[asterisk-users] Nokia E60/61/70 and SIP
Hi list, Just wondering -- has anyone used the SIP phone feature on the Nokia E60/61/70 phones? We're trying to see if this would be an OK phone to get for the company, particularly since we're already running Asterisk. Not asking for a review of the phone, but rather how well the built-in SIP client works. Link: http://reviews.cnet.com/4520-6454_7-6358771-1.html?tag=txt Thanks, El Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Crazy Boy wrote: Hi, Here I am posting my problem. I am getting this problem since 8 days. I have studied documentation and looked previous posts in forums. But, I am unable to solve this problem. Please show me a solution. I am from India. We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN-Eg. Mobile), I am not getting the callerid number of the caller and getting callerid as Asterisk in my softphones (XLite). SNIP When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console: Error Message: Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8) Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread: CallerID feed failed: Success Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1' Please tell me the solution. Looking forward to your kind response. Do you actually _HAVE_ caller ID on that PSTN line? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk GUI
Barry Fawthrop wrote: Tried to install and get this extension_dir does not exists /usr/lib/php/extensions/no-debug-non-zts-20020429 when entering index.php Any Ideas ? Could you write down what is exactly displayed on the screen, and send it via email to [EMAIL PROTECTED] It looks like there might be something to do with your php.ini settings. The LMS application requires you to have the Zlib extensions installed. Best regards, El Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk GUI
Barry Fawthrop wrote: Tried to install and get this extension_dir does not exists /usr/lib/php/extensions/no-debug-non-zts-20020429 when entering index.php Barry, Could you also check and make sure that PHP is running properly on the server? Perhaps PHP is trying to load some extension, but can't find where the extension is located. Regards, Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Asterisk GUI
Hello, We've just released our Libero Management System application, a web-based interface to configure and manage your Asterisk-based PABX. Designed for the not-so-novice Asterisk administrator in mind. LMS is simple to install, has minimal requirements (no external databases or components required), and runs on Apache and PHP. Some features: * Point-and-click dialplan configuration * Javascript-based real-time PABX monitoring tool * configure Zap/IAX/SIP channels/trunks/extensions * Zaptel interface configuration * CDR reporting * Sound file and MOH management. A 30-day evaluation version is available. For more details, please see http://www.lanvik-icu.com/lms/index.php Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Stops Receiving Calls
Alex Robar wrote: Hi all, I have a Sangoma A200 card with hardware echo cancellation. The card has 12 ports (10 of which are active; All FXO). Twice on this particular card I've seen all ports simply stop receiving incoming calls. There is no other indication of this, however. I am able to place outgoing calls just fine, and call other extensions without issue. When someone calls in, the line just rings and rings, with no indication that the card even sees the calls. I'm not even sure where to begin looking into this. Could anyone give me some pointers as to what I might need to be looking for? Did this just happen, e.g. was your system working fine before? Does it happen randomly, or have you seen any indication of a pattern of behavior? Perhaps if you could post your zaptel/zapata configs, and maybe some CLI output when this happens, it would be easier for us to help you out. We've got a client who's been using an A200 with 24 ports over the past 7 months, without any problems like what you mentioned. Cheers, Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keep Zap Channel from answering
voiplist wrote: Anyone know how to keep an Analog Zap channel from answering? I know I can answer it and send it to voicemail or do any number of other things with it once it's answered. I want to keep Asterisk from answering it, completely ignoring it while still having the line connected for outgoing purposes. assuming the line is attached to the trunk context, try the following in your dialplan: [trunk] exten = s,1,Congestion exten = s,2,Hangup Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP configuration by group
Sharon Lim wrote:
Hi there,
I would like to ask, is it possible to group sip user? Means group A with
sip user 100,200 and group B with sip user 100,200?
thanks in advance.
in your dialplan, define the following variables:
GROUP_A=SIP/100SIP/200
GROUP_B=SIP/150SIP/200
and in your dial string
exten = blah,1,Dial(${GROUP_A})
exten = moreblah,1,Dial(${GROUP_B})
Flynn
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Re: [asterisk-users] DTMF detection and Sangoma cards
Christopher Snell wrote: Hi, I posted earlier about Call parking breaks suddenly. I believe that I have narrowed this down to a problem with DTMF detection and the Sangoma A101 card that we use. Earlier, DTMF detection was not working at all. Then, I set 'relaxdtmf=yes' in zapata.conf and it works...sort of. When I call into the PBX from a digital desk phone, Asterisk is able to detect the agent's DTMF and parks the call as requested. However, if I dial in from my cell phone, the agent's DTMF is not detected and the caller (me) hears the DTMF on the lines. Does anybody have any ideas? We've been running a Sangoma A104 at a client site for the past 12 months without any DTMF issues whatsoever, neither from the inbound nor the outbound side. That unit's connected to a multitude of analog and IP phones, as well as a large legacy PABX behind the * box. Some numbers on the PRIs are provisioned to hit a lengthy IVR menu tree, so I know the DTMF works. Bear in mind it's running much older versions of Asterisk, Zaptel and libpri. Are you only having this problem for call parking? Any issues when the caller is navigating an IVR? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI tutorials
Rizwan Hisham wrote: Anybody who knows a good source of AGI tutorials on the net? plz share How about the Asterisk Wiki? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] menu system - configurator
bram kortleven wrote: We are currently looking for a way to easily configure a 'auto attendant' system on our asterisk pbx. More in detail, I'm looking for a webbased (or something similar) configuration generator, that has a feature like asking me how many 'menu levels' I want, what text to play, and in the first, how many items I want, and then per item what text and description it has to set, etc ... Is there something out there that does something similar? Or does anyone know how to make such a script? If possible, we prefer mysql-driven menu's... as all other stuff is in mysql already... Look up Asterisk GUI in the Wiki. There's plenty to choose from. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
Rizwan Hisham wrote: Hi, i need to set the dtmf mode on my quintum tenor a400 gateway. You might want to check the a400 manual on how to do that. i cant dial any extension thru my normal digital phone which is connected to asterisk thru the quintum gateway. it always falls to 's' extension. So plz help This is most likely a misconfiguration of your dialplan and/or sip.conf files. it would help if you post it here? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 woes
Jim Lynch wrote: I attempted to install a new A200 module with one each FSX-2 and FXO-2 module. I connected an internal power connector to the board as instructed, but when the system reboots, it just beeps at me. It doesn't even let me get to a bios prompt. I removed both of the modules and it still behaves in the same way. The only other boards in the system are the ethernet and video boards. Does it still not boot when you've removed the card from the box? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Hardware Reliability
M.Hockings wrote: Even now, given that I don't know what caused the problem or what solved the problem (for the time being). I might expect that powering the system off may cause software errors due to partially written files but I would NOT expect it to damage the hardware, particularly just a comm card. Hence *my* feeling for *this* card is that it is unreliable. It is however reassuring to hear that overall the reliability of the Digium hardware is good. When you say the card just worked after it apparently went dead, did you switch it around to a different PCI slot? Or did you leave it in the same place, and after some time it worked again? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to ask for number to dial and then dial it?
Robert La Ferla wrote:
I want to create an extension say 8000 that prompts the user to enter
a number and then dial that entered number according to a set of
rules. The rules for dialing out are in different context (dial-
out-rules).
[some-context]
exten = 8000,1,Playback(please-enter-the-number)
; toll-free numbers out pots line
exten = _1800XXX,1,Dial(${ANALOG_POTS}/${EXTEN})
exten = _1800XXX,n,Hangup()
; long-distance out voip line
exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],30)
exten = _NX,n,Hangup()
exten = i,1,Playback(that-number-is-invalid-ha-ha)
exten = i,2,Congestion
And point the related extensions to [some-context].
Flynn
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Re: [Asterisk-Users] recording all calls patch through asterisk
Michael Sampson wrote:
Basically I will have a call come in a PRI trunk and be routed out the
same PRI trunk. The point of this is so I can use asterisk to record the
call. Has anyone set up a system like this? I know how to get asterisk
to record a call from and extension, but not a call that is just
passing through the system.
I'm assuming the call comes in through one PRI line (Zap group 1), and then goes
out again via another PRI line (Zap group 2) into some other device.
[incoming]
exten = _X.,1,MixMonitor(${UNIQUEID}.gsm))
exten = _X.,2,Dial(Zap/g2/${EXTEN})
[outgoing]
exten = _X.,1,MixMonitor(${UNIQUEID}.gsm)
exten = _X.,2,Dial(Zap/g1/${EXTEN})
make sure to set Zap group 1 to the incoming context and set zap group 2 to
the outgoing context
Flynn
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Re: [Asterisk-Users] Voicemail
Khaled Chehab wrote: I am using trixbox,I want ot disable and enable voicemail from command line At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully snip But at trixbox its not working Any ideas pleas Did you try checking with the people who _wrote_ trixbox? Perhaps they have a forum or at least mailing list of some sort that could answer your question(s)? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 hangup detection
chan (Alpha Trilogies Networks) wrote: Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? snip [channels] context = from-pstn3 switchtype = national usecallerid=yes hidecallerid=no transfer=yes echocancel = yes echocancelwhenbridged = yes echotrainning = yes busydetect=yes busycount=1 callprogress=yes relaxdtmf=yes rxgain =-2.5 txgain =-2.5 signalling=fxs_ks group=1 channel=3-4 Any advice? A couple of things: 1. The switchtype setting is only for PRI lines. 2. Try setting callprogress=no, call progress analysis is supposedly only valid in the US. 3. Tune your gain settings until you get an optimal signal level -- google the list or the Wiki, it's quite thoroughly documented. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Hardware Reliability
M.Hockings wrote: snip Today the power went out due to a mis-configuration on my part the UPS shut down before the machine shut down. Now, I would not think this should be a problem but the Digium card no longer responds. lspci does not show it either so I presume it dead While I don't use the cards anymore, I think it's a bit harsh to immediately point the blame to the card. It could be that the power outage affected the PCI slot where your card sits. Have you tried moving it to another slot? Have you tried using the card on a different machine? Do that first before immediately knocking Digium/Sangoma/whatever off. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Cancellation
Douglas Garstang wrote: General question. If you install a Digium card in an Asterisk system, and install zaptel drivers, do this give any benefit of echo cancellation? Our PSTN gateway is a separate Audiocodes box, so the zaptel card wouldn't actually be connected to anything. I'm wondering though doing this would help, in general, with echo cancellation. I wouldn't think so, since the echo cancellation settings are in zaptel-land. So the call path would have to go through a zap channel before the echo canceller can work. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with incoming SIP routing
Christopher Aloi wrote:
Hello -
I currently have 10 DID's coming into one Asterisk server, I seem to be
having some difficulty routing based on the DID dialed and am hoping
someone
on the list can assist me.
snip
Unless I'm misunderstanding you, how about trying this:
1. In your sip.conf:
[general]
useragent=Asterisk
port=5060
context=default
tos=lowdelay
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtptimeout=300
rtpholdtimeout=600
2. In your extensions.conf:
[default]
exten = s,1,Goto(${CALLERIDNUM},s,1)
[123456789]
exten = s,1,Answer()
exten = s,2,Playback(beep)
exten = s,3,GoTo(queue-test,s,1)
So if you get an incoming SIP call from 123456789, it enters the default
context and is then routed to the 123456789 context.
Flynn
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Re: [Asterisk-Users] Call length limitation
Andrew Nowrot wrote: Hi I have a problem with Dial application. The dialplan looks like this: ; exten = x,1,Dial(Sip/|30|L(6:3:1)) exten = x,2,Hangup() exten = h,1,DadAGI() ; The call is limited to 60 sec and after that time the conversation stops, but Asterisk never reach the h extension. I could use the S() option in Dial application but I want to have the announcements. Is there any way to force Asterisk to execute the DeadAGI. Or maybe there is some other solution to achieve a goal. What does the CLI show when you make the call? That might help in diagnosing your problem. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail number of recorded messages
Khaled Chehab wrote: How can I limit extension voicemail messages to 10 messages per user ? If you look in the voicemail.conf.sample file in the source, you can find the following lines: ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is . ;maxmsg=100 So you would set this to 10 for your voicemail.conf Cheers, Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] massive screetch and echo from Treo 700w
Curt Shaffer wrote: snip the iax.conf config but the sound is ridiculous. The echo is horrible and there is a screeching in the background on the receive end. Is there anyone snip You could also discourage your pet parrot from playing around with the phone... haha its been a long day... Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] analog call progress - can I use backgrounddetect
Jerry Geis wrote: Hi, There seems to be no solution for call progress on analog lines and using outgoing spool call files . My wave file starts playing before the person has answered the phone so the first part of the message is missed. Can the backgrounddetect app be used for this. I have tried but the message still plays before I answer. I generated 60 seconds wave file. Some other approaches: 1. Have you tried setting callprogress=yes in zapata.conf? This will only work if you're in the US though, and your mileage may vary. 2. An unelegant approach without using backgrounddetect would be to set it up like so: - playback a prompt that asks user to press a key to accept the call - only deliver the audio payload once user has pressed a key You could also tinker with #2 to cater for scenarios where you've hit a voicemailbox, e.g. after two/three loops of the first prompt just deliver the message anyway and hope it gets recorded in the callee's mailbox. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiicemail NFS Cutting Out
BILL GITONGA wrote: I have two asterisk systems that share voicemail on an NFS. I recently upgraded to Asterisk 1.2.9.1. After the upgrade, the voicemail gets cut out after about 5 seconds of recording. Any ideas on what might be causing this? What does it show on the CLI when this happens? More info would be helpful. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.
Anderson Ling wrote: Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. [root]# cat zapata.conf snip callgroup=1 pickupgroup=1 immediate=no useincomingcalleridonzaptransfer=yes busycount=4 callprogress = yes You might not want to use callprogress, as this is more related to US type of call progress analysis. The tones are different in Malaysia. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limit to number of queues
Hello, Does anyone know the maximum number of queues that can be defined in an Asterisk system? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB headsets?
[EMAIL PROTECTED] wrote: We have some laptop soundcards that are really bad and I would be glad if you could share your experiences when changing to a USB headset instead of using the built in soundcard in your computer. Well, IMO if the soundcards are already crap to start out with, there's no way a fancy-schmancy USB headset -- or any other headset, for that matter -- will sound good when plugged in to the laptop. Because, remember, it's the soundcard that generates the audio and sends it out the heaphone port. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] USB headsets?
[EMAIL PROTECTED] wrote: I don't quite follow you? There are USB headsets that don't require a soundcard at all. They have a built in soundcard which (I suppose) could be better than the crap they build into most laptops. well, slap me around and call my silly :) I haven't ever used one of those, so I guess it's time I caught up with what's out there.. heheh ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web Admin
Sharon Lim wrote: Are you looking for an web interface that write to asterisk config files? if yes, you can look at freepbx.org . Hello, just out of curiosity -- are you based in Malaysia? Cheers, Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help -- voicemail
chan (Alpha Trilogies Networks) wrote: Hi, Did someone experience that Asterisk OS 1.2.5 voicemail issues? Problem description: Some one call to the extensions 200, After 10 sec ring then go to voicemail [EMAIL PROTECTED] Announcement Please leave me a messages.blar blar.. When I completed to leave a message... IF : I press the pound #key ... Then it says Transfer IF : I Press the zero 0key Then it say Please confirm your recording IF : I hangup after leaving a message...then things get normal. check your Dial command, looks like you've enabled the CALLER to transfer -- which is why you get the Transfer when you hit the # key. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stability and motherboard questions with TE406P and TE410P
Kyle Sexton wrote: TE410P: - zttest will never report 100% for me across different motherboards (Supermicro P8SCT, Dell 850) - Crash/instability of about once per two weeks where I have to power cycle the server, i.e. phone calls stop working and a reboot fixes it TE406P: - zttest runs flawlessly on this card, 100% across the board - PRIs will go from up and working fine, to Provisioned, Down, Active after the server has been up for around 10 minutes, this may be related to rxfax and txfax being installed? Has anyone had an issue with this specific card? We have had this experience across multiple motherboards. After working with two different cards and across multiple motherboards I am starting to lose faith on the stability of the Asterisk platform, but I know others are having lots of success. My boss is looking for something that says With the TE406P, we have had zero issues on X motherboard, does anyone have any recommendations? Has anyone else had stability issues with the digium 4 port cards? Kyle, While my reply probably doesn't help you any, I just want to say that I've been experiencing the same sort of problem. I've got a TE410P on a server with an Intel SE7210TP1-E Entry level server motherboard; we're connecting the card to four Rhino T1 channel banks. I'm also experiencing the random crash issue, albeit perhaps not as frequent. Some symptoms: - server hangs and is generally bogged down. Even when I'm at the console in front of the server typing one key on the keyboard takes 4-5 seconds to get a response - random noise, echo and badness starts to appear on the phones connected to the channel banks. This is because something's eating up the CPU processing power and the server isn't able to service the 1k interrupts the zap card requires One of the things I've had to do, as a jerry-rig type of fix is to have Asterisk restart every day at 3am. This has lengthened the duration in-between server crashes, but isn't really a good solution. What we're going to do is to scrap the TE410 and use Sangoma's A104 card. In the same installation, we've got a server (same _identical_ specs as the one above) with one A104 -- two incoming PRIs and two outgoing to an Avaya PABX. This one has not crashed since it went into production last august. This is probably just a rant, but I thought you'd like to know that you're not the only one struggling with the TE410 cards. Hope you'll be able to get your setup fixed soon, good luck. Cheers, Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native MOH - Convert mp3 to ulaw
John Novack wrote: Wavepad works well, without complaining about libraries, and you can even edit. listen to the results and back out,if need be. Harder to use for those who aren't sighted, though John Novack You could also use Audacity, which has a bunch of filters and effects that you can use, like echo, equalization, high/low pass, noise filters etc. Speaking of noise filters, I really like the one used in Goldwave. Quite useful for people who use el-cheapo mics and record in a somewhat noisy environment (yours truly included). Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot leave voicemail, Asterisk/Zaptel/libpi v1.0.9
Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP phone connected to Box A, upon leaving voicemail I get in the following: -- x=0, open writing: /mnt/nfs/123/INBOX/msg0004 format: wav49, (nil) Mar 21 17:28:18 WARNING[8576]: app.c:706 ast_play_and_record: Error creating writestream '/mnt/nfs/123/INBOX/msg0004', format 'wav49' Mar 21 17:28:18 WARNING[8576]: app_voicemail.c:787 base_encode: Failed to open log file: /mnt/nfs/123/INBOX/msg0004.wav: No such file or directory -- Executing Hangup(Zap/5-1, ) in new stack Browsing through previous posts and other resources led me to believe there was a permissions problem on the shared folder. As an extreme measure for testing purposes I've chmoded 777 the /mnt/nfs directory and all its contents, but the same problem persists. From Box A, I can also create new files etc in that exported directory from the command line, so I think permissions may not be the issue here. I've tried saving it in different formats - wav49, gsm, wav -- but problem still persists. Even stranger, while msg0004.wav is not saved msg0004.txt _is_ created in that directory! Box B can leave voicemail in the same directory without any problems, here's what I get when I reroute a call for extension 123 to Box B: -- Recording the message -- x=0, open writing: /mnt/nfs/123/INBOX/msg0004 format: wav49, 0x9fb5490 -- User hung up Any clues as to what may be the problem here? Could it be on Box A it's coming up with format: wav, (nil) while on Box B it says format: wav49, 0x9fb5490 ? Can anyone tell me what this might mean? Thanks in advance, Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming the Manager API
Douglas Garstang wrote: I just started poking around with writing a python module to interface to the Manager API, and it suddenly hit me... how the heck are you supposed to program this thing? All the events seem to be dumped to all the open connections. If I send a command, such as a login, there seems to me to be no way to determine which response are intended for me, and which may be intended for another open session. There's no reference number, or anything which indicates who events are for. This would seem to make it pretty much impossible to program at all. Anyone done this??? Doug. There's a Java packaged that contains numerous classes allowing you to interact with Asterisk. It interacts via the FastAGI protocol as well as the Asterisk Manager API. Check it out at http://asterisk-java.sourceforge.net/ Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming the Manager API
Douglas Garstang wrote:
Yikes. Java. Yuck. I'll stick with Python... Thanks anyway.
I just worked it out... you can supply an actionid to the request to know what
reply to look for, although it will still be tricky filtering out the noise.
Well, with the Asterisk java code it's pretty much cut and dried. Take a look at
some of the examples and you'll be surprised how quickly you can come up with a
workable app. It's as easy as:
if (event instanceof NewChannelEvent) {
/* Change icon color from green to red */
}
else if (event instanceof HangupEvent) {
/* Change the icon color from red to green */
}
The filtering bit is kind of done for you there so you won't have to muck
around with parsing what is thrown back at you by *.
Although you might want to download the source code and see how that parsing is
done.
Flynn
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Re: [Asterisk-Users] queue_log
Anton Krall wrote: Guys, anybody has some info regarding the format that queue_log has and how to interpret it.. I found some info on the wiki about the conditions of a call but the first fields I still don't know what they are for, although I can imagine one of them is a call identifier, etc. but want to be sure. Look for the queuelog.txt file in the doc directory within your asterisk source code, it's explained in there. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ncurses Asterisk Manager Interface
Hi Sig, I'm trying to compile the assman package, but some errors come up: dceptcons:/usr/local/src/libassman-current # make make -C libassman make[1]: Entering directory `/usr/local/src/libassman-current/libassman' cc -I../inc -Wall -c -o assman.o assman.c In file included from assman.c:8: manager_status.h:4:16: db.h: No such file or directory manager_status.h:9:16: db.h: No such file or directory In file included from assman.c:8: manager_status.h:57: error: parse error before '*' token manager_status.h:57: error: parse error before '*' token manager_status.h:57: warning: type defaults to `int' in declaration of `manager_database_init' manager_status.h:57: warning: data definition has no type or storage class manager_status.h:60: error: parse error before '*' token manager_status.h:61: error: parse error before '*' token manager_status.h:62: error: parse error before '*' token manager_status.h:63: error: parse error before '*' token make[1]: *** [assman.o] Error 1 make[1]: Leaving directory `/usr/local/src/libassman-current/libassman' make: *** [all] Error 2 is the db.h file missing from the current tarball? I just downloaded it a couple of minutes ago. cheers flynn Sig Lange wrote: I am currently developing a asterisk ncurses interface using the manager API. The project is currently awaiting sourceforge's approval but I have a beta online at http://sig.lange.googlepages.com/assman . The projects real home will be assman.sf.net. This project really consists of two parts, libassman is a C manager API and assman is the ncurses portion. It's still beta but I have been running it for quite some time on a production server w/o any major glitches. Soon as the sf.net approves the project I will have SVN and the latest versions online. Feedback is welcome as well as requested features. Thanks. -- Sig Lange http://www.signuts.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ncurses Asterisk Manager Interface
Sig Lange wrote: I am currently developing a asterisk ncurses interface using the manager API. The project is currently awaiting sourceforge's approval but I have a beta online at http://sig.lange.googlepages.com/assman . The projects real home will be assman.sf.net. This project really consists of two parts, libassman is a C manager API and assman is the ncurses portion. It's still beta but I have been running it for quite some time on a production server w/o any major glitches. Soon as the sf.net approves the project I will have SVN and the latest versions online. Feedback is welcome as well as requested features. Thanks. I'm just not sure if ASSMAN is an apt name for the project heheh.. Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ncurses Asterisk Manager Interface
Alexander Lopez wrote: That may be the best one yet, It is pulling the information out of Asterisk's BackEnd. :-) From the looks of the project's screenshots, assman needs to be able to handle a lot of shit coming out of the back end, for cases when a busy server is generating a lot of events... hahaha flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] saydigits
Jerry Geis wrote: I was searching on voip-info.org for saydigits. I see no indication it is not valid in 1.2.4 asterisk. however, when trying to use it I get and error no application saydigits. what is the correct way to echo back digits in asterisk 1.2.4? I tried say digits 123 and saydigits 123 both gave no application error Jerry, I have it on my box: demo*CLI show version Asterisk 1.2.4 built by root @ demo on a i686 running Linux on 2006-02-27 07:15:32 UTC demo*CLI show application saydigits demo*CLI -= Info about application 'SayDigits' =- [Synopsis] Say Digits [Description] SayDigits(digits): This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. See the LANGUAGE function for more information on setting the language for the channel. demo*CLI You might want to check if the application is loaded. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 error
Mike Clark wrote: snip Have you called Sangoma's tech support number? I just implemented the same card about two weeks ago and really didn't have any installation issues using fc3 and trunk, however their documentation is a little on the rough side. Install info seems to be a little in one file, a little in another, etc, and it assumes the reader has the same level of skills/knowledge as the Sangoma writer. If your cards are not fully populated with modules, then the placement of the modules does matter. Because we weren't sure of exact ordering of modules when we put them in place, I ended up having to configure channels 1-2, and 5-8 in zaptel.conf and zapata.conf. Also, we got some of the earliest production cards and had to upgrade the firmware from version 4 to version 5 for the hardware echo cancellation support. A couple of other pointers: 1. Each A200 card will occupy 24 channels. So if you have two A200 boards, the first one recognized will have zap channels 1-24, and the second one will have 25-48. This is true even if you only have one module on each card. So if you have 2 FXO ports on each card, your zaptel.conf will look like: fxsks=1-4 fxsks=25-28 2. If you plug in any of the remoras (the expansion boards) into an A200, you'll need to plug in the power supply, EVEN IF you are _not_ using any FXS modules. For those moving from Digium TDM400 to Sangoma, this can be tricky. Took a while to figure that one out. 3. For those running hotplug, you might have some issues with the A200 (as with the rest of the AFT cards). Make sure to disable hotplug before you load up the wanpipe drivers One thing I've noted is that starting up the wanpipe drivers take about 10 seconds or so. Kind of long compared to the wctdm drivers. However, their engineer has mentioned that this is due to some debug/initialize code during the driver load process, which will be phased out as things start to stabilize. Cheers Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 error
Michael Kenjie Nukui wrote: Hello, i am trying to install sangoma a200d to my centOS server but i am receivig this error message: ZT_CHANCONFIG falied on channel 1: invalid argument (22) How is your hardware set up? Do you have just the one A200 board, or do you have additional Remoras hooked up? What/how many modules are you using? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximizing audio quality
Wolfgang Borgon wrote: A RAW file I created after converting from MP3 and WAV, sounded raspy. Does anyone have any tips for creating the best quality voice recordings? Generally you'd use a good-quality microphone for your recordings. The adage Garbage in = garbage out couldn't be more true in this instance. If you're looking for studio-quality recordings, use studio-quality equipment. Those $5 mics won't be satisfactory :) Then there's issues of sibilance, which isn't that apparent when you're recording at a higher rate, but is really pronounced when you downsample to 8k for the GSM files. The raspiness you encountered was probably sibilance, where words that have the ess sound in them are boosted due to the position of the microphone relative to the person being recorded. If you're going the budget route, at least get a decent quality sound card to record with. Another important factor to consider is your recording location -- try and record in as quiet a place as you can find. Some audio processing software (Goldwave, Audacity et al) have filters that can knock out background noise, alter volume, apply equalization etc. You can use these effects to enhance the recording. But again, if your original recording already sounds bad there's not much you can do to make it sound nice. Cheers, Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Difference between CDR dispositions..
Hi there, I've got a client complaining about the dispositions in the CDR report we built for them: 1. User calls an extension, which rings three SIP phones in the group. Entry in extensions.conf: exten = 100,1,Dial(SIP/200SIP/201SIP/202) 2. On three test calls, she dials extension 100 and makes sure no-one picks up any of the three phones. 3. In the CDR, two of the calls' disposition are listed as No Answer, while on another CDR entry disposition is listed as Busy. Does anyone know what might cause this? TIA, Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip accounts
Kong wrote: can i know where to start? SIP is such a big topic. Try looking for SIP configuration (sip.conf) in the Wiki, it's got lots of examples. Or you can also try looking it up on google. Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help with broken voicemail
Andy Goss wrote: Oct 11 20:32:58 WARNING[3048]: file.c:415 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/default/5926/INBOX/msg.wav Oct 11 20:32:58 WARNING[3048]: file.c:793 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5926/INBOX/msg (format ulaw): No such file or directory can you check that /var/spool/asterisk exists, and that all its subdirectories are intact? perhaps it got deleted by accident somehow? you might also want to check the file permissions on the directories. flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clicks, pops and noise
Rich Adamson wrote: snip One other item to check is to ensure the digium T1 card is on its own dedicated interrupt. Use 'cat /proc/interrupts' from the system command line. It is on one interrupt, first thing I checked when the problem cropped up. One thing I did notice was interrupt latency when doing a 'lspci -v'.. should that number be 0? If so, does anyone know how to set that at boot time? Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clicks, pops and noise
Rich Adamson wrote: It is on one interrupt, first thing I checked when the problem cropped up. One thing I did notice was interrupt latency when doing a 'lspci -v'.. should that number be 0? If so, does anyone know how to set that at boot time? I played around a fair amount with the latency thing and could not identify any noticable differences. I doubt that making changes there will have any impact. Rich, Thanks for the info! That'll save me some time since I don't have to bark up the wrong tree :) On another note, I was told to double-check the memory on the server, _just_ in case that might be the source of all my problems. We're running the Memtest86 app overnight, maybe something will turn up tomorrow. Cheers, Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] adding new indication tones
oner asterisk wrote: Hi all, I would like to add indication tones , What I did is enter data to zonedata.c and indications.conf than compile zaptel. and restart asterisk. But it's not working what else I should do ? Regards, Öner did you check that the new tones are loaded in zaptel.conf? flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clicks, pops and noise
Hi all, I've got a TE410p connected via T1 to four channel banks (2 FXO and 2 FXS), no PRIs. Some users are complaining that they hear clicks and pops on the FXS lines, generally when they pick up the phone it's noisy. This happens only after a while, e.g. after a fresh restart of everything, all is fine but after some time these noises start appearing. From what I've read on the list, this could be down to frame slips or some problem due to synchronization. Since there's no incoming PRI to sync to, this means everything needs to be internally clocked. Could it be the internal clock source on the card has gone wonky? Or is something else in the server screwing up the clock signal? Anyone else experienced this when connecting four channel banks to the TE410? Zaptel.conf: span=1,0,0,esf,b8zs fxsks=1-24 span=2,0,0,esf,b8zs fxsks=25-48 span=3,0,0,esf,b8zs fxsls=49-72 span=4,0,0,esf,b8zs fxsls=73-96 TIA, flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one extension goes straight to voicemail, others don't
taran wrote: i have one extension going straight to voicemail, while others that are configured identically don't, so i don't think it's an overall config problem. nor do i think it's a callerID problem. maybe it's an enduser operation that i can't find documentation on? snip it would be helpful if you could post the pertinent part of your extensions.conf. Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk scenario
housi mueller wrote: I am new to asterisk and would like to know if a configuration like shown on the picture with asterisk is correct? Thank you in advace Housi Mueller Looks good ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Callgroups
Wolfgang Pichler wrote:
Hi all,
at time i am trying to get a better idea of callgroups and pickupgroups
(especially within the SIP Channel)
A Pickupgroup is relative clear - everyone in the same pickupgroup may
pickup a call
And a callgroup does what ? - The same ?
example:
phones A, B and C are configured with callgroup=1 in sip.conf
phone D is configured with pickupgroup=1 in sip.conf
when a call comes in to A, B, C or any combination of them, phone D can pick up
the call on their behalf.
I thought that a callgroup would act like the ZAP groups - so that you
then can dial SIP/g1 - and every SIP Client which is in the callgroup 1
does then ring - Why isn't this so - What sense does it make to define
callgroups - when you then have to specify each SIP Client in the Dial
Command to get the callgroup Working ?
You're mistaking group for callgroup. Zap example:
group=1
channel=1-24
this means when you do a Dial(Zap/g1) phones 1 to 24 would all ring. If you
wanted to emulate the same sort of behavior for your SIP phones, you could do
something like:
[globals]
SIPGRP1=SIP/100SIP/101SIP/102
then in your dialplan you could do a Dial(${SIPGRP1}) to ring all the sip
phones.
Flynn
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Re: [Asterisk-Users] Utility to find length of wav49 file
Malcolm Taylor wrote: Can anyone point me in the direction of a utility which will let me determine the length (in seconds) of a wav49 file created by Asterisk? Many thanks, Malcolm if you're talking about the duration of a voicemail, you could do: grep duration msg.txt from the command-line. each voicemail left has an accompanying text file that gives details about the message. unless you're talking about something completely different... Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Going crazy with FAX :-(
Michele O-Zone Pinassi wrote: I've upgraded Asterisk from CVS, spandsp and app_txfax and app_rxfax but i'm still unable to send/receive faxes :-(. I'm using amp_fax to send and this is what i get from logs: snip Sep 6 11:06:13 VERBOSE[10750]: -- Executing System(Zap/1-1, tiff2ps -2eaz -w 8.5 -h 11 /var/spool/asterisk/fax/1125997512.2.tif | ps2pdf - /var/spool/asterisk/fax/1125997512.2.tif.pdf) in new stack Sep 6 11:06:13 VERBOSE[10750]: -- Executing System(Zap/1-1, mime-construct --to [EMAIL PROTECTED] --subject Fax from --attachment .pdf --type application/pdf --file /var/spool/asterisk/fax/1125997512.2.tif.pdf) in new stack Sep 6 11:06:14 VERBOSE[10750]: -- Executing System(Zap/1-1, rm /var/spool/asterisk/fax/1125997512.2.tif /var/spool/asterisk/fax/1125997512.2.tif.pdf) in new stack snip but pdf is illeggible and TIFF corrupted ! You might want to check the other components in the chain -- tiff2ps and ps2pdf -- to make sure that your version of libtiff is able to create the .tif file properly. My suggestion is to: 1. Just receive the fax as a TIFF file, and see if you can open that up 2. If step 1 is okay, look at tiff2ps and make sure it's converting the TIFF file to postscript properly. Use ghostview or something like that to look at the output .ps file and make sure it opens up nicely 3. If step 2 is okay, look at ps2pdf and make sure it's converting the postscript file properly. We had some similar issues, ended up removing the tiff2ps and ps2pdf from the processing and emailing the TIFF file directly to the recipient. Cheers, Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI in and out
Rod Bacon wrote: I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty of asterisk work over the last 6 months to PRI circuits, but not with a PBX being involved. I know I can use asterisk and digium cards in this manner, but do I need separate cards for the PRI - Asterisk side to the Asterisk - PBX side, or will a 4-port PRI card do the job? (I already have a spare one of these). Short -- yes, can be done. We just did one install with two incoming E1s connected to ports 1 and 2 on a Sangoma A104, and ports 3 and 4 were outgoing to an Avaya PABX. In other words, can I use SPAN 1 as a timing source, then provide timing to the PBX connected to SPAN 2 of the same card? That should do it. You'd just need to configure zaptel.conf properly. There's an example somewhere on the Wiki about integrating Asterisk with legacy systems. Cheers, Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110p and E1
Stephen wrote: Hi All, I have configure my Asterisk as follow (using [EMAIL PROTECTED]): [zaptel.conf] span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 loadzone = uk defaultzone=uk try this in your zaptel.conf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 CRC required for E1 from telekom IIRC snip I am connecting my TE110P with a Balun converter where the two BNC connectors are from connected to a Fibre optic line. When I reboot my pc , I got my TE110P LED flashing RED slowly. But nothing happen when I hooked up to Balun Convertor. Balun convertor is 75ohm on BNC and 120ohm on RJ45. I am using straight through cable to connect my TE110P to Balun. Are you sure you have the zap modules loaded? Got the same type of setup going and it works fine, as far as the physical connection from E1 - balun - card. Try connecting it first and then loading the zap modules, although it shouldn't make any difference. In any case, start up asterisk with -vdgc and see what kind of errors are showing up. Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nested dial, or jump to another line to continue dialing.
Joseph wrote: Is it possible to do nested dial() command on one line, Dial number, wait new seconds, dial another number etc. or dial number and jump to another line to continue dialing. D(ww) doesn't work as it sends DTMF but before the call is bridged, and I need to send numbers after the call is bridged. If you do a show application dial at the CLI: snip 'D([digits])' -- Send DTMF digit string *after* called party has answered but before the bridge. (w=500ms sec pause) Hmm.. it does say that DTMF is sent *after* called party has answered. it's been working for me since asterisk-1.0-RC2 Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nested dial, or jump to another line to continue dialing.
Joseph wrote: Here is a session with D() exten = _51,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww218)) Executing Dial(SIP/11-3dec, SIP/[EMAIL PROTECTED]|30|D(ww218)) in new stack -- Called [EMAIL PROTECTED] -- SIP/pstn-5665-713c is ringing -- SIP/pstn-5665-713c answered SIP/11-3dec -- Goto (office-open,s,1) -- Executing Wait(SIP/pstn-1270-e0f5, 2) in new stack -- Attempting native bridge of SIP/11-3dec and SIP/pstn-5665-713c -- Executing Answer(SIP/pstn-1270-e0f5, ) in new stack -- Executing NVBackgroundDetect(SIP/pstn-1270-e0f5, welcome|t) in new stack -- Playing 'welcome' (language 'en') -- Executing Goto(SIP/pstn-1270-e0f5, 1|1) in new stack -- Goto (office-open,1,1) It is not passing DTMF(218) ---end session D()- Without looking at your dialplan for the context that SIP/4791270 belongs to, what's most likely happening is the pauses are too short or too long. I've just dialed out to my cellphone to test this, and yes i do hear the DTMF when I pick up my cellphone. You can create a simple test for this -- dial to your mobile phone or landline, pick up the call and see if the DTMF is passed. Flynn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Motorola A910 WiFi + GSM phone
Hi all, On the Wiki it says something about the motorola WiFi/GSM hybrid phone, the Motorola CN620. Don't know whether that one ever made it to the market or not, but I read a review on c|net about another upcoming model, the A910. The A910 is Linux-based, and offers WiFi on top of GSM, GPRS and Bluetooth. You can see the picture at http://asia.cnet.com/reviews/handphones/0,39001713,39094707p,00.htm? Looks like it also has color screen, but not much other specs listed yet. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 - two processes
Ronald_Wiplinger wrote: Brian West wrote: If you use mp3nb from the sample configs you will have exactly 1 per class. Great! Where can I read more details about it? (musiconhold.conf) in musiconhold.conf: [classes] default = mp3nb:/var/lib/asterisk/mohmp3 Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disconnecting a call on asterisk
peiyin wrote: Dear all, I want to create a php web front end to disconnect a SIP call (from a particular sip phone) which is in progress. Any ideas how to do so? Google for Flash Operator Panel. Or look in the Asterisk wiki for it. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Mahler's Book - New Project
Noah Miller wrote: snip In addition to largely being a rehash of existing docs on the internet, there are many editorial errors in the version that I have. Before I was comfortable with the conf files, these editorial errors were very confusing. The editions coming out now may have fixed these, but if not, it's just another reason to avoid thee book. I'd agree that the best way to get started is to get your hands wet. Be prepared to devote some time to learning asterisk. You'll find that in the end, it is still the quickest way, and well worth your effort. Someone told me of an O'Reilly book on Asterisk, and looking in their catalog i've found it: http://www.oreilly.com/catalog/asterisk/index.html Authors are credited as Jared Smith, Jim Van Meggelen and Leif Madsen, and it's due out in September '05. Has a picture of a starfish on the cover. Since it's not yet out, has anyone here proofread the thing, or has had an early copy, and willing to comment? Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connect 30 phone lines to asterisk how to
Bryce Chidester wrote: Assuming you mean you have 30 analog POTS lines, the way to go about this would be with a couple channel banks and a quad-T1 (I haven't seen a two-port around, but that's all that is needed) card. For the record, 30 individual analog lines is generally inefficient and would be done more cleanly with an E1 or 2 T1s. Some other possibilities: 1) get an E1 channel bank for the 30 analog lines, and a single T1/E1 card from digium 2) get two T1 channel banks for the 30 lines (split half-half I suppose) and a 2-port Sangoma E1/T1 card Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk LAMP Developer
Dominique Kull wrote: _Description_ We are looking for an expert LAMP (Linux, Apache, MySQL, Perl, and PHP) developer snip I will coin a new phrase for this list: LAMPEPA developer - a developer of solutions based on Linux, Apache, MySQL, Perl/PHP and Asterisk haha ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP or Asterisk
Dave Morrow wrote: Hi all. I have been using Asterisk for sometime now and have recently come across AMP for the first time. I am wondering if someone could enlighten me a little as to the advantages and disadvantages to using AMP as opposed to the do-it-yourself Asterisk? Is this documented someplace? Any advise would be greatly appreciated. AMP and others of the same ilk: Pros - web-based - some apps provide a graphical approach, e.g. Flash Operator Panel - abstracts most of the complexities into point-and-click interfaces - easier to manage, up to a certain point Cons - you're forced to work within the constraints of the application - can't simply modify the config files - some restrictions when you want to do complicated stuff with the dialplan - complex constructs hand-rolled into the dialplan may screw up the app - certain features not included - some apps don't work with Zap interfaces, for example - harder to integrate custom modules Do-it-yourself approach: Pros - highest level of flexibility in implementing your * box - add your own custom modules - complex dialplan constructs - won't have to run more services than necessary (no need for Apache etc) - forces you to learn more about the internals Cons - lots of editing text files - prone to typos - dialplan programming can be tedious - forces you to learn more about the internals My 2 cents. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP - Squished Faxes
Richard Cook wrote: Hello, Has anyone had issues with faxes showing up squished in the TIFF file? Any ideas what could be causing it? We had some issues while getting fax-email and email-fax working. As far as I can tell, it ended up being a wonky version of libtiff that was causing it. On our box we've got libtiff 3.6.1. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceXML? question
dave cantera wrote: hi, is there anything going with VoiceXML in asterisk??? is this the list to query regarding this or should I put this on the dev list? thanks, dave cantera I don't think there's anything built-in to support VoiceXML, but you _can_ do something like this: 1. get a developer account on Voxeo (http://community.voxeo.com/account/register.jsp) or some other VoiceXML provider 2. create your VXML app, and point to it appropriately on the developer account pages 3. connect via SIP from Asterisk to your VoiceXML app. 4. Fini Voxeo provides facilities to call in via Free world dialup, and your hosted applications can be accessed via a FWD number. I've got a simple demo running on our pbx and it works. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Karl
Libel Lawyer wrote: This is the guy that has a ton of email addresses. Almost as many as he has phone numbers. google kvj He doesn't like our president either: Here's look at a MISERABLE FAILURE and I use facts: garbage snipped Er.. did you type in the wrong email address in the To: field? flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENTt: GPL Asterisk Billing Software
Darren Wiebe wrote: Good Day, I'm finally getting around to officially announcing ASTPP. For the last 6 months, I've been working on converting ASTCC into a decent billing package for asterisk. The link in the original email opens a page that says Download the latest version of the code from http://www.aleph-com.net/astpp.html Has anyone else been able to download this code? I can't seem to find a link on their site to the code itself, and the astpp.html page brings up a Not Found... Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for people to test calls
Anton Krall wrote: Why disregard from MX? :) You might want to check the archives, or Google for Vonage staff arrested in Mexico, or something along those lines.. flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] play gsm files in windows
Brett, Gary wrote: Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated You could try Audacity (http://audacity.sourceforge.net). You have to use the Import Raw Data feature and open it as a GSM file. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programs to parse queue_log
Johann wrote: What third party programs are available for parsing the queue_log file and CDR file? I know about XC-AST, but management would prefer a php based solution. What have other admins done to retrieve detailed call information about the queue system? Anyone develop their own that they don't mind sharing? --johann Johann, What we do as our first step is to have a cron job run every night and export the queue_log file into an SQL table. From that point onwards it's relatively simple to come up with your own reports. You will have to read up the queuelog.txt file in the doc directory of the source code, it gives a lot of information about what each row in the queue_log file means. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone ever implement an *outbound* dial-by-name??
Michael Jones wrote: Hi All; I'm a newbie so please be gentle. I'm a new * user and am using it to control the 3 IP phones in my house. I'm using asterisk because I enjoy the flexibility and I'm sort of a tinkerer. Here's my question: Everyone has used the dial by directory function where you dial the user's name to connect to that extension. Instead of an inward dial, I'm thinking how cool it'd be to have an outward dial-by-name, where from any extension you can spell a name and dial it outbound via a trunk line. Off the top of my head.. specify a context in voicemail.conf: [outward-dial-by-name] 1000 = 1000,John Smith 1001 = 1000,George Lucas then another context in extensions.conf [outward-dial-by-name] 1000 = Zap/g1/5551234567 ; john smith's phone number 1001 = Zap/g1/555123 ; george's mobile phone and finally in your dialplan (assuming you use a context called internal for all your internal phones..) [internal] ; some other stuff exten = 123,1,Directory(outward-dial-by-name|outward-dial-by-name) disclaimer: untested stuff. your mileage may vary. don't sue me if it don't work Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting variable for a context for all extensions?
Mark Wormgoor wrote: Hi, Is it possible to set a variable for a context for all extensions? I haven't been able to find it. Try looking up the application SetVar: demo*CLI show application SetVar demo*CLI -= Info about application 'SetVar' =- [Synopsis]: Set variable to value [Description]: Setvar(#n=value): Sets channel specific variable n to value Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Asterisk do well in this application?
snip This is the sort of thing that AGI is great for. When I was first starting with Asterisk I wrote an AGI script to ask the caller for their Zip Code, then connect to weather.com, download the current weather conditions for that zip code, massage the text, run it thru a text to speech system, then play the resulting audio file to the user. You could also take the concept and connect it to a third-party VoiceXML provider like Nuance. You can host your VoiceXML script(s) in their sandbox environment and send a couple of calls over via SIP through FWD. Then you could make the Asterisk box look like a sophisticated speechrec-enabled system with a reasonably nice-sounding TTS engine to boot :) Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Asterisk do well in this application?
El Flynn wrote: snip This is the sort of thing that AGI is great for. When I was first starting with Asterisk I wrote an AGI script to ask the caller for their Zip Code, then connect to weather.com, download the current weather conditions for that zip code, massage the text, run it thru a text to speech system, then play the resulting audio file to the user. You could also take the concept and connect it to a third-party VoiceXML provider like Nuance. You can host your VoiceXML script(s) in their sandbox environment and send a couple of calls over via SIP through FWD. Then you could make the Asterisk box look like a sophisticated speechrec-enabled system with a reasonably nice-sounding TTS engine to boot :) Ahaha.. ahem.. i think i mean Voxeo instead of Nuance.. sorry, been a while since I did the vxml stuff.. flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Calls with Asterisk?
Jim Lists wrote: snip I'm still left wondering if Asterisk supports multiple lines at once? If I had one land line, voip line, and asterisk setup and 10 people called my number, would all 10 people be able to speak to their appropriate party at the same time, or would the other 9 get a busy signal? If all 10 called your land line at the same time, only one will get through and the rest gets the busy signal. There will be variations to this, depending on if you've got value added stuff like call waiting etc on the landline. Also, could this work with outgoing calls? If this DOES work, can anyone explain how it works? Would what work? Ten outgoing calls at once? If that's what you mean, then the answer would be not likely :) More importantly, at what point does the analog line move to the VOIP line, and how is the connection maintained? It would depend on how you've got the stuff set up. You didn't mention if the VoIP line was internal or incoming from a VoIP provider. Is your question really How does the incoming VoIP call get through to my (analog) extension or How does the internal VoIP call go out through my landline? In both cases, you should check out the Wiki at http://www.voip-info.org/wiki-Asterisk for more information on how it works. Cheers, Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk dialplanner
Hello all, For those of you who've attempted to use the Dialplanner, but could not receive the exported dialplan, we sincerely apologize for the problem. There was an internal misconfiguration on our mail server which stopped the dialplan from being emailed. We've since corrected the problem and you should be able to receive the dialplan now. Regards, Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weird call transfer problem
Anton Krall wrote: Guys. I just had a weird problem. I have my Dial cmd configured with mwtWT as parameters however, a call came in thru a zap channel and I answered on a sip phone. I tried using # as configured on my features.conf file to transfer the call but the transfer prompt never came in, so I asked the person on the zap channel to do the same and voila, he did get the transfer prompt and entered and extension, but what happended is that I was the one that got transfered! Not him! So. Any ideas whats wrong? The sip phone is an ata, a handytone 286 and zaptel cards. Why cant I do the # transfer and they can but Im the one been transfered? The T option allows the *calling* user to transfer the call, which is what happened to you. The t option allows the call recipient to transfer the caller to another extension. So to stop that from happening, remove the T option from the dial command. As to why you yourself can't transfer it might have something to do with the ATA itself, check what dtmfmode is specified in sip.conf. From the sample sip.conf, it says: dtmfmode=info ; either RFC2833 or INFO for the BudgeTone flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] invalid extension (need help)
amna saleem wrote: hi! I was wondering if the i extension works ,i mean i have included this in my extensions.conf ie exten = i,1,Answer exten = i,2,Playback(pbx-invalid) exten = i,3,Hangup but it doesn`t seem to work,i am getting no announcement when i dial an invalid no. rather i get the invalid tone (which we usually get on our analog phones at home) can someone help??? if you've already answered the call earlier, you don't need to Answer it again in the invalid context. flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX calls between asterisk boxes works 1 way only
James Bean wrote:
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a
On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found
On Box 2
Apr 11 17:26:07 NOTICE[2157]: chan_iax2.c:6573 socket_read: Rejected
connect attempt from 192.168.254.100, who was trying to reach '690@'
Error, so I obviously missed something and can someone smack me upside
the head and point out my error.
snip
Just had this happen a couple of minutes ago on our test boxes. You need to
double-check that the Box2's username/password, as specified on Box 1, is
entered properly in Box2's diaplan when dialing to Box 1.
e.g.
Box1 iax.conf
=
[box2]
username=box2
secret=box2secret
Box2 dialplan
=
exten = 777,1,Dial(IAX2/box2:[EMAIL PROTECTED]/${EXTEN})
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Re: [Asterisk-Users] Authentication with DB Support
kritikus Araklidas wrote: Hi: Somebody know how to configure the Authentication cmd with DB (Mysql) suport. its work with single password and password file, but i cannot find information for use database in conjunction with DB. Any help will be appreciated. Unless I'm mistaken (haven't been keeping up to date with the CVS stuff) you'll probably have to roll your own. Perhaps some simple AGI script that reads the password from the user and does a DB lookup. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Enhanced Queue App Revisited
Matt Roth wrote: Preferably, I would like an out-of-the-box solution, but custom-coding is an option as long as the necessary data is available from Asterisk. If anyone could point me in the right direction, it would be greatly appreciated. You're right in that most of the things you're asking for requires custom programming. All the data should already be there to get the data you want. It's just a matter of coming up with the application to generate said reports. You might want to check out XC-AST, do a search on the list for that. It's a web-based application that should report most (if not all) of what you're looking for. Otherwise you'll have to roll your own. We've got our own queue reporting app, doing stuff like what you mentioned. It shouldn't be too difficult, as long as you've got some means to extract the data from the queue log files, as well as the CDR. Of paramount importance is understanding what means what in the queue log file. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over
Rich Adamson wrote: No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. Dataprobe makes a range of A/B switches, some with more intelligence that you might be able to use in this scenario. One of their products (check out http://www.dataprobe.com/switch/ab_net.html) has a feature which pings a specific IP address, and switches over once it stops getting a response. Some of their products are programmable too, where you can send TCP messages to initiate the switching process. Check out their website for more products. Flynn p/s I am in no way related to Dataprobe. This is just some stuff I received from them when asking a similar question on the list about six months ago. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background apps that plays music on hold
Kong wrote: Is there any application that actually work like Background, but instead of playing a specified file, it plays the streaming music from music on hold? the reason i am asking this because i come across a dialplan that goes this way, if a person gets to an extension that is busy, it will playback a message like. the person is current busy, press 1 to leave a message or hold on the line so when this message is finish, it starts to play music on hold. if the user wants to leave a msg now, it can press 1 to do so, else stay on the line still the busy user is done with the call and connects to it. 1. go to the cli 2. type show application waitmusiconhold flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream and Transfers
Hi all, Just wondering if anyone's come across this issue, and what might be a fix for it: We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The phone can do proper supervised transfer, but _only_ once. If the user attempts to transfer a second time, it won't work. any suggestions/hints/tips are welcome.. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX softphone on WinCE/PocketPC
Hi, Is anyone aware of an IAX client that's made for the Windows CE/Pocket PC platform? Or even the Palm platform for that matter. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Apple links Asterisk
Matthew Boehm wrote: From macintouch.com: Apple is distributing an open-source Asterisk install package for Mac OS X: A complete IP-PBX in software. SNIP If anyone's interested, Benjamin Kowarsch from Sunrise Telephone systems Ltd is doing that. Check it out at http://www.sunrise-tel.com You can also google the mailing list for his email, if interested. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Music Volume ?
Mateo Meier wrote: What do you mean ? My etc/asterisk/musiconhold.conf looks like that: [EMAIL PROTECTED] root]# more /etc/asterisk/musiconhold.conf ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3 ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered = mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf = quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces) ;manual = custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s what he means is for you to change the line ;loud = mp3:/var/lib/asterisk/mohmp3 uncomment it, and in your extensions.conf file: exten = exten,priority,SetMusicOnHold(loud) Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vonage, broadvoice et al
Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same account, at the same time? Or do these providers typically only allow one call to be in place at any point in time? Thanks in advance. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue-timeout- press button to remain on hold
Jay Wilton wrote: Hello, Is it possible to use a timeout in a queue and have the option of pressing a button to remain on hold? I have been using: [qbert] 1,1,Queue(qsales|t|||180) 1,2,Voicemail(u22) [qout-sales] ;dtmf-out context from queues.conf /[qbert] *,1,goto(qbert|1|1) Problem - I return to the back of the Queue. thanks - CVS-HEAD-01/31/05-03 JJ what about doing the reverse: specifying a longer timeout period, with the option of pressing a button to go into the voicemail? that way the caller still mantains his/her position in the queue... flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM series + kernel 2.6
[EMAIL PROTECTED] wrote: Hello, I have looking into the TDM series of wildcards. All these card are for linux kernel 2.4. If I were to use FC3 which is based on kernel 2.6, will I have any compatibility issues. Thanks I'm not sure about Fedora, but we're running SuSE 9.1 with the 2.6 kernel and have not had any problems with the TDM cards. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI only 2 calls
Altus Snyman wrote: Good day all I downloaded bristuff RFC3 and asterisk,zaptel,libpri versions 1.0.3 This is to install my quad bri card All installed well I coped over some old config files.All 4 ports are available,so that gives 8 open lines for incoming or outgoing,correct me of I'm rond The problem is,asterisk can only handle 2 calls at a time if there is 1 incoming(into pstn) and there someone already made a call out of the pstn,you cant make any other calls out or in On the cli it just show,when you try dialing out,Zap/4-1 got Hangup Even when you change the channels in zapata.conf,it keeps on showing,trying to make call Zap/10-1/012020121.Zap/10-1 got hangup? All the zttool and zttest shows its up and working Can this be a Telecoms provider problem please advice Thanks altus erm.. do you have all the lines plugged in to the card? I'm assuming you've got four BRI lines.. it sounds like only one is plugged in, hence only allowing you to have 2 simultaneous calls... flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Group Extension
Edgar de Leon wrote: Hello, i got a question, i need to create a group extension, to make calls to 6 sw phones, but i need to know if asterisk can do help me to get a unique number and check what extension has received less calls than the others, and pass the new call. We got a call center and want to know if we can distribute the calls depending in what extension is available and from the extensions that are available pass the call to the operator that has answered less calls, can i do this with *? can i get statistics from the use for an extension? can anybody help me?? it sounds like you're wanting to use asterisk's call queueing capabilities. look at http://www.voip-info.org/wiki-Asterisk+call+queues for more info. Especially look at the Strategies section on that page, which has a fewestcalls strategy, which basically rings the extension which has taken the fewest calls to date. flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor calls timeout
jurgen wrote: snip Problem is, Asterisk times out and disconnects after 10 seconds, stopping the recording. If I run something else in the context, say the infamous Monkey Sounds, everything's fine, and the call just keeps going, annoying the people on the line with monkey sounds. For some reason, the *monitoring* always stops after 10 seconds. did you try setting using AbsoluteTimeout in the context? e.g. exten = s,1,Answer exten = s,2,AbsoluteTimeout(0) exten = s,3,Monitor(wav,testrecod,m) I also once had a problem where my TDM400P card thought the far end had disconnected even though the two parties were still talking to each other. It was happening after roughly a minute and 40 seconds into the call. Setting busydetect=no and callprogress=no in zapata.conf helped a bit, although I suspect it might actually had something to do with the phone line itself. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Processing incoming calls with multiple contextst over PRI
Jason Brown wrote: So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. snip Just a guess about your problems, but if you have a PRI line incoming, wouldn't you need to connect it to a PRI card and not the TDM400P (which is for analog POTS lines)?? Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to stop ringing after congestion.
Jon Gabrielson wrote:
When there are no zap channels available, I signal congestion
using the following:
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Playtones(congestion)
exten = _9NXX,3,Congestion
The congestion sound plays correctly, but the ringing continues
in the background. Why is it still ringing and how do I make it stop?
try
exten = _9NXX,3,Congestion(5)
which will stop the tones after 5 seconds.
flynn
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