s100u notes [was Re: [Asterisk-Users] S100U hissing noise..]
>>> [EMAIL PROTECTED] wrote a 1.2KB message. i replied >>> . > > > > I have got a hissing/humming noise on my handset that is > > connected to the S100U USB device.. > > > The USB port is VERY sensitive to static electricity. It's not rated > production quality, only proof of concept. You should really use > the new TDM400P which supports one to four lines. Some notes about the s100u (which i've finally managed to somewhat stabilize and cooperate with an x100p in the same box). -- Static . People have reported static sensitivity problems with the device. I found it to be very unstable until i noticed that both the usb and pstn connections are very sensitive to motion. I wrapped a single strand of insulated wire a few times around the device, including both the usb and phone wires in the binding. I also mounted it on a water pipe in our basement (where the system is). This seems to prevent the connections from breaking and it also keeps the device out of the way. -- Integration. The entire process of loading the modules is highly dependent upon order of hardware detection and configuration. When attempting to reload the wcusb module make certain to unload your usb controller driver (usb-uhci,etc). before reloading .. Make sure you run ztcfg after reloading zaptel modules. This may be done automatically in some cases. Also, zttool is your friend. If you dont have it on your system you probably need to install newt. -- IRQs. Configure your system such that your ethernet/sound/zaptel/usb devices are not sharing interrupts. This is very important, these things generate loads of interrupts. $0.02, --Michael. -- Michael Jastremski | Network Engineer Megaglobal Networks | Megaglobal.net Open Photo Project | Openphoto.net West Philadelphia | Westphila.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI CallerID problem
On Wed, 2003-08-20 at 18:31, Anthony Wood wrote: > Is this the scenario you are describing? > > You call the office from home, the direct number of a collegue. > > He is not in, asterisk forwards it to his mobile. > > His mobile rings and gives the PRI number as the caller ID, but you want it to give > out your home number? > Yes, that's correct. > > > > Yes, we can send CID info to our PRI provider. If we make a call with > > our Cisco 7960, we can send any phone number we enter into > > Even numbers not associated with your PRI? > > If you can, then your provider is very trusting, letting you spoof caller ID. > Call it what you will but it's intentional. It only allows me to change the Flex ANI, not the actual BTN. So it's still traceable. That's another discussion. > Perhaps your provider will only let you set caller ID to a number in your range of > numbers. > > Otherwise, perhaps the provider of the called parties line will only let you provider > send caller id from it's numbers, and your home number (in my example) is not > part of that. > Nope. I can send whatever digits I want. I can send 206-555-1212, 000-000- or 111-111- using SetCallerID(). I just can't seem to get Asterisk to forward the CallerID info from an inbound call (my home phone in your example) to another caller (a cell phone in your example) through Asterisk. I think it's a bug, but I may be forgetting a setting. It's not only happening on our PRI but on another customer T1 as well. Thanks Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 core dump on reload, works fine at startup
I'm running the CVS from last week and from day one (over 4 months now) I've had this problem where asterisk core dumps when using chan_h323. It appears to be a problem with pwlib and the console, but I'm not sure how to read the below output from gdb. I can start Asterisk just fine and chan_h323 works great when sending and receiving calls. I only have this core dump problem when sending a reload to Asterisk via the CLI or "asterisk -rx "reload"". Environment paths: LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 Core dump info: (gdb) bt #0 0x4831e20c in PSocket::os_select(int, fd_set*, fd_set*, fd_set*, PIntArray const&, PTimeInterval const&) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #1 0x48315a2a in PSocket::Select(PSocket::SelectList&, PSocket::SelectList&, PSocket::SelectList&, PTimeInterval const&) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #2 0x483151a7 in PSocket::Select(PSocket::SelectList&, PTimeInterval const&) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x48aec440 in H323TransportUDP::DiscoverGatekeeper(H323Gatekeeper&, H323RasPDU&, H323TransportAddress const&) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x48b023b9 in H323Gatekeeper::StartDiscovery(H323TransportAddress const&) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x48b020e3 in H323Gatekeeper::DiscoverByAddress(H323TransportAddress const&) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x48aa10ae in H323EndPoint::SetGatekeeper(PString const&, H323Transport*) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x41ef5d11 in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x41efe680 "65.39.220.195", secret=0x41efe700 "") at ast_h323.cpp:949 #8 0x41eeed81 in reload () at chan_h323.c:1595 #9 0x08055362 in ast_module_reload () at loader.c:159 #10 0x0806b63a in handle_reload (fd=10, argc=1, argv=0x4effe6dc) at cli.c:105 #11 0x0806b42a in ast_cli_command (fd=10, s=0x0) at cli.c:1006 #12 0x08080aa0 in netconsole (vconsole=0x80b6348) at asterisk.c:192 #13 0x4002f2b6 in start_thread () from /lib/tls/libpthread.so.0 Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 core dump on reload, works fine atstartup
On Mon, 2003-09-01 at 11:19, Brian West wrote: > Are you using the recommended pwlib and openh323 versions? > Yes. lrwxrwxrwx1 root root 12 Aug 17 20:39 pwlib -> pwlib-1.4.11 lrwxrwxrwx1 root root 15 Aug 17 20:01 openh323 -> openh323-1.11.7 Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming call on LineJack's LINE/FXO is not answered by *
[output END B] Is the ring cadence values from "rcvd ring" shown above normal? I am not sure if it's the cadence from the line that is causing Asterisk not be signaled that there is an incoming call (thinking about going into the source code to check how this is done) If I change the settings in "phone.conf" to "mode=immediate" or "mode=dialtone", I can hear the demo-congrat voice from the Asterisk system and navigate the demo menu without a problem. So...I am stuck on this problem for the moment Anyone has any insight or suggestion on this?? Did I miss a configuration somewhere? Or misconfigured something? Help me please! :) By the way, I am using RedHat 7.2 with kernel 2.4.19. Thanks! Michael W. p.s. module ls output: root> ls -l/lib/modules/2.4.19/kernel/drivers/telephony/ total 172 -rwxr-xr-x1 root root 159956 Dec 26 13:33 ixj.o -rwxr-xr-x1 root root 4864 Dec 26 13:33 phonedev.o p.s. with debug and verbose on, and with mode=immediate, the console outputs what Asterisk is doinghowever, with mode=fxo, the console prompt just sits there...nothing is outputed... - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming call on LineJack's LINE/FXO is not answered by *
Yes... I did restart Asterisk after I changed the 'mode' in phone.conf. And after I restarted Asterisk, I always use 'cat /proc/ixj' to make sure that the line "Port" from the output reads "Port pstn" when 'mode=fxo' and reads "Port pots" when 'mode=immediate' or 'mode=dialtone'. I used "*CLI> stop now" to stop Asterisk, and then started it again with command line "asterisk -dc". (I modified logger.conf to have debug output to 'console',too.) I even tried unplugging the phone set from the Linejack card when I am using 'mode=fxo' and restarted Asterisk... But still, Asterisk doesn't pick up the incoming call and play the nice greeting message... not even showing some kind of debugging message...:( Any more ideas?? Thanks, Michael W. >From Tilghman Lesher <[EMAIL PROTECTED]>: > On Friday 26 December 2003 08:05, Michael wrote: > > Hello All... > > > > I have searched in the archive and also followed Zara's instruction > > on getting incoming calls to work with Asterisk...but I still can't > > get Asterisk to answer incoming call on Linejack's LINE port. > > > > I attached a phone set to the PHONE port, and telco line to the > > LINE port on the Linejack(ISA) card. > > The drivers for the Linejack do not support using the card as an FXO > device and an FXS device at the same time. You must choose one > functionality for the card and use it that way. Changing the > functionality requires a restart of Asterisk. > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P driver modprobe failed
Hello Everyone, I just got my Dev Kit TDM today... :D I installed the X100P ok (wcfxo); however, when I tried to 'modprobe wcfxs' for the TDM400P(TDM10B), I got this error message: /lib/modules/2.4.19/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters /lib/modules/2.4.19/misc/wcfxs.o: insmod /lib/modules/2.4.19/misc/wcfxs.o failed /lib/modules/2.4.19/misc/wcfxs.o: insmod wcfxs failed I checked the TDM10B card installation; I checked the 12V power connecting to it; I moved the card to a different PCI slot...but all of that didn't help the driver installation :( Help? Any suggestion would be appreciated! Thanks, Michael W. p.s. I compiled the driver from zaptel-0.7.0.tar.gz. - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Earpiece Connections
Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of hardware that do this (41D switch matrix?) But I need to find one that asterisk can use. I will then build some custom scripts to handle the "Pickup" and "Hangup" parts of it. Anyway any ideas or websites I could research for this type of thing would be most helpful. Thanks Micahel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID Trunk Lines and Caller ID
Title: Message I have an installation which is currenly using 14 DID Trunk Lines. I need to be able to use Caller ID information and currently it is not available on these lines. Is there another way to access this information? Thanks
[Asterisk-Users] Re: Earpiece Connections
Andrew, I need a keypad-less phone to listen and talk. I would like to have up to 10 of these and I would like to have each of them have their own extension so that calls can be routed directly to them. I would then be able to trigger the Pickup and Hangup functions on the server itself. Have you heard about a resource for these types of devices? Thanks Michael From: "Andrew Thompson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Earpiece Connections Date: Mon, 5 Jan 2004 09:33:57 -0500 Reply-To: [EMAIL PROTECTED] - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, January 04, 2004 7:31 PM Subject: [Asterisk-Users] Earpiece Connections > Does anyone know of a piece of hardware that can allow multiple > earpices to be connected directly to a server running Asterisk. Just to listen? Or, do you mean a keypad-less phone(listen and talk)? > I hope I am not being to vague but basically I am looking to allow a > call center to user the server to do all of the "Pickup" and "Hangup" > functions. The operators will merely have to have th earpiece in their > ear. I have seen serial pieces of hardware that do this (41D switch > matrix?) But I need to find one that asterisk can use. I will then > build some custom scripts to handle the "Pickup" and "Hangup" parts of > it. > > Anyway any ideas or websites I could research for this type of thing > would be most helpful. > > Thanks > > Micahel I don't think you're going to be able to get any simpler than AgentLogin, where the user dials into a queue, and stays on the line receiving calls as they come in. You'll still need some kind of phone device, like a channel bank to split calls to regular handsets, or some IP phones. Unless I really don't understand what you're trying to do? - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange Call waiting problems - SNOM 200 & Grandstream Budgetone
Hi I am setting up an Asterisk System in an office environment, Incoming and Outgoing calls are working ok, but i am having a few strange problems regarding call waiting. With the SNOM 200 (firmware 2.02t) phones, if you are on a call and a 2nd call comes in, the call waiting beep is played and the light flashes, but if you hang up the 1st call, instead of the phone ringing, it connects the 1st call and the 2nd call together! The 2nd caller ends up speaking to the first caller (almost like a conference call, but you cannot hear or speak to the 1st or 2nd callers) as you can imagine, this could cause a few serious problems :-( . we have not been able to recreate this problem with the grandstream, but we are having another strange problem with the Budgetone. if you are on a call on the Budgetone 101 and a 2nd call is received, instead of a call waiting beep being played, it rings on the handset speaker! which makes it almost impossible to speak to the 1st caller, but if you hang up the 1st call, the phone rings and it is possible to answer the 2nd call normally. Anyone got any ideas on what could be causing these problems ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Kedpad less extension
Does anyone know of a resource for extensions in which the server (whether asterisk or custom scripts) can trigger the phone to be answered? So for example an operator can have a headset and when a call comes through the call is automatically (through a script) connected to the headset instead of the operator having to manually answer the call. Any responses, help or ideas of a type of supplier to contact for more information would be greatly appreciated. Thanks Michael Blood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Kedpad less extension
I was looking for a hardphone since it made sense to me that it would be better quality. But I like the scalability of this option since it is an IP Phone. Also may there is another option here. Is there any way that an server could have multiple instances of an iax type client and then have multiple headsets attached to each? Thanks again. Michael Blood -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Van Donselaar Sent: Thursday, January 08, 2004 4:01 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Kedpad less extension On Thu, 8 Jan 2004 13:58:45 -0700, <[EMAIL PROTECTED]> wrote: >Does anyone know of a resource for extensions in which the server >(whether asterisk or custom scripts) can trigger the phone to be >answered? So for example an operator can have a headset and when a call >comes through the call is automatically (through a script) connected to >the headset instead of the operator having to manually answer the call. > >Any responses, help or ideas of a type of supplier to contact for more >information would be greatly appreciated. I don't know if you are wanting a hard phone or a softphone. If a softphone is acceptable, iaxComm can do this on a Windows or Linux PC with the intercom feature. > >Thanks > >Michael Blood > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX clients
Has there been any additional work on this yet? I haven't found anything in my searching for an iax client in java and I would really like to find one. I understand that java may be less efficient but I am hoping to find a portable solution which can be used on both a windows workstation or linux workstation Michael Blood From: Alastair Maw Subject: Re: [Asterisk-Users] IAX clients Date: Mon, 08 Dec 2003 07:08:58 -0800 On 08/12/03 13:29, Rattana BIV wrote: >Is there IAX client in Applet JAVA which can be embeded in a web page ? Nope. But I'm working on a Java IAX2 library that would let you easily build one. It'll be a little while yet, though. :) Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Postgres And Custom Scripts in a Turnkey Solution
A while back (November) I sent out a request to find providers who sell Turnkey systems. Some replied explaining that they have systems which are turn key AND are based on the postgres database for simple configuration and that they have built Turnkey systems. I am also looking for some customization of scripts in order to assist in the correct routing of incoming calls to custom extentions. If you can supply systems that meet these criteria please or even better if you previously spoke with me please email me back since I have lost your contact information. Thanks Michael Blood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX -> RT Integration
Greetings, I had been working on Asterisk (http://asteriskpbx.org) about 2 years ago . http://www.marko.net/asterisk/archives/0210/0107.html Last night with the help of Jesse's rt-soap-server.pl (and some prodding) I implemented a much cleaner, more repeatable * -> RT phone gateway with some notes: http://megaglobal.net/docs/asterisk/html/rtasterisk.html Questions or comments appreciated. --Michael. -- . Michael Jastremski .. .. Network Engineer > Megaglobal Networks > Megaglobal.net .. Photographer > Open Photo Project > Openphoto.net . .. Resident > West Philadelphia > Westphila.net . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallingCard Applications
I want to build my own calling card system on Asterisk. I looked at this page - http://www.voipinfo.org/wiki/view/CallingCard+Applications and it has listed some applications that I thought could help speed up the development process though the link down the bottom doesn't work. Does anyone know of any AGI etc applications to build a Calling Card system on Asterisk? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial command
When I call an extension on my Asterisk system, and the extension is unplugged, I just get silence for the 30 seconds (Dial command ring time) before it goes to voice mail. How can I get around this? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk spoken digits
How do I customize the digits 0 to 9? I have tried changing the paths in say.conf and nothing changes. I would like to do this without over writing the existing files, so I can have all my custom files in one location. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ignoring context= in sip.conf
I put context = xyz in the sip.conf upline supplier configuration and it ignores this and seems to place it in to default, as the incoming call rule in extensions.conf only works when placed in [default] ruleset. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ignoring context= in sip.conf
I put context = xyz in the sip.conf upline supplier configuration and it ignores this and seems to place it in to default, as the incoming call rule in extensions.conf only works when placed in [default] ruleset. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Country numbering plan resources
Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. I was also thinking maybe people present could contribute some information on this list for now. The countries I am after are below. To start this off I will provide the information for Australia +61 and New Zealand +64. NZ Cellular: area code 21 and 29 followed by 6, 7 or 8 digits - Vodafone GSM area code 27 followed by 6 or 7 digits - NZ Telecom CDMA note that there is number portability so the above is a guide. NZ Landline: area code 3, 4, 6, 7 and 9 followed by 7 digits (first digit will be in the range of 2-9) NZ toll free: area code 508 and 800 followed by 6 digits NZ premium: area code 900 - though I doubt any of you will be routing these calls AU cellular: area code 4 followed by a 2 digit network code, and then a 6 digit number Networks include: Optus, Telstra, 3, Vodafone, Virgin and others. All use GSM and there is number portability. AU landline: area code 2, 3, 7 and 8 followed by 8 digits (first digit will be in the range of 2-9) AU toll free: area code 1300 or 1800 followed by 6 digits OR area code 13 followed by 4 digits. AU premium: I'm not sure though someone present may fill us in. Following is the list of countries I need information on: ; ANDORRA ; ARGENTINA ; AUSTRIA ; BAHAMAS ; BELGIUM ; BRAZIL ; BULGARIA ; CANADA ; CHILE ; CHINA ; COLOMBIA ; CROATIA ; CYPRUS SOUTH ; CZECH REPUBLIC ; DENMARK ; ESTONIA ; FRANCE ; GERMANY ; GREECE ; GUADELOUPE ; GUAM ; HONG KONG ; HUNGARY ; ICELAND ; INDONESIA ; IRELAND ; ISRAEL ; ITALY ; JAPAN ; JORDAN ; SOUTH KOREA ; LUXEMBOURG ; MALAYSIA ; MARIANA ISLANDS ; MEXICO ; MONACO ; NETHERLANDS ; NORWAY ; PANAMA ; PERU ; PERU LIMA ; POLAND ; PORTUGAL ; PUERTO RICO ; ROMANIA ; RUSSIA ; SAN MARINO ; SINGAPORE ; SLOVAKIA ; SLOVENIA ; SPAIN ; SWEDEN ; SWITZERLAND ; TAIWAN ; THAILAND ; TURKEY ; UNITED KINGDOM ; UNITED STATES ; VENEZUELA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, 13 Dec 2008 16:24:56 you wrote: > One of the problems you'll run into is that in larger countries like the > US, and/or countries with greater amounts of telecom interconnection, > competition and deregulation, this information cannot be reduced simply > to a convenient algorithm. > > The North American Numbering Plan (www.nanpa.com) does provide some > basic standards for valid numbers, but aside from that, there exists no > special numerological distinction between incumbent and competitive, > fixed-line and mobile, or VoIP, and extensive number portability throws > even more complexity into the mix. > > I'm not saying it can't be done - just be aware that the undertaking > you're proposing is very complicated, and the information would come > from innumerable data sources (a great deal of them commercial and > expensive) and a bewilderingly overlapping array of standards bodies. Yes, but calls to the USA and Canada landline/cellular cost the same. I need as many countries in the list that I can get info on because in many cases cellular calls and landline calls are priced differently and I need to make routing distinctions in my dial plan. Yes you are correct, Australia and New Zealand are an easy plan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
> In general you don't need to worry about that, as when you go to buy your > "routes", the splits are given to you. For example, though you have split > up New Zealand nicely I don't need that information, as the termination > provider I buy New Zealand from gives me one price for what they deem > "proper" (01164) and another several for what they deem "mobile" > (01164900, 011648, 011642). Whatever destination is dialed simply picks > the route that it most matches, and I know what the charges are. Case in point (and why we should have a community orientated approach to this) If that is how your carrier has divided it up they have given you inaccurate information. Let's forget about USA/Canada for now as from my/most people's point of view the routes are all so cheap (and blended) that it does not matter. I think it is more important to focus on other countries. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
> Only when it's simple. When a country is small or is big but has a > single state telco incumbent and a few mobile carriers, that's not too > hard. > > Of course you can get blended domestic US48 termination - most people > do. But, two things happen when you hit a large traffic volume that > cause that to go away: > > 1) *You* lose on the blended rate for high-density and/or highly > competitive destinations you could route to for rates under the blended > plan. > > 2) Your providers get increasingly nervous about their exposure to your > all-over-the-map traffic patterns, ability to adhere to a theoretical > 80% RBOC blend, or whatever. So, high amounts of traffic start to get > broken apart into much, much more granular (and therefore numerous) > tiers, sometimes down to the terminating carrier. Well, hopefully some people outside of the USA/Canada will assist me with other destinations on the list. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
> Hmm, I looked over your summary again against the route prefixes I just > gave and they seem to match. They aren't as detailed, but that isn't > important, as long as I can tell a cellular from a landline, which those > prefixes do accomplish. I don't really care how accurate they are either, > as long as my carrier will honor the prices for the prefixes they have > provided me. Great! I will send you some 900 calls lol :-) > > Let's forget about USA/Canada for now as from my/most people's point of > > view the routes are all so cheap (and blended) that it does not matter. I > > think it is more important to focus on other countries. > > You have no idea what an uphill battle you will be fighting, and one that > is constantly changing. If the idea is to compile all this info to make a > master routing list for making purchases, you really don't need to bother. > They will be given to you buy your carriers. NANPA is complex, but for > purchasing at the wholesale level blended routes are pretty common, which > actually makes it one of the simpler ones. Try the Dominican Republic - I > currently have over 1200 routes to this small country, and they cannot be > any further collapsed... Yes, but with an A-Z carrier, this can become risky when landline calls are charged very differently to cellular calls, as is the case in NZ, Australia and many other countries, unless someone is just a 'virtual' provider and letting their up line do the invoices. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sat, 13 Dec 2008 16:45:11 Alex Balashov wrote: > Michael wrote: > > Let's forget about USA/Canada for now as from my/most people's point of > > view the routes are all so cheap (and blended) that it does not matter. I > > think it is more important to focus on other countries. > > What is your traffic volume such that you are claiming to speak for most > people? > > For genuinely large traffic volumes, it most emphatically _does_ matter. Easy - you are based in the USA, so very likely most of your traffic volume will be in this general area. Where I am based, while there is a lot of traffic volume to North America, there are also large volumes to Pacific and Asia. So therefore the over all USA and NA % is smaller from this part of the world, hence the up line can make enough profit over all that they are less likely to view it as a loosing proposition. There is of course also the psychological element that companies are more polite to people overseas... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk / Hylafax
On the subject of faxing - is there a way to interface Asterisk and Hylafax running on the same box? I have tried the directions at- http://www.voipinfo.org/wiki/view/T38modem+configuration+with+Asterisk and I just can't get it to work. Does anyone know of a way? (Even mention paid software)... Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / Hylafax
On Sat, 13 Dec 2008 21:52:25 you wrote: > Check out my tutorial at: > > http://blog.evaristesys.com/?p=24 > > It does use IAXmodem for conventional G.711u analog pass-thru (assuming > you're using VoIP) rather than t38modem, but if you need to use T.38 I'm > sure it can be easily adapted. 1. Do I need T.38 between Asterisk and Hylafax as I am using T.38 over SIP to the upline PSTN termination provider? 2. If this is workable, does using IAX to SIP cause extra CPU load / loss / issues? Totally off topic, but people may feel sorry for the residents of the Cook Islands. Their local ISP is selling 56 to 256k ADSL connections as 'broadband" and the data caps and low and prices huge. http://www.oyster.net.ck/about/index.php?about=idsl (NZ$ 1.00 = US$0.55) Thanks, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / Hylafax
I am working my way through it and this is the error message- [2008-12-13 22:27:00] Modem started [2008-12-13 22:27:00] Setting device = '/dev/ttyIAX0' [2008-12-13 22:27:00] Setting owner = 'uucp:uucp' [2008-12-13 22:27:00] Setting mode = '660' [2008-12-13 22:27:00] Setting port = 4570 [2008-12-13 22:27:00] Setting refresh = 60 [2008-12-13 22:27:00] Setting server = '127.0.0.1' [2008-12-13 22:27:00] Setting peername = 'USERNAME' [2008-12-13 22:27:00] Setting secret = 'PASSWORD' [2008-12-13 22:27:00] Setting cidname = 'Net Trust Fax' [2008-12-13 22:27:00] Setting cidnumber = '098391001' [2008-12-13 22:27:00] Setting codec = alaw [2008-12-13 22:27:00] Opened pty, slave device: /dev/pts/2 [2008-12-13 22:27:00] Created /dev/ttyIAX0 symbolic link [2008-12-13 22:27:00] Registration failed. My configuration in iax.conf is as follows: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw mailboxdetail=yes iaxcompat=yes jitterbuffer=yes [FaxDSP0] type=friend username=USERNAME secret=PASSWORD host=127.0.0.1 port=4570 disallow=all allow=alaw qualify=yes PS: Is 'peername' in the IAX config the same as the iax.conf 'username'? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / Hylafax
For some odd reason the call registration issue doesn't seem to stop it working, except a few seconds after Hylafax answers the call it hangs up, I suspect because Asterisk only supports T38 pass through. Here is my data path PSTN => T.38 aware SIP provider => Internet => My machine (Asterisk) => IAX modem on localhost => Hylafax. While I am aware from being on the Hylafax mailing list that IAX does work, no one there has been able to advise of a way it can be done with a SIP upline provider. I think people generally use it for ISDN PRI => Asterisk => IAX modem => Hylafax setups. Now Call Weaver apparently does have some better fax capabilities though I can't find any info on how to integrate it with Hylafax. Does anyone know how it can be done with Call Weaver and/or if there is any 3rd party software to enable what I am after? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 codec files
Which of these files is best for a Core2 Duo? I'm not sure whether to choose Pentium or i686. http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-32/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sun, 14 Dec 2008 19:57:20 you wrote: > On Dec 12, 2008, at 7:10 PM, Michael wrote: > > Is there any good free / accurate online resources with detailed > > country > > numbering plans? Failing that let's get something running ourselves. > > [snip] > > I'll avoid the good discussion on this thread that has been made > already to date. The answer is: "It's not simple." I wish it was, > but that data is typically in the hands of companies who do NOT want > to share it, because errors on your part make money on their part, and > those companies don't really want to see you exist at all in the first > place. It would be great if this could be a shared resource of some > sort, but I don't expect that we'll ever see that with E.164 numbering > - the entrenched interests in that number space have zero interest in > making the data available for many political, technical, and fiscal > reasons. [snip] Thanks John. You are clearly someone who has experience, and is well informed, though your contribution reflects those of other people from USA, whereas from where I (and most others) are from, USA is only 1 or 2 billing destinations out of 100+. So the validity of the idea/suggested project still stands. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
On Sun, 14 Dec 2008 20:14:22 Tzafrir Cohen wrote: > On Sat, Dec 13, 2008 at 10:57:20PM -0800, John Todd wrote: > > On Dec 12, 2008, at 7:10 PM, Michael wrote: > > > Is there any good free / accurate online resources with detailed > > > country > > > numbering plans? Failing that let's get something running ourselves. > > > > [snip] > > > > I'll avoid the good discussion on this thread that has been made > > already to date. The answer is: "It's not simple." > > Right. So for those of us who want to do simple things and avoid > complicated stuff such as telephony in shoddy continent of North > America, could you please provide data for your country? > > So far we have AU, IL and NZ. We have Uruguay as well. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / Hylafax
> This path will not work. As You mentioned, * supports T38 path through > only. In Your setup there will be a conversion on the * box between T38 > via SIP provider and IAX (which uses G711 codec in this case). > > To make it work, use newer versions of t38modem and replace the iaxmodem > with it. Newer versions of t38modem supports SIP, so that Your path will > be > PSTN => T.38 aware SIP provider => Internet => My machine (Asterisk) => > T38-modem on localhost => Hylafax. I am using t38 modem 1.0.0, which AFAIK is the latest. I have also tried the t38 method (as documented on voip-info using opal and pwlib) and have not been able to get this to work. As a side when I start the t38 modem it complains it can find libavcodec, which is on my machine. I don't know if this is material or not? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Variables for dial plan
I want to have a arbitary named variable within the client's user details in
sip.conf
[client1]
dialplan=NZ
..
In extensions.conf (Logic expressed using PHP style)
if ($dialplan == NZ) {
$NAT = 0;
$INT = 00;
};
and in the [outgoing] section
; Australia
exten => _${INT}61[278]NXX.,1,Set(CDR(UserField)=AUSTRALIA)
exten => _${INT}61[278]NXX.,n,Dial(SIP/SIP_PROVIDER/0${EXTEN:4:9})
How can I implement this in Asterisk style?
Thanks,
Michael
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Re: [asterisk-users] Variables for dial plan
On Mon, 15 Dec 2008 21:31:56 you wrote:
> Use setvar=variablename=value
>
> Eg: under [client1]
> setvar=dialplan=NZ
>
> Then just reference ${dialplan} in your extensions.conf
>
> Cheers
> Andy
Thanks, now how do I achieve the following logic?
if ($dialplan == NZ) {
$NAT = 0;
$INT = 00;
};
Michael
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[asterisk-users] UDPTL setup
This setting here- ; UDPTL start and UDPTL end configure start and end addresses ; udptlstart=4000 udptlend=4999 Does this need to be allowed on incoming or outgoing firewall rulesets of the machine running Asterisk? Thanks in anticipation, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Netcomm V90s + Asterisk + conference
This might be a curly one- I have a Netcomm V90s VoIP phone that has 4 line function - L1 to L4. It appears the only way to use the conference function is to set up a 2nd (and any subsequent) VoIP account. Has anyone found away around this that does not involve setting up multiple SIP accounts? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / Hylafax
> Recently i also posted some rough configuration sample of my setup on > http://lists.digium.com/pipermail/asterisk-users/2008-November/222531.html > > Please mind, that if you're trying T38modem, you should get versions > exactly as specified in voip-info.org, otherwise they might not work > with Opal (which adds SIP protocol, as T38modem was originally for > H.323) I used the SVN versions as recommended. Do you have any idea why T38 modem complains about libavcodec on start up? Thanks, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] user entry as variables
I want to take series of user entered (via phone keypad) options/numeric entry fields and use these with an AGI script. I have looked through voip-info and I can't find any Asterisk functions specifically for this. Any guidance please? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AGX addons compile issues
Has anyone seen this before, and know what is happening? u...@host:~/asterisk/agx-ast-addons# ./build.sh -- Configuring done -- Generating done -- Build files have been written to: /root/asterisk/agx-ast-addons [ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o Linking C shared module dist/app_devstate.so [ 11%] Built target app_devstate [ 22%] Building C object CMakeFiles/app_nv_backgrounddetect.dir/app_nv_backgrounddetect.o Linking C shared module dist/app_nv_backgrounddetect.so [ 22%] Built target app_nv_backgrounddetect [ 33%] Building C object CMakeFiles/app_nv_faxdetect.dir/app_nv_faxdetect.o Linking C shared module dist/app_nv_faxdetect.so [ 33%] Built target app_nv_faxdetect [ 44%] Building C object CMakeFiles/app_pickup2.dir/app_pickup2.o Linking C shared module dist/app_pickup2.so [ 44%] Built target app_pickup2 [ 55%] Building C object CMakeFiles/app_rxfax.dir/app_rxfax.o cc1: warnings being treated as errors /root/asterisk/agx-ast-addons/app_rxfax.c: In function 'phase_e_handler': /root/asterisk/agx-ast-addons/app_rxfax.c:126: warning: implicit declaration of function 't30_get_local_ident' /root/asterisk/agx-ast-addons/app_rxfax.c:127: warning: implicit declaration of function 't30_get_far_ident' /root/asterisk/agx-ast-addons/app_rxfax.c: In function 'rxfax_exec': /root/asterisk/agx-ast-addons/app_rxfax.c:380: warning: implicit declaration of function 't30_set_local_ident' /root/asterisk/agx-ast-addons/app_rxfax.c:383: warning: implicit declaration of function 't30_set_header_info' /root/asterisk/agx-ast-addons/app_rxfax.c:385: warning: passing argument 2 of 't30_set_phase_b_handler' from incompatible pointer type /root/asterisk/agx-ast-addons/app_rxfax.c:386: warning: passing argument 2 of 't30_set_phase_d_handler' from incompatible pointer type make[2]: *** [CMakeFiles/app_rxfax.dir/app_rxfax.o] Error 1 make[1]: *** [CMakeFiles/app_rxfax.dir/all] Error 2 make: *** [all] Error 2 u...@host:~/asterisk/agx-ast-addons# ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOID phones
> As part of this, I am going to be putting in a 30 handset Asterisk > solution. We are trying to keep the costs down as much as possible, as this > job includes cabling, I am looking at POE solutions. > > On the switch side, I am considering something like some Netgear ProSafe > FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24 port > switch. About 4 of these will run the phones and computers on the network > connecting back to a gigabit switch handling the phone and other servers. > > On the phone side VOIP phones > > The price range sort of limits me to: > > * Aastra 9112i > * Snom 300 > * Polycom 320 > * Cisco CP-7906G (But I believe this won't handle SIP out of the box?) With the exception of mass market stuff like their ATA's (which are actually all right) stay clear of anything Linksys. It's over priced, over marketed and undelivered, poorly supported crap. Too add to your owes a lot of it is very proprietary and designed to lock you in to Linksys. Some of the above could also be said about Cisco though I do have a Cisco router I am very happy with and they are the gold standard. Having said that you are not a carrier or an ISP, and you are obviously on a limited budget, so I would not use any of their VoIP stuff in your situation. The only Snom phone I ever used was a total piece of s*. Having said that some people seem to like them - I don't know why. If you can get Netcomm where you are I'd recommend their V90S phone. Over all it's a nice phone, works well, quite classy and importantly not proprietary. It does have a couple of things that need improvement on - namely that you have to set up multiple SIP accounts on the phone to support conferencing (It's conferencing button doesn't function with server based conference) and it's web based GUI doesn't work with Firefox atm. MSIE needed for Windows or Konqurer on Linux works fine. If you want these phones and you can't find them locally drop me an email. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
> Mentioning costs, one might be tempted to mention "grandsteam", but for > some people, those have a bad reputation, although i have two of thos > phone for over two years without any problem.. > > OTOH, why not consider the Siemens A580-IP? > Recently i bought a package, containing the DECT-base-station (direct > IP-interface) and two handsets, (each 6 sip-entries), two > handset-chargers for about 100 Euro's. > Audio-quality is good. Don't think you can buy SIP-phones any cheaper... My experience with Grandstream is that are one of the better 'cheap' ones, but cheap non the less. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Mon, 22 Dec 2008 02:27:29 you wrote: > Michael wrote: > > My experience with Grandstream is that are one of the better 'cheap' > > ones, but cheap non the less. > > I am yet to run into a worse IP phone than the Grandstreams - although > having said that, I should say that I've always steered clear of most of > the Chinese "no-name" brand phones. They're unstable, temperamental and > upgrading the firmware is a crapshoot half the time since you never know > what new bugs will be introduced and quite often you can't downgrade the > firmware if you don't like the newer firmware. +1 I STRONGLY recommend to the O.P. that whatever they do, whatever path they decide to take, that they *only* buy one or two units to test, and test them fully, until they are absolutely sure the item is not a POS. Nothing worse then being stuck with 30x POS. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
> One person's trash is another's treasure. > > I've used many linksys phones, including the SPA962 and found the > sound quality and usage to be simply sub-par. In several set-up's I > found the sound quality of a Grandstream 286 ATA to be much better > than a SPA962 IP phone. But, I agree in I find the polycom to be > amazing phones — and you end up paying for it (although on long term > cost of ownership it might not be that bad. I bought a Polycom 601 > years ago, use it heavily, and it sounds just as good today as the day > I bought it. The SPA962 went on ebay within 3 months of me buying it. > I have a few grandstream 286's I like to use for traveling and placing > in remote areas of an installation. > > Fred Posner 3 months... that long? I have a Linksys SPA9000 IP PBX I want to quit. Mint condition with all packing etc. Nothing 'wrong' with it (except that it's a Linksys) I just hate proprietary stuff which is what the Linksys is. Still it may well suit someone present on the list who doesn't mind that they will have to buy Linksys brand phones to work with it, and they want it because of the nifty "any n00b can use this" set up utility. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AGX addons
Is this package capable of receiving faxes where the up line connectivity is FoIP through a T38 SIP connection? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4withLenny
On Mon, 22 Dec 2008 23:46:28 Andrew Thomas wrote: > ...described in the README file ;). > > SpanDSP - 0.0.4 pre 16 > LibTiff - >= 3.8 but <4.0 > > > I had to trawl around for the right SpanDSP - but I can e-mail a copy to > whomever wants one (drop me a personal e-mail and I'll attach it by return) > > HTH > Andy How can I compile this without first installing Libtiff 3.8+ ? I have a patched version of 3.7.2 that I need to keep to maintain my Hylafax Jbig/Jpeg support. (Unless someone knows of a more recent Hylafax libtiff patch) Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4withLenny
On Tue, 23 Dec 2008 00:06:19 you wrote: > I'm not sure mate - as I don't use HylaFax on that particular server. > Hopefully, someone else can help. Sorry. I have installed the pre-requisite versions asked for and successfully got AGX addons installed. Except it doesn't work. "Failure to train remote modem at 2400" from the sending system. This suggests to me that it is trying to use VoIP communications. Is there a way to make this support T38 or do I just have to conclude that so far every option is lousy? I mean - how obvious is it to create something that works with T38? We don't all have cheap PRI circuits available. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem: no such extension 'xx' in context 'default'
> sip.conf : > > [general] > port=5060 > bindaddr=0.0.0.0 put "context=default" here > > [10] > type=friend > secret=1234 > host=dynamic > context=internal > > [11] > type=friend > secret=1234 > host=dynamic > context=internal > > extensions.conf > > [default] > exten =>2,1,Playback(digits/2) ; > exten =>2,2,Goto(default,10,1) > exten=>3,1,Playback(pbx-invalid) > exten=3,2,Goto(default,4,1) > exten=4,1,Playback(vm-goodbye) > exten=>4,2,Hangup() Change it to the following: exten =>_2,1,Playback(digits/2) ; exten =>_2,n,Goto(default,10,1) exten=>_3,1,Playback(pbx-invalid) exten=_3,n,Goto(default,4,1) exten=_4,1,Playback(vm-goodbye) exten=>_4,n,Hangup() > [internal] > exten => 10,1,Dial(SIP/10,10) > exten =>10,2,Background(vm-nobodyavail) > exten => 11,1,Dial(SIP/11,5) > exten =>11,2,Background(vm-nobodyavail) > > now when I dial 10, I got the following error : no such extension '10' in > context 'default' Change it to the following: exten => _10,1,Dial(SIP/10,10) exten =>_10,n,Background(vm-nobodyavail) exten => _11,1,Dial(SIP/11,5) exten =>_11,n,Background(vm-nobodyavail) The only time I am aware of that you can leave out the prefix underscore is for "exten => s" and "exten => i" Hope this helps, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem: PART TWO
> now when I dial 10, I got the following error : no such extension '10' in > context 'default' As anorther important note, your PBX is correct. You should change the line Goto(default,10,1) to Goto(internal,10,1) assuming that's what you want! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem: no such extension 'xx' in context 'default'
I was typing so quick I made some slips- (anyway you should get the idea...) Change it to the following: exten =>_2,1,Playback(digits/2) ; exten =>_2,n,Goto(default,10,1) exten=>_3,1,Playback(pbx-invalid) exten=>_3,n,Goto(default,4,1) exten=>_4,1,Playback(vm-goodbye) exten=>_4,n,Hangup() > [internal] > exten => 10,1,Dial(SIP/10,10) > exten =>10,2,Background(vm-nobodyavail) > exten => 11,1,Dial(SIP/11,5) > exten =>11,2,Background(vm-nobodyavail) > > now when I dial 10, I got the following error : no such extension '10' in > context 'default' Change it to the following: exten => _10,1,Dial(SIP/10,10) exten =>_10,n,Background(vm-nobodyavail) exten => _11,1,Dial(SIP/11,5) exten =>_11,n,Background(vm-nobodyavail) The only time I am aware of that you can leave out the prefix underscore is for "exten => s" and "exten => i" Hope this helps, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wich gateway is much better?
On Sat, 27 Dec 2008 18:42:12 Abel Monzon wrote: > Hello everybody, I have a doubt > > > If I want to send every call from a server asterisk to a gateway to a > line PSTN, in the gateway what type of port I need FXO o FXS? I need > to know wich gateway to buy, with port FXS or FXO? > > regards, > Abel FXO = Goes to PSTN FXS = Goes to telephone Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith, Music-on-Hold Parody--outstanding.
On Fri, 02 Jan 2009 11:44:40 Paul Hales wrote: > Andrew Joakimsen wrote: > > On Wed, Dec 31, 2008 at 22:09, Paul Hales wrote: > >> Karl Fife wrote: > >>> Allison Smith just created a hysterical parody music on hold Parody. > >>> Whatever you were doing, stop, and dial this number to listen to it: > >>> 360-519-5689. 2 minutes. > >>> > >>> I just gave her a few ideas, but she took it and ran with it--she > >>> chose the audio and did the mix-down and everything. Really funny!! > >>> > >>> -Karl > >> > >> Any chance of us non-us citizens hearing it? > >> (podcast, download...) The .wav didn't work so I phoned up. She talks too much. That alone would make me want to hang up. Michael Auckland, New Zealand ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco VoIP QOS
I have currently following the instructions at- http://www.voip-info.org/wiki/view/QoS+Cisco I am using a Cisco 877 router with an ADSL2+ connection. My upstream bandwidth is around 1Mbps, and my downstream around 4-5Mbps. I have two PSTN side interfaces - atm0 and dialer0 (dialer0 is where the ISP details are configured and the interface that the WAN side IP is applied on) Now which interface to I apply the QOS on? I would have thought dialer0, but for some reason this interface does accept is citing that it only has "56kbps" of bandwidth for some reason. Now I could fix it's bandwidth at 1000kbps, but as anyone who uses ADSL will know, this may not be a good idea as ADSL is rate adaptive, meaning that the bandwidth can increase and decrease due to line conditions. What do I do? I tried "bandwidth inherit" though I really don't know how this command works and it doesn't seem to make it want to 'inherit' the current status of atm0... Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet8 hacked
On Fri, 23 Jan 2009 20:13:22 Peter Evans wrote: > On Fri, Jan 23, 2009 at 07:08:16AM +, Gordon Henderson wrote: > > On Fri, 23 Jan 2009, Dean Collins wrote: > > > Nope it's going to a sedo advertising domain parking site. > > > > Looks like a normal website to me (from the UK). I'd check that the DNS > > servers you're using haven't been hacked... Fine from New Zealand too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted information
> Cut this crap from your email. Its wasting my bandwidth. +1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
> >Lets start with some logical points here: > > > >1) 24 Ports x 15.4W/Port = 369.4Watts + Switch Power = ~400Watts... now > >Power Supply isn't that efficient so you're getting probably a 500Watt > >Power Supply (assuming 80%)... > > It'd still be a 400W PSU if it supplies 400W. > > >2) with a 1U chassis, you can't blow air up or down... only front and > >back.. so you're stuck with a 40mm fan.. > > And the sides... > > ... you can fit a muffin fan horizontally as suggested and allow it > to draw in or to blow air at the top or botton within the height of > the 1U. Muffin fans are 25 to 35 mm "high". This whole thread is getting stupid and I'd hope the people involved would desist from this O/T drivel. If you want a switch go to the shop, hand over some money and buy one... Like every one else does and they're perfectly happy with their purchase. The O.P. is not going to change the world and quite frankly the designer/manufacturer of the product knows a lot more about the subject then they do ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2950 -> 4 connections -> Cap of 512 Kbps -> How to bond ?
On Fri, 13 Feb 2009 12:41:51 Jeff LaCoursiere wrote: > Get a Cisco with five ethernet ports. Use one for your connection to > asterisk. Use the other four as your connection to the ISP, and MUX them. > > Great way to spend 5K :) > > j > > On Thu, 12 Feb 2009, Vikas wrote: > > I have asked the ISP to rate limit a single port to 2M but my requests > > have got me no where, > > > > I would really appreciate any suggestions on what I can do at my end > > since I have given up hope of the ISP co-operating with me, As someone who works for an ISP, the best advice I can give you is to tell them where to go (*after* fully setting up and testing a new ISP that is). With the economies of the world tighter then usual at present, and ISP's a plenty, I can only suggest they are idiots or for some reason they don't want your business Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with asterisk
On Mon, 16 Feb 2009 22:14:49 Fabio Mosti wrote: > Hi All, > > I need to setup asterisk to receive fax. > > I'm try Spandsp (opensource) and Attrafax (commercial) both on > asterisk 1.4.23) but the results are disappointing. > with spandsp many times the fax arrives cut. > with Attrafax i have some problem. > > Anyone have any idea or solution (Opensource or commercial) to suggest me ? Unless you have a very stable connection with no contention (like a leased line or fibre) to your provider, give up now. ADSL, cable etc doesn't work properly. FoIP is a lot worse then VoIP. Michael. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with asterisk
> > Anyone have any idea or solution (Opensource or commercial) to suggest me > > ? > > > > Best Regards > > > > Try hylafax with IAXmodem. The best results i had it the multitech modems > > directly connected to FXS PCI card, you have a nice web interface if you > wish also (avantfax) You can find some nice install scripts at the elastix > forums. Best results are with Hylafax and Multitech serial modems connected directly to the PSTN. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SpanDSP question for Steve
Hello- Firstly thanks very much for the work you have put into SpanDSP and the time you spend to assist people here :-) I am currently running SpanDSP 0.0.5 with Call Weaver. Is there any or sufficient gain to be had from upgrading SpanDSP? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile
I am trying to test the chan_mobile functionality. i have not been able to compile the SVN version. I get an error about needing ncurses when doing make menuselect. I have verified that I have libncurses5-dev installed. I have complied version 1.6.0-beta9 w/ asterisk-addons-1.6.0-beta4 with success, but encountered an issue when trying to pair the device. It will not accept the PIN. I am using linux distro: Debian unstable 2.6.26-1-686 #1 SMP any recommendations? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed
On Wed, 18 Feb 2009 13:37:57 John Todd wrote: > I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on > Sunday in Los Angeles, and the topic of my talk is "Open Source in an > Economic Downturn". I've got lots of talking points for this talk, > but it would be interesting to hear some short anecdotes about how you > in the Asterisk community are thriving, or at least surviving, by > virtue of the benefits of Open Source. I find that real-world > examples are worth more than all of the bullet points in the world, > and timely stories from the community would be more interesting than > hearing me prattle on. What "economic downturn"? I'm sick and tired of hearing this mantra. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed
> > What "economic downturn"? > > > > I'm sick and tired of hearing this mantra. > > Same here (in the UK). > > As long as people need to make phone calls ... > > Gordon The economy (and indeed humanity as a whole) needs periods of removing sludge, deadwood, and general stupidity. Having said that I would think that many of the list contributors are based on the USA, so no surprises they would feel that way. I thought our media left a bit to be desired until Youtube came along and I could see the propaganda and mindnumbing dross trotted out in the corporate controlled media there. So people should decide for themselves whether to think positive or negative thoughts. I for one intend to take full advantage of the opportunities presented by this period of transition in human affairs. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed
On Thu, 19 Feb 2009 08:33:56 Chris Bagnall wrote: > > Perhaps there isn't a downturn in a country with more sheep than people > > That's a little harsh, New Zealand is one of those places that really > appeals as a decent place to live. It is. And when it isn't so hot (mid summer) Australia is also quite decent. > At the end of the day, we got ourselves into this mess (cheap credit, > lenders not checking people could actually repay their debts, etc.), we > have no-one to blame but ourselves. Add to this (from the perspective of anyone living there) - a stupid war and throwing handouts at car makings who make products that people don't want to buy - watch 'who killed the electric car' on youtube. The poster is based in the USA, so it could well be that bad for him. As for his opinions about the rest of the world, well Fox and CNN are always reliable and educational. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
On Thu, 19 Feb 2009 13:35:25 Steve Edwards wrote: > On Wed, 18 Feb 2009, michel freiha wrote: > > I suggest please if someone advice to me a free PDF book just dedicated > > for AGI and nothing else > > It takes a rare individual to put the effort required to write a book and > then distribute it for free. > > I would like to write it, but my kids have grown accustomed to eating :) This is everything that is wrong with Open Source - no body wants to pay for anything ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
> >> I would like to write it, but my kids have grown accustomed to eating :) > > > > This is everything that is wrong with Open Source - no body wants to pay > > for anything > > Hi my name is nobody. I like to pay for many (FOSS) things. Not as > much as I'd like but when the right Powerball ticket makes it into my > hand I'll do more. :-) And you're going to stay this way because the ahem... 'customer' knows it's free and pays you accordingly. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
> This has absolutely nothing to do with the fact that something is > opensource. The fact that the source is "open" has nothing todo with its > pricetag. Sometimes opensource products are more expensive then closed > source products. > > If you want support/maintenance/dedicated_features/you-name-it you'll > have to pay for it. But you only pay for what you want/need, and not > because some egghead decided what he wants to put together as a > sales-package. > > Opensource is about the freedom to check and to change, security, > quality. If you doubt it, check with SLES/RHES/ABE/... > There even seems to be companies that do _only_ support on open > products, like typo3, openoffice, And make a living out of it. Big companies, especially those with major computing systems use paid software because they want a vendor they can hold responsible for it. As for OSS and FOSS, it is majorly used by the sort of businesses and individuals who call me (and other IT pros) up and talk the talk, but they don't have a 2 dimes to rub together. This problem is only going to get worse as the so-called 'recession' bites... fellow I.T. professionals - get used to your clients trying to weasel free service out of you. Everything I am hearing from fellow I.T. people is that there is no shortage of 'work' but a lot of clients are resisting paying. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
\ > No... there's no shortage of work that needs doing. But there's a > definite shortage of money to pay those to do it -- hence the massive, > worldwide layoffs. Your little corner may not be affected, but to > discount basic economics because you don't see it? Well... that's > incredibly short-sighted and provincial. > > Expect a bigger push to FOSS simply because fewer companies can afford > what they used to be able to afford. They can't get loans. The people > who buy their services and wares have all but vanished, so they have no > influx of capital. This is not some 'media-created' concept. > > There's some incredibly good OSS and FOSS out there (Asterisk is a case > in point). People who sneer at companies that use it, saying they're > somehow lesser than companies that don't are, I usually find, those who > are making a living overcharging for their products. This could become quite a long winded O.T. discussion. Unfortunately not appropriate for this list. I use mainly FOSS in my business. But the days of me providing technical support services to others who do are fast coming to an end. FOSS by and large attracts a certain type of 'client', one who is rarely commercially viable. Given that I have other business interests where firm orders just turn up, almost nil support is required, and 100% of the customers pay their bills (nonwithstanding a few are slightly late). The hassle, the tyre kicking, the 'you have to be dreaming' and the chasing up of overdue accounts, including having to write off some of them, the endless discussions and briefings with customers on small value projects. It's a PITA I'd rather do without. Sorry people. It's only a few short years ago I was in the nerd camp and trotting out the same mantras about how great it is to have open source... But I am now a bit older, slightly more wiser and with a lot less ideological view of things. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
> Not entirely. I've been trying for two years to get someone to work with > my small Linux system. One guy never had time to come. I finally got > someone out who was going to charge either $125 or $175 per hour (USD) > depending on whether he decided it was a computer problem or a network > problem (which is about twice what I charge for Embedded hardware and > software development). He spent an hour here and had to go to his next > appointment. My little Samba problem was beyond his ability to solve! > Fortunately, he realized that he hadn't done anything and didn't charge. That sounds quite expensive. The only way ahead I can see for FOSS to become widely used in business is to integrate it in to *commercial* solutions, where the client is buying a *commercial* product, complete with the "i's" dotted and the "t's" crossed, like Cisco or Avaya does. And their solutions are expensive. The service is expensive. And guess what... the customer buys it! Where the 'client' is using FOSS to save costs or for ideological reasons - this is where the headaches are for I.T. professionals. The clients more often then not are nerds, DIYers, running a business in a 'tight' industry (like Internet Service provision), or just simply not in a position to pay the bill. Sorry if this sounds overly cynical, but when 9/10 cases can be fitted in to this category, the other 1/10 does not make a good business case. Further more I have been in the industry for long enough to have heard the same flawed, unworkable, commercial non goer business schemes more then enough times over... I do not wish to have any involvement in yet another wireless ISP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
> I also think you should check the economic stimulus package. There are > billions of dollars allocated to ISPs. It could be a windfall. Yoohoo! Let's print some more money. I don't think that's been tried before ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
> You must of course know all about how things are working here in the > states while you sit in NZ. The money is not being printed. It is being I am well informed thanks to the global reach of such excellent purveyours of quality news and information such as Fox and CNN. > raised and accounted for in debt, which we will have to shoulder going > forward. I'm a bit put off by your recent holier than thou comments with "holier then thou"?... that's a bit strong. Of course I know the US government is borrowing the money from the privately controlled "Federal Reserve", so the US taxpayers can foot the bill with interest. > regard to what is going on over here. It is a serious situation that is > affecting many people, myself included. I have lived overseas and know > the opinion most other countries have of the US. They think we look down > on them and how they live. Which is *exactly* what happens. > stock and think about keeping your comments to yourself. You never know > when they will come back to bite you. Oh... dear... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
> Hey I am against any Federal bailouts (wrote in Ron Paul and donated to his > campaign for presidency as well as being an active member of the "Campaign > for Liberty" http://campaignforliberty.com) , but they are a reality. Shame Ron Paul was shafted of fair media attention. He would be an excellent president for the US. > Only a fool would not align themselves with the market and the money. > Maybe that is why you seem to have such difficulty with obtaining "Paying" > customers. You are mistaken. I have no difficulty finding "good clients" But then I aren't trying to squeeze myself in on free software deals with them either. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
> Money is borrowed at the Prime Rate which is between 0% and .25% but thus > far, China and Japan have been "lending" the largest portions as well as > other foreign countries. I am not saying this is any better but you should > get your facts straight. I am well aware of this. I am also well aware that this subject is O.T. and as such my two cents worth is kept deliberately simplistic. But yes, the China Development Bank is (supposedly) sitting on US$ 2.5 trillion of greenbacks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
On Fri, 20 Feb 2009 14:28:23 you wrote: > Is this the asterisk users list? or some political list? or maybe i > dunno, i am confused Sorry. I'll make that my last post on the subject. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] weird problem
I am running Asterisk 1.4.22.2, though I have also found this problem with 1.4.23.x Sometimes after I hang up the system continues to spew packets to my phone causing it to become unusable until I restart Asterisk. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] weird problem
On Wed, 25 Feb 2009 10:18:02 you wrote: > use wireshark or somethink like it (tcpdump) and see if the "bye" is > reaching asterisk. > if this is the problem you can use rtptimeout option in the sip.conf or > iax.conf. > David > > 2009/2/23 Michael > > > I am running Asterisk 1.4.22.2, though I have also found this problem > > with 1.4.23.x > > > > Sometimes after I hang up the system continues to spew packets to my > > phone causing it to become unusable until I restart Asterisk. > > > > Michael It usually seems to happen after I use voice mail. I have no 'rtptimeout' set, what should this be? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA
> > I am wondering how can I get the bandwidth.com service, > > Why would you persist with a company who can't service your needs when > there are others who will? +1 This industry is full of companies staffed by morons who don't give a s*. Then these companies go bust... and the idiot owners wonder why. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rfc2833 vs. sipinfo and network weirdness
Further to a recent post about a problem whereby the server continues to spew packets to the phone after hangup (sometimes, not every time), I have found that this problem appears to be alleviated by using RFC2833 instead of SIP INFO, however in switching to RFC2833 I introduce another problem - that DTMF tones for navigating menus become unreliable. RFC2833 and SIP INFO are the only 2 options supported by the phone. Inbetween the phone and the server is the internet and 1 NAT traversal. Any ideas? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Called number as variable - how to?
I want to obtained the called number (aka DID) as a variable. How is this
done?
(Upline connectivity is via SIP provider).
With verbose = 100, the called number is clearly shown as
follows "Wait("SIP/[called number]-64e6", "1")" for example, but I can't find
the info as to how to get it as a variable.
Michael
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Re: [asterisk-users] $20 Bounty
>I'd be willing to wager someone another $20 that he does have a > "taker" on that bounty within the next 48 hours. For those of us that > reside in countries where a single cup of coffee can run more than 20% > of that complete bounty, we scoff at the opportunity. For others, their > family may be able to live for half a week on that money. For better or > for worse, it's a global economy now when it comes to this kind of > stuff, and I have no doubt that Dean will get a taker on his offer. I'm in New Zealand (yeah, I know - where's that?) and a cup of coffee can run to the NZ$ equivalent of US$ 2.00, so I am not interested either. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $20 Bounty
> >>I'd be willing to wager someone another $20 that he does have a > >> "taker" on that bounty within the next 48 hours. For those of us that > >> reside in countries where a single cup of coffee can run more than 20% > >> of that complete bounty, we scoff at the opportunity. For others, their > >> family may be able to live for half a week on that money. For better or > >> for worse, it's a global economy now when it comes to this kind of > >> stuff, and I have no doubt that Dean will get a taker on his offer. > > > > I'm in New Zealand (yeah, I know - where's that?) and a cup of coffee can > > run to the NZ$ equivalent of US$ 2.00, so I am not interested either. > > here in North America, starbucks coffee with all the whipped topping and > cinnamon, etc will set you back about $5, so $2 is actually a bargain :) That's from a normal reasonably priced cafe, around NZ$4.00 At Starbucks here it will cost around US$ 2.5 - 3.00. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] faxing via linksys SPA3102 half page goes through
On Wed, 04 Mar 2009 19:25:38 Joseph wrote: > I'm faxing from stand alone fax machine via linksys SPA3102 but most of > the time only half or quarter page goes through. > > Did anybody have any experience like this? Should be obvious but does your up line SIP provide support T.38? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk dial plan conditional on not busy
Here is the current dial plan section:
[custom-michael]
exten => _900,1,Playback(custom/extn-xfer)
exten => _900,2,SayDigits(${EXTEN})
exten => _900,3,MixMonitor...
exten => _900,4,Dial(SIP/${EXTEN}|${DEFRT})
exten => _900,5,Playback(custom/extn-xfer2)
exten => _900,6,Goto(custom-michael,901,4)
exten => _901,1,Playback(custom/extn-xfer)
exten => _901,2,SayDigits(${EXTEN})
exten => _901,3,MixMonitor..
exten => _901,4,Dial(SIP/provider/${CELL1}|${CELLRT})
exten => _901,5,Goto(vmail|900|1) <-- go to voice mail context
DEFRT = 30 seconds, CELLRT = 25 seconds.
Currently if the first phone is busy, or is not answered within 30 seconds, it
will jump to the next phone (a cellular phone) and try that for 25 seconds
before going to voice mail.
What I want it to do is only try the cellular phone if the extension '900' is
not answered. If busy I want it to go straight to voicemail.
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[asterisk-users] Dial in / dial out
Anyone know of a good dial plan example for call in / call out? I want to be able to call my Asterisk server, auth, and then call out any number. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial in / dial out
Thomas Stein wrote: > On Monday 23 March 2009 11:01:40 Michael wrote: > >> Anyone know of a good dial plan example for call in / call out? >> >> I want to be able to call my Asterisk server, auth, and then call out >> any number. >> > > http://www.voip-info.org/wiki-Asterisk+cmd+DISA > > t. > > > Thank you - though I can't get it to work past the password entry. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
haw haw haw... April Fools Day is over in this part of the world. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
On Wed, 01 Apr 2009 21:01:28 you wrote: > 2009/4/1 Michael > > > haw haw haw... > > > > April Fools Day is over in this part of the world. > > Hey dont kill the magic ! :) April Fools Day ends at 12.00pm (mid day) here. It is now 9:07pm. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ring group howto
How do I manually set up a ring group? All the info I've Googled tells me how to do this using Trixbox or FreePBX. I am using standard Asterisk 1.4 configuring at the CLI. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring group howto
On Fri, 03 Apr 2009 12:32:03 you wrote: > A group of phones that ring all at once? > > Like: > > exten => > 5226001454,1,Dial(SIP/3615221401&SIP/3615221402&SIP/3615221407&SIP/52260014 >0 5,20) > > Take out the line breaks. > > Or were you looking for something else? > > CF That is what I am currently doing - though is there a cleaner way? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP and FW settings
Hello all- As a general rule, the INPUT FW rules for SIP should be 5060 UDP and 1-2 UDP right? Is TCP used in any part of the SIP structure? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and FW settings
On Tue, 14 Apr 2009 20:47:29 you wrote: > Hi michael, > > you should open both tcp,udp 5060,5061 too and as you mentioned between > 1-2. AFAIK 5061 TCP is for TLS SIP which isn't used much yet? Is TCP the default for 5060, with UDP as fallback, or is this provider dependent? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Tue, 14 Apr 2009 23:00:02 Florian Hackenberger wrote: > Can somone spot the problem? Is someone using t38modem with asterisk > successfully? > > Cheers, > Florian The best advice I can offer is to give up now and use Callweaver otherwise you can spend hours, or days, with no working result. Fax support is Asterisk 1.2/1.4 is effectively non existant and nobody on this list or Hylafax list is interested. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion
On Wed, 15 Apr 2009 00:30:43 David Backeberg wrote: > Once it's a wav you can mp3 it with lame or your preferred encoder, > but encoding and playing mp3s takes more cpu than just playing it in > gsm, or stopping after sox and playing as a wav. > > > Has anyone got any suggestions based on previous experience? I convert to MP3 and delete the original wav file using a script automatically executed by the Monitor command (because wav is very large files). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Wed, 15 Apr 2009 00:43:45 you wrote: > Now for the part I do know something about. Native asterisk fax > support and native asterisk sip support improved in 1.6. With 1.6 > there is a built-in app_fax module which works quite well for sending > fax over SIP with T.38. I found the configuration and debugging > simpler and more understandable. I never really knew why my > asterisk-1.4 faxing experiments went so badly, and I had no reason to > go find out once my asterisk-1.6 faxing worked so well. > > I will say as an opinion: > Openh323 + ptlib_unix + t38modem + hylafax + asterisk-1.4 > > sounds like a lot harder to troubleshoot than: It's hell (from experience) > asterisk-1.6 with app_fax built-in > > Try 1.6. You'll be glad you did. While I have not tried Asterisk 1.6 because I settled on Callweaver at the time (which has native T38 support), I *strongly* recommend going with software that has native T38 support. This could be Asterisk 1.6 or Callweaver. So +1 for the above. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP ports
[Apr 15 11:12:19] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:37259, rtcp halted Operation not permitted [Apr 15 11:12:23] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:38563, rtcp halted Operation not permitted What is the specific nature of this traffic? Despite the above the call still functions. What is the appropriate FW rules? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Fri, 17 Apr 2009 21:39:53 Florian Hackenberger wrote: > Could you please describe your setup in more detail? Where does you SIP > provider send FAX calls to? Directly to callweaver or to asterisk 1.4 > which forwards it to call weaver? Do you have hylafax in the mix as > well, or does you callweaver instance terminate and originate the FAX > calls itself? Trying to link Hylafax < - > Callweaver or Asterisk is unfortunately a waste of time. You need to use the built in fax support and write from scratch the necessary scripts and dial plan to deal with faxes. I have successfully got a T.38 set up working, but it wasn't devoid of a lot of bother along the way. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Fri, 17 Apr 2009 22:35:04 you wrote: > And how do the T.38 calls get to callweaver? Directly from the SIP > provider, or does asterisk forward them to callweaver? Which version of > asterisk (on which distribution) and which version of callweaver are > you using? Directly from a T.38 capable SIP provider (1). I use the 1.2.0.1 of CW on Slackware. 1. If you want you can run this on the same box as an Asterisk setup. Just make sure you multihome the system and bind all of Asterisk's / Callweaver's services to IP's. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Sat, 18 Apr 2009 00:54:07 you wrote: > Well, not really a waste of time. As I mentioned - Hylafax has many > desktop clients, it's better to just write few scripts than to design > desktop software for your own setup. If You have a need of sending > faxes, You'll probably need a desktop client too. I wish it wasn't a waste of time because Hylafax would make a great front end to a T.38 SIP system. I use Hylafax myself, but not with SIP. The problem is that there is no reliable, or really any viable way to achieve this when using T.38 as the carrier uplink. The setup you describe does not have a audio data path connection to Hylafax and I wonder why the convoluted method when the same could be achieved using Callweaver alone and some custom scripting. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
