail information? I read Digium sell G.729 codec license. Is it support all differentformatted of G.729 codecs or just one? Regards, Balgaa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POE
Hi Steve 3COM 4400 PWR 802.3af switches (there's about 6 on E-Bay at the moment) and then use 3cnjvoip-cpod to convert the 802.3af feed to be used with the Cisco 79xx phones (it's specifically designed to do this). I'm using this setup (also with 3cnj205 wall switch for traffic prioritisation) on 15 Cisco 7960's and not a PSU in site...works great. Paul -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Saturday, August 28, 2004, 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] POE Steve Maroney wrote: Hey guys, I was wondering what POE solutions are being used ? Ive done some searching on google and found that PowerDsine seems to be good brand. Any comments,suggestions, and experiences on POE hubs other POE products would be greatly appreciated. Thank you, Steve Maroney ___ 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 Since the Cisco 79XX phones preceded the PoE standard, they are different--polarity is reversed. IANAE, but as I understand the PoE devices, there are two types--one always applies -48VDC to the brown pair while the other senses (as per the PoE spec.) whether the device at the other end requires power. I'm not willing to risk a $300 Cisco set, so I'm still using the wall wart. Is anyone providing LAN power to 79XX phones at a reasonable cost? Mike ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting
I have been unable to get the asterisk voicemail to work reliably with broadvoice. -Original Message- From: James Jones [mailto:[EMAIL PROTECTED] Sent: Sunday, August 29, 2004 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting if you have anyone questions about your service you can contact us at the support 978-418-7300 James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Ben Wern Sent: Sat 8/28/2004 4:34 PM To: Asterisk Users Subject: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting Can anyone who is using Asterisk with Broadvoice tell of their experiences with 3-way calling and call waiting? I can't get Broadvoice to respond to my question, but I understand that there is a per minute fee (3.9 c/minute?) if you go over your use allowances. My question is, how are 3 way and call waiting calls handled? Because Asterisk would just handle them as two different channels/calls -- does Broadvoice allow BYOD customers to have two active lines and then start charging for a third? If so, does anyone have any configuration examples of limiting the number of sessions to a single provider? Ben Wern ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap channels stuck in 'Rsrvd' state
Good Evening I found your post about this problem. Did you ever find a fix for it? I'm experiancing the same problem. Thanks. Quoting Steve Creel [EMAIL PROTECTED]: I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets stuck off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being read as a flash instead of a hangup?). A 'soft hangup zap/27-1' will not do anything (though it makes an attempt). Does shortening the rxflash time sound like it may help this? (Does anyone have a good explanation, or link to one, of the prewink, wink, preflash, flash, start, rxwink, rxflash, debounce timing functions?) Thanks, as always... Steve ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Telephony with Asterisk book
Steve Underwood Wrote: Joseph Shi wrote: Does anyone know if there are any reseller for the book VoIP Telephony with Asterisk in Hong Kong/Asia region? I'm interested in purchasing the book but the shipping charge to Hong Kong is expensive. Thanks. Joseph Just wait for the simplified Chinese version to appear in Shenzhen's Book City. :-) That's great! Will it have the English version as well? Any idea when it will be there? Thanks, Joseph. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
[EMAIL PROTECTED] wrote: On 30 Aug 2004 at 0:26, Steve Underwood wrote: [EMAIL PROTECTED] wrote: Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss? How much work would be required to change this? I guess it couldn't really be an option because of the totally different structure... Would it be possible for one person to make those changes or would it require the authors of all modules to recode? I haven't even completed by soft fax machine, and you are trying to be it completely useless. :-) Think about that. What you are suggesting is not really a satisfactory solution to anything, but certainly breaks things. :-\ Is this English?! my soft fax? make it completely useless? Okay, I think I understand you now... This surely wouldn't concern your code unless your code does it's transmission via IAX, SIP, OpenH.323 etc? And unless I'm gravely mistaken fax won't work over IP anyway... Is this a well thought out response? FAX won't work over IP? Doesn't changing the timing in the core of * affect the PSTN channels as well as the IP ones? Doesn't everything - caller ID, my soft fax machine, SMS, etc. - that works within * all go through the * core? Won't this screw up everything just to keep you happy? Won't this actually fail to keep you happy, since you don't seem to have thought through the whole jitter buffering issue, anyway? So many questions. So few meaningful answers :-) Regards, Steve ___ 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] ${CONTEXT}
I have some problems with my extensions.conf. When a call from pstn comes
in, the call gets put into the [from-fxo] context. From there the caller
is able to dial sip extensions that are included from the [sip-extenions]
context.
When a sip extension is dialed and connected, and then at some point
transfered, the ${CONTEXT} variable is changed from [from-fxo] to
[from-sip]. This leaves the caller from the pstn open to all extenions
that normally only my sip (trusted) clients would be able to dial, such as
outgoing calls on my other FXO ports.
Is the changing on the ${CONTEXT} variable by design (and needs to
secrured in my dialplan) or a bug ?
Thank you,
Steve Maroney
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[Asterisk-Users] Still unacceptable echo on X101P
I am still having unacceptable echo on my X101P and twidling with the rx/tx gain levels and echo settings appears to have no discernable effect. Some questions for those who may have more significant electrical engineering background than I. 1. This impedance match thing ... will it affect this solution having other phones in parallel with the X101P? This is done so that I can test while not having the system pickup/handle all the calls in the house until I'm ready to launch it. 2. What about the effects of it being downstream from a DSL line filter? 3. If impendance mismatch is the (or a major contributing) factor, can we not devise some interface circuit which will allow a variable rate on the impedance so we can dial out the echo based on individual line conditions? dbc. -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
On 30 Aug 2004 at 10:38, Steve Underwood wrote: [EMAIL PROTECTED] wrote: On 30 Aug 2004 at 0:26, Steve Underwood wrote: [EMAIL PROTECTED] wrote: Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss? How much work would be required to change this? I guess it couldn't really be an option because of the totally different structure... Would it be possible for one person to make those changes or would it require the authors of all modules to recode? I haven't even completed by soft fax machine, and you are trying to be it completely useless. :-) Think about that. What you are suggesting is not really a satisfactory solution to anything, but certainly breaks things. :-\ Is this English?! my soft fax? make it completely useless? Okay, I think I understand you now... This surely wouldn't concern your code unless your code does it's transmission via IAX, SIP, OpenH.323 etc? And unless I'm gravely mistaken fax won't work over IP anyway... Is this a well thought out response? Not really! :-) It was 5am...just before I went to sleep... FAX won't work over IP? Unless you use T.38 or are connecting to a machine with no lag/packet loss. We use your software to convert from fax to tiff, email the file, tiff to fax at the other end... Doesn't changing the timing in the core of * affect the PSTN channels as well as the IP ones? The PSTN channels (TDM400P) are already clocked to the 1000hz interrupts. The T1/E1 channels are clocked to the card remote end. IP is clocked to incoming packets. (even though not all of those packets are sure to arrive etc) Doesn't everything - caller ID, my soft fax machine, SMS, etc. - that works within * all go through the * core? Through is an interesting word, but interfaces with, yes. Won't this screw up everything just to keep you happy? One would hope not! And it's not just to keep me happy. I just noticed that we seem to have a few problems at the moment that could be resolved by not using the incoming packets as a clocking source. (i.e. Silence detection, Packet Concealment, JitterBuffering etc etc) Won't this actually fail to keep you happy, since you don't seem to have thought through the whole jitter buffering issue, anyway? LOL. 1. Keeping me happy means assuming I'm happy at the moment. 2. Yes I have thought through the jitterbuffer issue as it seems to be causing some problems here (we get clicks etc as it's size is changed by a large amount). So many questions. So few meaningful answers :-) I'm sorry I'm not sure what to respond to this. Regards, Steve Kind regards, Matt Riddell P.S. I LOVE YOUR WORK! I.E. SpanDSP it seems to be working here really well...what's up with the site though? Been down for a while. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue Announcement not until after # acceptcallpressed
We're using the patch and it's working alright aside from the MOH suspension
issue. I've got a C guy in our office I could put on the problem if anyone
can tell me in general what needs to happen. (I tried to figure it out
myself but haven't worked in C in nearly 6 years...)
-Corey
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward Eastman
Sent: Saturday, August 28, 2004 11:36 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: {S} RE: [Asterisk-Users] Queue Announcement not until after #
acceptcallpressed
This is something I'm after as well, what I have found is the following:
http://bugs.digium.com/bug_view_page.php?bug_id=0001082
http://lists.digium.com/pipermail/asterisk-dev/2004-February/003201.html
which pretty much does what I(you) want, the one problem with it is that
while the agent is listening to the pre # announcement, MOH for the queued
party stops. Other than this I can confirm the patch works well with CVS
Head 08/03/04.
Does anyone else have anything better, or any status on the above patch?
Thanks
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Brown
Sent: 27 August 2004 15:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Queue Announcement not until after # accept
callpressed
When using the callback feature on agents I notice that when the queue calls
one of the agents and the agent picks up the call they hear nothing until
pressing the # to accept the call.
Only then does my announcement play back to the agent after which the call
is immediately connected.
Is there a way to have the announcement played to the agent before they
press # to accept. I have ackcall=yes in agent.conf
Can't find anything on the wiki.
Thanks
Andrew
[exten.conf]
exten = s,1,Answer
exten = s,2,background(custom/100)
; Sales
exten = 1,1,ringing(2)
exten = 1,2,playback(custom/101)
exten = 1,3,queue(sales)
[queue.conf]
[default]
;
; Default settings for queues (currently unused)
;
[sales]
music = default
announce = sales_queue; This not played until after # pressed .. How can
i get announce to play as soon as call answered?
announce-frequency = 20
strategy = roundrobin
timeout = 15
retry = 5
maxlen = 0
member = Agent/7001
member = Agent/7005
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[Asterisk-Users] AgentCallbackLogin by other means
Hi, Were looking at options for logging agents into the system programmatically via Perl/PHP and I was wondering if anyone else is doing this and if so, how. We're using AgentCallbackLogin now but would like to set up a web interface instead. I've been looking at Asterisk::Manager and didn't see anything relevant and wanted to ask the group before we dove into the Asterisk source. Any input would be immensely appreciated... -Corey -- Corey S. McFadden ([EMAIL PROTECTED]) McFadden Associates - Technology Consultants phone 215-825-2121 ext 510 - web.csma.biz * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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] zaptel configuration
Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: installed asterisk: make clean make linux 26 (for fedora core 2) make install installed zaptel: make clean make make install did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device :00:0f.0 wcfxo: daa mode is 'FCC' found a wildcard fxo: wildcard x101p ... in zaptel.conf: fxsks=1-2 in zapata.conf: signalling = fxs_ks channel = 1 channel = 2 yet when i run asterisk, the zap show channels command doesn't work. in a previous thread they mentioned this is because some chan_zap.so file isn't loaded because of the zaptel installation. I was told I had to REINSTALL asterisk after the zaptel stuff, which again didn't do anything. How can this be so hard to even get installed? Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extens and number converting so that i can dial following one standaard.
On Sun, 29 Aug 2004, Johannes van Hulst wrote:
For asterisk I am using more than one sip providers.
The provider in Holland would like to have the international calls like
00 31 20 1234567 but the provider in the US likes it like 0011 31 20 1234567
Can I make a rule in asterisk so that I can dail 00 31 20 1234567 and
asterisk dails 0011 31 20 1234567 to the US provider?
sure. You'd use a Dial() command like this for the provider in Holland:
Dial(SIP/[EMAIL PROTECTED])
and something like this for the provider in the US:
Dial(SIP/0011${EXTEN:[EMAIL PROTECTED])
so to route any extension starting with 0031 through the US provider:
exten = _0031.,1,Dial(SIP/0011${EXTEN:[EMAIL PROTECTED])
for example. You didn't mention how you want asterisk to know/decide which
of the two providers a particular extension should be routed to. You'll
likely need to write a different exten = line than the sample I gave.
Greg
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[Asterisk-Users] Help debugging voicemail problem
Hi, I am fairly new to asterisk. I am currently testing my first setup. I've been able to debug most of the problems to make asterisk work with my hardware setup until this time. Currently I have the following issue: Voicemail is running but when I test to leave a voicemail thru my incoming PSTN channel (voicetronix / vpb), asterisk will not detect sound (according to the log) on that channel and outputs the following: -- Executing VoiceMail(vpb/1-1, u3001) in new stack -- Playing 'voicemail/default/3001/unavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/3001/INBOX/msg0001 format: wav49, 0x8149590 -- x=1, open writing: /var/spool/asterisk/voicemail/default/3001/INBOX/msg0001 format: gsm, 0x81496b0 -- x=2, open writing: /var/spool/asterisk/voicemail/default/3001/INBOX/msg0001 format: wav, 0x81497c0 Aug 30 00:05:07 WARNING[19475]: app_voicemail.c:1442 play_and_record: No audio available on vpb/1-1?? -- User hung up -- Executing Hangup(vpb/1-1, ) in new stack == Spawn extension (incoming-pstn, 3001, 4) exited non-zero on 'vpb/1-1' == vpb/1-1: Hangup requested == vpb/1-1: Ending record mode (1/yes) vpb/1-1: stopped record thread on vpb/1-1 == vpb/1-1: Ending play mode on vpb/1-1 vpb/1-1: Setting state down == vpb/1-1: Hangup complete Restarting monitor Trying to reawake monitor Monitor restarted Monitor got null event Any advice/pointers/suggestion are greatly appreciated :) Lethol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel configuration
On Mon, 30 Aug 2004, Imran Akbar wrote: Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: installed asterisk: make clean make linux 26 (for fedora core 2) make install installed zaptel: make clean make make install did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device :00:0f.0 wcfxo: daa mode is 'FCC' found a wildcard fxo: wildcard x101p ... in zaptel.conf: fxsks=1-2 in zapata.conf: signalling = fxs_ks channel = 1 channel = 2 yet when i run asterisk, the zap show channels command doesn't work. in a previous thread they mentioned this is because some chan_zap.so file isn't loaded because of the zaptel installation. I was told I had to REINSTALL asterisk after the zaptel stuff, which again didn't do anything. How can this be so hard to even get installed? Thanks, Imran Assuming your zaptel.conf and zapata.conf files are correct, you should have to issue a ztcfg -v. The output you see should match your configuration. In the future if you make changes to your zap*.conf files, you need to stop asterisk, re-run ztcfg, then restart askterisk. The reload command doesn't do anything with the zap*.conf files. I hope this helps. This is first time helping on this list. Thank you, Steve Maroney ___ 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
