Re: [asterisk-users] A TrixBox 2.0 seems to be asleep...
Mauro Zanin wrote: Hi everybody, I installed a 3.0Gb 512MB TrixBox with a Celeron inside. The PC seems smart enough an rapidly the CentOs and Asterisk were loaded on it. I has only 2 extensions(SIP telephones, one GPX2000 and one Grandstream 486) and one ISDN adapter with Bristuff. Web server has about a delay of 20 seconds to startup, while after started the response is acceptable. But incoming and outgoing calls are delayed about 5 second before the * takes a decision to ring an extension or to place a call. Is like the whole system is taking an holiday each time! I have installed many TrixBox and asterisk, but never had such an issue! Any suggestion, apart the one to trow away Celeron and put a pentium in its place... Ciao Mauro Mauro, This sounds like an IPv6 or DNS issue. Verify good connectivity to your primary DNS server and in /etc/sysconfig/network-scripts/ifcfg-eth0 set IPV6INIT=no. You will probably need to reboot if the latter change is required so that the IPv6 modules are unloaded. To speed up DNS maybe a local caching name server would help. Bob... ___ --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] chan problem
[EMAIL PROTECTED] wrote: I experienced the same problem. The only way I could get both ISDN and analog working was unloading kernel modules for zaptel and mISDN after boot and then load them in the order: zaptel first and then mISDN. Still need to find out how to configure load order in linux. grz, Hans Feringa On Mon, Jun 18, 2007 at 12:50:23PM +0200, Josu Lazkano wrote: Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) This means your /etc/zaptel.conf is incorrect. Start by correcting it. Or maybe the driver has not loaded correctly. Which OpenVox card is it? What is the output of: lspci cat /proc/zaptel/* zaptel.conf: loadzone=es defaultzone=es fxsks=1 - Hans, Have a look at the man page for modprobe.conf, specifically the install directive. There is an example of how to force the order. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Termination with automatic debit
Douglas Garstang wrote: Can anyone recommend any wholesale SIP termination providers that will automatically charge a credit card? Most seem to want upfront payment and a credit balance but that sucks when you have to watch it like a hawk to make sure the balance never hits zero. I’m looking for a provider that can automatically charge a credit card. Douglas. Douglas, Voice Eclipse. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing Volume
Jadrien Wauthier wrote: Does anyone know how to adjust the volume of the ringing application? I have done a lot of internet searching and have not found much. You cannot do this in Asterisk. Some SIP phones might allow you to do so by setting an option on the phone, but you would have to ask the company that makes that specific phone how to do that. If Asterisk generates the audio, then it seems that there would be a source file that I could edit if nothing else. I looked at the app_dial.c, but I didn't see anything. Maybe I over looked something. If I lower the volume on the phone, then all audio on the phone would be lower. I am just interested in lowering the volume of the ringing. Basically, rings from the pstn is at one level, and the rings from Asterisk are at another level. I need to normalize the Asterisk volume. Thank you so much for your help with this. Jad Jad, Are you referring to the ring back (progress tones) when you call out? I have the same issue. Depending on the type of interface you have to the PSTN, you could try raising the inbound gain from the PSTN to match that of asterisk. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MRTG with 4 graphs
Ronald Wiplinger wrote: How can I set-up a MRTG with 4 graphs, whereby: 1 data in 2 data out 3 ONLY voice(/video) data in 4 ONLY voice(/video) data out bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ronald, What do you want to gather data from? Switch, router, asterisk? Model/Manufacturer. Are there MIBs specifying the info you want to gather. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID on Dish 301 Receiver
Hugh L. Johnson wrote: I have a Dish 301 receiver that will not display CallerID when connected to FXS module on TDM400. Uniden phone connected to the same FXS module does display CallerID. When Dish 301 receiver is connected to IAXy CallerID is displayed properly. Any suggestions on getting the CallerID to display on the Dish 301 receiver through the TDM400 FXS module? Hugh Hugh, Maybe tweaking the gain will help. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
Bernardo Vieira wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Gordon Henderson wrote: If you only have one * box behind the NAT gateway then I don't really see a big issue with it to be honest. Port-forward on the firewall/router device (5060 and 1 through 2) to the * device, and use STUN on the client device to help it get through it's local NAT firewall/router. I use the same strategy and it works just fine, however, I did have an issue, here's the scenario: SIP Ext 1 +---+ |--- NAT -- Internet -- NAT -- Asterisk SIP Ext 2 +---+ Each of the SIP extensions work individually, but if I try to use both of them, only the first one registers. Bernardo, Just a thought: Try using a different SIP port for one of the extensions, if possible. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
Bernardo Vieira wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bernardo, Just a thought: Try using a different SIP port for one of the extensions, if possible. Bob... Bob, Tanks for the tip. I had actually done that before, as a matte of fact that's the solution I have in place now. The thing is, even though it works it is not exactly the best solution as it forces some non-standard configuration on the clients. It's not my case, but imagine the hassled it would impose in an ITSP environment. Bernardo Bernardo, Yes. I have a home system running on Trixbox with an ITSP that provides two channels, but I can only register one of them as I cannot register twice on port 5060 with one IP address at each end. Both DIDs work, I just cannot determine which one a call is coming in on. The ITSP always reports the DID I registered last. The other drawback is the inability off-load the media path. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/RTP Nat problem, can't solute it.
Isn't that what externhost=sip.server.com.ar my server name on the internet localnet=192.168.5.0/255.255.0.0 my LAN is supposed to do? Bob... Rudolf Ladyzhenskii wrote: NAT changes address of the packet, but does not go inside of the SIP packet itself. And SIP packet contains address as well. If you look at debug output, you will see that SIP packets have remote host local address in them, not the public IP as one would expect. At least this is the problem I have. Basically one needs some software to NAT the addresses inside of SIP packets. STUN server is one alternative. I am about to put one in. Rudolf On 1/7/07, C F [EMAIL PROTECTED] wrote: Change To canreinvite=no On 1/6/07, Facundo Barrera - GMail [EMAIL PROTECTED] wrote: Dear list: I have the typical one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar my server name on the internet localnet=192.168.5.0/255.255.0.0 my LAN nat=yes canreinvite=yes And this are the ports i opened on my firewall script iptables -A INPUT -p udp -m udp --dport 8766:35000 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT But still can't hear a thing from an outside call, any hel will be appreciate Thanks a lot -- _ Facundo Agustin Barrera -- www.openlabs.com.ar Let the penguins do the work - Buenos Aires - Argentina _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe a NAT problem
Facundo Barrera - GMail wrote: Hi list: This is my first post and first off all i want to wish a good year for everone! well my problem is; i already installed asterisk on a server and created a channel and a couple of extensions, all seems to work just fine, y can make calls and receive them, i'm using the x-lite client that also works very good, this is the topology of the net (LAN - some clients) || Internal interface-private IP(server Running Asterisk)external interface-public IP ||-INTERNET Well i configure * to bind all address, so it's service listen on the two interfaces, when i make a call from a client inside my LAN to a client on the INTERNET, the person receives the call and listen me perfectly, but i can't listen any audio from him, i read about the issue and it seems to be a problem of nating, keep in mind that this server is masquerading all my LAN ips, so i can share my internet conenction, so when i receive a call form the outside world in fact x-lite shows me that the call originate from my inside interface IP of the server, but this is the strange thing the packets that originate the call from the outside world arrive just fine but when i answer the call i can't hear any audio at all. Any ideas how to solute this? hope not receive too much flames of this common issue Thanks a lot In your SIP configs specify that the extensions are natted: nat=yes externhost=External IP address localnet=Local IP subnet/local subnet mask These are global settings. It might also be helpful to set canreinvite=no for each extension. There are probably firewall tricks you can do as well, but its early and I'm a couple cups of coffee shy. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)
Kenneth Padgett wrote: Bob, It looks like the gnutls development package is called gnutls-devel: 'yum install gnutls-devel' should get the package installed. Yah, I thought that would be it. I have that installed, as well as gnutls. (I basically installed both packages you can find with yum search gnutls). Any other thoughts, can I just d/l the libs and uncompress them somewhere? -Kenneth Kenneth, I don't have a Centos machine at home, but under Fedora Core 6 autogen.sh and ./configure work after installing gnutls-devel. I followed the instructions at: http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk as you suggested in your first post. Without gnutls-devel installed the autogen.sh step fails. I would have thought that FC6 and Centos 4.4 would be pretty close as far as directory hierarchy. Here is a list of the pertinent files from the FC6 gnutls-devel package: rpm -q --filesbypkg gnutls-devel gnutls-devel /usr/bin/libgnutls-config gnutls-devel /usr/bin/libgnutls-extra-config gnutls-devel /usr/include/gnutls gnutls-devel /usr/include/gnutls/compat.h gnutls-devel /usr/include/gnutls/extra.h gnutls-devel /usr/include/gnutls/gnutls.h gnutls-devel /usr/include/gnutls/openpgp.h gnutls-devel /usr/include/gnutls/openssl.h gnutls-devel /usr/include/gnutls/pkcs12.h gnutls-devel /usr/include/gnutls/x509.h gnutls-devel /usr/lib/libgnutls-extra.a gnutls-devel /usr/lib/libgnutls-extra.so gnutls-devel /usr/lib/libgnutls-openssl.a gnutls-devel /usr/lib/libgnutls-openssl.so gnutls-devel /usr/lib/libgnutls.a gnutls-devel /usr/lib/libgnutls.so You might want to check it against your Centos installation. If it's different try: ./configure --prefix=/usr --with-libgnutls-prefix=PFX Where PFX is the where libgnutls is installed (from ./configure --help). Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)
Kenneth Padgett wrote: I'm working from the docs here: http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk and getting an error doing the ./configure on the iksemel module: checking for getaddrinfo... yes ./configure: line 20399: syntax error near unexpected token `,' ./configure: line 20399: `AM_PATH_LIBGNUTLS(,' It seems to want the libgnutls-dev package as per the documentation. Problem is, I can't seem to find such a package for centos 4.4. Anyone have any advice? Thanks! -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Kenneth, It looks like the gnutls development package is called gnutls-devel: Available Packages Name : gnutls-devel Arch : x86_64 Version: 1.0.20 Release: 3.2.3 Size : 503 k Repo : update Summary: Development files for the gnutls package. Description: The GNU TLS library implements TLS. This package contains files needed for developing applications with the GNU TLS library. Someone needs to fix this description. 'yum install gnutls-devel' should get the package installed. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE - Analog Phones with FSK/Stutter MWI
Doug, Thanks for the info. I'm glad it works. One question: Is there some sort of one-button way to dial in to your voicemail? It seems I read something about it, when I was doing similar research? I think it was the Uniden CLX-465, which claims support of Phone Company voicemail. I could not find one locally, however. Happy Holidays Bob... Doug Crompton wrote: After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have phone company compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk and my SPA-3000. When there is a message waiting both the LCD display and a flashing indicator on the phone alert you. This is true for all extensions on the system, up to 8. These work with both FSK and Stutter tone. I did not turn on the tone MWI as the FSK worked fine. KX-TG5776S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with a 1.5 Full-Color (65k Color Capable) Backlit LCD on Handset $119.95 KX-TG5771S 5.8 GHz FHSS GigaRange. Expandable Digital Cordless Answering System with Talking Caller ID $99.95 KX-TG5622M 5.8 GHz FHSS GigaRange Dual-Handset Phone System $89.95 KX-TG5761S 5.8 GHz FHSS GigaRange Expandable Digital Cordless Phone with Talking Caller ID $89.95 In order for the external MWI to work you must turn on the message indicator and for units that have answering machines the machine must be turned off. Perhaps we could put together a list of analog phones that have this feature. I have been told that both Uniden and ATT have models that work but I have no knowledge of all that do in their entire line. Each brand has their own features and while the Panasonic is solid - I had a 2.4G system for years and really liked it - the Unidens seems to have more for the money but in this case not MWI. I guess you could tell I really wanted this MWI to work! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones A-B-C-D
The free version 1.31 has all 16 keys on the keypad. Bob... Al Bochter wrote: Are you sure IdsFisk will do all 16 DTMF tones? I have the free ver of IdeFisk and that one does only the base 12 DTMF tones Base 12 DTMF are 1 2 3 4 5 6 7 8 9 0 # * The 16 DTMF are 1 2 3 4 5 6 7 8 9 0 # * A B C D If the paid ver of IdeFisk has that may have to pay the money but first I must know for sure. :-) I want to use A B C D for control IVR's Not Everyone knows about the 16 tones like the Hams do 8-) Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Zoa wrote: Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it). Zoa Al Bochter wrote: Ok does anyone know of any softphones that will dial DTMF tone keys A B C D And do you know if Asterisk will take the DTMF Tones for A B C D ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 11:16:05 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using SIP with NAT (technical code question)
You might want to pass that one by the asterisk-dev list. Bob... On Mon, 2006-12-11 at 15:18 -0800, je . wrote: My mistake, I misread it. So if a hostname is provided (e.g. [EMAIL PROTECTED]) instead of an IP (e.g. 123.123.123.123) and the recipient of the INVITE is not using NAT then ast_gethostbyname will be run - is that correct? In this case, why the distinction between a NATted and non_NATted implementation? --- Bob Chiodini [EMAIL PROTECTED] wrote: It looks to me that if the test clause is false then ast_gethostbyname is called. Presumably not needed when NAT is enabled. Bob... je . wrote: In chan_sip.c, line 5876 (Asterisk-1.2.13), the function parse_ok_contact returns whether the host that requested an invite is a valid or invalid host. In line 5925 the following clause is tested: if (!(ast_test_flag(pvt, SIP_NAT) SIP_NAT_ROUTE)) hp = ast_gethostbyname(n, ahp); If this clause is true then Asterisk will attempt to retrieve the IP address by using the hostname provided in the invite. My question is, is this test always going to be true if a user (who receives the invite) uses NAT? (this is set up in sip.conf as nat=yes) Is there a reason why this was set up only for NAT? Thanks, Jez __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Power requirements on the TDM-400 card
Gustavo, Take a look at this thread http://lists.digium.com/pipermail/asterisk-users/2006-October/169627.html Presumably the supplemental 12v supply is for ringing voltage. I did not see anything on Digium's support pages about the card itself. Maybe a call to tech support may help. Bob... On Mon, 2006-12-11 at 13:09 +, Gustavo Felisberto wrote: I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power will this drain from the 12 and 5 V connector when all ports are in use? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using SIP with NAT (technical code question)
It looks to me that if the test clause is false then ast_gethostbyname is called. Presumably not needed when NAT is enabled. Bob... je . wrote: In chan_sip.c, line 5876 (Asterisk-1.2.13), the function parse_ok_contact returns whether the host that requested an invite is a valid or invalid host. In line 5925 the following clause is tested: if (!(ast_test_flag(pvt, SIP_NAT) SIP_NAT_ROUTE)) hp = ast_gethostbyname(n, ahp); If this clause is true then Asterisk will attempt to retrieve the IP address by using the hostname provided in the invite. My question is, is this test always going to be true if a user (who receives the invite) uses NAT? (this is set up in sip.conf as nat=yes) Is there a reason why this was set up only for NAT? Thanks, Jez __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
Doug, The Uniden CLX465 supports stutter dial tone (SDT) and provides a MWI. Might be overkill since it is an answering machine as well. There are a few others. Google for stutter dial tone or phone company compatible voice mail. The SPA3K can produce SDT. The Budgetone 102 also has an MWI. I never thought about painting the phone's case. The handset might be an issue. Sounds like an interesting marketing opportunity, like cell-phone covers. Bob... On Thu, 2006-12-07 at 10:14 -0500, Doug Crompton wrote: John, Two questions on your comments I have no seen an Insteon computer controller similiar to the old bottle rocket. Is there such a device? I am thinking of getting an Insteon starter kit bit I have so many X10 devices it will be awhie before, if ever, that I get it all changed over. Many items, like spotlights, are not available in Insteon. I would be interested in the Ethernet MWI. I am using many phones on an SPA3000 fxs and I can't seem to find an MWI on an analog phone that works with Asterisk and the SPA3000, although I have been told that there are some that do??? The quick answer would be to put a SIP phone with MWI where your wife wants to be able to see the light. I have a Budgtone 200 and MWI works fine on it. Of course then you have styling and color issues that might not past the muster. Doug On Thu, 7 Dec 2006, John Marvin wrote: I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk freezes when DNS not working: a BUG??
Giorgio, You could set up a caching name server in your local network, use it as your primary DNS server and your ISP's as a secondary. This would cache your ITSP's address(es) locally limiting your reliance on your ISP. Bob... On Wed, 2006-12-06 at 10:43 +0100, Giorgio Incantalupo wrote: Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP so I cannot use it. Is there anybody who had a problem like this? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3
Eric, It looks like the definition for PTHREAD_MUTEX_RECURSIVE is within an #ifdef __USE_UNIX98 (on Fedora Core 6, anyway). You could try defining it within the Makefile. Similar to the _GNU_SOURCE definition in the app_cepstral.so: app_cepstral.c stanza. Bob... On Wed, 2006-11-29 at 13:38 -0500, Earle Clubb wrote: Hall, Eric M. wrote: Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks Eric, I had similar compilation issues when trying to use app_cepstral. This doesn't answer your question, but I've had good success using app_swift. http://www.loopfree.net/app_swift/ Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
Larry, There is later firmware 3.1.10 dated March 2006. I gave up on the SPA3K. I could not solve the echo problems. Rich Adamson indicated that the SPA3K did not have logic to fall back to the PSTN on SIP failure, only loss of link to the network. I would have thought that Sipura could have used a registration failure as a SIP failure. The registration time would need to be relatively short to detect a failure in a reasonable amount of time. To test registration failover to PSTN, change the password for the SPA3K on the asterisk side. That should cause a registration failure. Then try placing a call to the PSTN line from elsewhere (a cell). Try calling out as well. Neither Linksys nor Sipura seem to have updated manuals. Try the latest firmware, but get a copy of the current, just in case. Bob... On Wed, 2006-11-15 at 23:22 -0600, Larry Alkoff wrote: Bob I have a further question about Fallback: On my Line 1 tab, the last item is VoIP Fallback To PSTN but there is no setting that can be changed. I _think_ my firmware is the latest when I bought the unit Oct 25, 2005 but _possibly_ I have no fallback. Bummer. Could you comment on this please? BTW, section 4.11 talks only about pstn calls ringing line 1: The voice path is (7) (6) (4) (2) (1). This feature is enabled by setting PSTN Ring Thru Line 1 to “yes”. If enabled, all incoming PSTN calls will ring the Line 1 phone regardless the VoIP gateway function is enabled on the SPA or not. Hence the same phone can be used to receive calls from Line 1 VoIP and from the PSTN. Section 4.9 talks about Fallback to PSTN but I'm not sure how to test this with my setup or even if fallback is implemented in my SPA3k: 4.9. Line 1 VoIP Fallback to PSTN When power is removed from the SPA-3000, the FXS port will be connected to the FXO port. In this case, the telephone attached to the FXS port is electrically connected to the PSTN service via the FXO port. When power is applied to the SPA, the FXS port will be disconnected from the FXO port. However, if the PSTN line is in use when the power is applied to the SPA, the relay will not be flipped until the PSTN line is released. This is done so that the SPA will not interrupt any call in progress on the PSTN line. When Line 1 VoIP service is down (due to registration failure or loss of Ethernet link), SPA can be configured to automatically route all outbound calls to the internal gateway if Auto PSTN Fallback ([Line 1] tab) is set to “yes”. The PSTN gateway applies the Line 1 Fallback DP to further limit the calls that can be made by the Line 1 caller during the fallback operation; this dial plan may be set to “none”. This case also belongs to call type #7 and the voice path is (1) (2) (4) (6) (7). Of course, I'm having a lot of trouble reading this complex manual g Larry Bob Chiodini wrote: Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSl and more then 1 call
Kelly, Could there be a mismatch at the branch switch? Such as ethernet interfaces operating at half-duplex when the switch is at full-duplex. This usually manifests itself as dropped packets. I have an older Dell box that cannot seem to negotiate with a Cisco switch. 50% of the time it comes up in half-duplex while the switch is at full. Bob... On Mon, 2006-11-13 at 15:03 -0500, Kelly Opal wrote: Hi It's defiantly the branch server. My main server handles 30 to 40 calls at a time on a regular basis. It is only happening on the branch server and it acts like it is using up all the bandwidth of the DSL. It is a 1.5 meg down and 512 up DSL line. I would think it could handle 2 simultaneous calls. I have tried using g729, ulaw, alaw and gsm. There is no difference in the behavior. Could it possible be a routing issue on the LAN side of server 2. Kelly - Original Message - From: Vicky To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 13, 2006 1:59 PM Subject: Re: [asterisk-users] DSl and more then 1 call Does it happen when you make more than one call from you main voip server alone ? Or it happens when there are more than 1 call on your branch server ? Pin the problem is in which server first , If main server can handle 2-3 calls with no lag then its probably problem in branch server . On 13/11/06, Kelly Opal [EMAIL PROTECTED] wrote: Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls going at the same time the ping times go up to 2500 and above and the call quality is horrible. If I add a third call the system becomes unusable. But if you hang up all calls except 1 (it doesn't matter which one) it works fine again. Any help you could provide would be greatly appreciated. Kelly ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] config template for Grandstreams
It appears to be up there now. From the header: ## Configuration template for GXP-2000 firmware version 1.1.1.14 Bob... On Tue, 2006-11-14 at 13:45 -0500, Todd- Asterisk wrote: Thanks- they did respond. I got a new template, but was asked to not share it for now - it'll be on their website in a few days pending committee approval thanks Todd On Nov 14, 2006, at 12:50 PM, Gordon Henderson wrote: On Fri, 10 Nov 2006, Todd- Asterisk wrote: I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate the config file from a template, but the templates posted on their website are for an old version of the phone firmware. Anyone have a tool or access to templates for the latest firmware versions? Email their technical support. I did this a few days ago for the latest one for the GPX2000 and they emailled it back the next day. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bellsouth issue ?
Dovid B wrote: I have a client that has a dedicated box. Running asterisk 1.2.10 with ztdummy on Centos. He is connected via Bell South DSL in the Miami, Florida area. He has been complaing about voice quality issues. The person he is calling can hear him fine however he can not has terrible quality issues. Is there anyone else that is having problems with Bell South ? (Qos has been taken care of). Thanks. Dovid Dovid, I am a residential Bellsouth DSL customer in central Florida and have not had an quality issues with or without Zaptel hardware. I typically get 380KB/second down and supposedly 48KB/second up. I tested this past weekend running CentOS on a 700 MHz athlon with a bittorrent of Fedora FC6 running on another PC in the network, no sound quality issues. What DSL service level does your client have? Also how was QoS taken care of? Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Latest Debian and latest zaptel
Christian, Could you be out of disk space? What is the output of df -k and mount. Also does /root/zaptel-1.4.0-beta2/tonezone.h exist? Assuming the source is at that directory level. Bob... Christian wrote: Hi, No, I still get that error as before. And I havent installed anything special. Many thanks, Christian On 2006-11-10 at 23:08 brandon kruz wrote: svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel cd zaptel make clean ; make distclean ; sh configure ; make ; make install modprobe ztdummy see if those give you any errors From: Christian [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re[5]: [asterisk-users] Latest Debian and latest zaptel Date: Sat, 11 Nov 2006 04:40:26 +0100 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc11-f6.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri, 10 Nov 2006 19:42:14 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id E822A2FC431;Fri, 10 Nov 2006 20:40:10 -0700 (MST) Received: from psmtp.com (exprod8mx48.postini.com [64.18.3.148])by lists.digium.com (Postfix) with SMTP id 836352FC427for asterisk-users@lists.digium.com;Fri, 10 Nov 2006 20:40:06 -0700 (MST) Received: from source ([81.216.65.12]) by exprod8mx48.postini.com([64.18.7.10]) with SMTP; Fri, 10 Nov 2006 19:40:25 PST Received: from localhost (localhost [127.0.0.1])by iggypop2.siwnet.net (Postfix) with SMTP id EE8A9205886for asterisk-users@lists.digium.com;Sat, 11 Nov 2006 04:40:22 +0100 (CET) Received: from iggypop2.siwnet.net (localhost [127.0.0.1])by iggypop2.siwnet.net (Postfix) with ESMTP id 7D2DC205881for asterisk-users@lists.digium.com;Sat, 11 Nov 2006 04:40:22 +0100 (CET) Received: from u (231.142.216.81.static.tab.siw.siwnet.net [81.216.142.231])(Authenticated sender: [EMAIL PROTECTED])by iggypop2.siwnet.net (Postfix) with ESMTP id 5006F20584Dfor asterisk-users@lists.digium.com;Sat, 11 Nov 2006 04:40:22 +0100 (CET) X-Message-Info: LsUYwwHHNt3GES/zhXTpUM1zfXR0iU5Qwpl6IbFbp8M= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com References: [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED] X-Mailer: Courier 3.50.00.09.1098 (http://www.rosecitysoftware.com) (P) X-Virus-Scanned: ClamAV using ClamSMTP X-DSPAM-Result: Spam X-DSPAM-Processed: Sat Nov 11 04:40:22 2006 X-DSPAM-Confidence: 0.6060 X-DSPAM-Probability: 1. X-DSPAM-Signature: 45554626975671671310483 X-DSPAM-Factors: 15, Content-Transfer-Encoding*quoted+printable, 0.99000,Date*2006+04, 0.99000, Received*Nov+2006, 0.99000,Content-Transfer-Encoding*printable, 0.99000,Content-Transfer-Encoding*quoted, 0.99000,From*Christian [EMAIL PROTECTED], 0.99000,Date*Nov+2006, 0.99000, Hi, 0.99000,Received*2006+04, 0.99000, both, 0.12167, what, 0.16866,com, 0.17363, up, 0.18386, Received*11, 0.19184, so, 0.19824 X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 11 Nov 2006 03:42:15.0633 (UTC) FILETIME=[6146DC10:01C70543] Hi, Well, I managed to find out what svn was and I also downloaded the latest 1.4 version of zaptel and Asterisk. svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 But the problem is still there. Cannot install Zaptel. Asterisk and libpri installs just fine. Want to use ztdummy. Soon giving up on this one! Many thanks, Christian On 2006-11-10 at 23:51 Michiel van Baak wrote: On Nov 10, 2006, at 11:12 PM, Christian wrote: Hi, But what is the problem, why doesnt it install? I am a little new to this so still learning. Many thanks, Christian Use the latest 1.4 svn version instead of beta2 That will probably fix your problem. I installed latest 1.4 svn checkout on debian etch 2 times today while playing with xen so I can confirm it's working great now. --- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and
Re: [asterisk-users] Latest Debian and latest zaptel
Christian, Either mkdir -p /usr/include/zaptel/tonezone or delete the tonezone.h link then re-run the build. The date stamp on the link does not correspond with the others the directory. Could be something leftover from an earlier attempt? Bob... Christian wrote: Hi, See my answers below. On 2006-11-11 at 19:15 Tzafrir Cohen wrote: I repeat: please give the output of: ls -la tonezone.h /usr/include/zaptel /usr/include/zaptel: total 72 drwxr-xr-x 2 root root 4096 2006-11-11 17:18 . drwxr-xr-x 101 root root 8192 2006-11-11 04:30 .. lrwxrwxrwx 1 root root22 2006-11-10 22:09 tonezone.h - ../tonezone/tonezone.h -rw-r--r-- 1 root root 53739 2006-11-11 17:18 zaptel.h What happens if you run that command again: install -D -m 644 tonezone.h /usr/include/zaptel install: cannot create regular file `/usr/include/zaptel/tonezone.h': No such file or directory If there is a problem with that: try: touch /usr/include/zaptel/tonezone.h echo /usr/include/zaptel/tonezone.h The file could not be found. Yes, it is there but to me it looks very small. Many thanks, Christian -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1862 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why dont my messages get through
There is an option on the list server membership configuration screen that will disable receiving your own posts to the list. Maybe the OP accidentally disabled this feature. Bob... On Wed, 2006-11-08 at 06:09 +0200, Dovid B wrote: I have seen this mainly with gmail. the logic is why do you need your own postings. Fish around to see if there is a setting in Gmail where it will keep the email. I know for myself I want the email's that I sent. It lets me know that they went out as well as it helps for sorting the emails. - Original Message - From: Alex Robar To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 08, 2006 5:08 AM Subject: Re: [asterisk-users] Why dont my messages get through They do get through. Messages you send to the list won't get sent back to you, because you sent them. On 11/7/06, Christian [EMAIL PROTECTED] wrote: Hi, My messages to the list don't get through. This must be the tenth message i am trying to send! Please ignore this test message. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3k wired to PAP2 for echo testing
I forgot about the levels. One thing that really helps is lowering the to PSTN gain. Unfortunately, you may find that to reduce echo to a tolerable level you will need to reduce the gain so low that the called party may have a hard time hearing you. I think I dropped the gain values by 4 dB. Bob... On Sun, 2006-11-05 at 21:55 -0500, Doug Crompton wrote: Yes I agree, the SPA3000 can be a bear with echo on the PSTN. I did find that using older fimware helped some and that the levels - there are 4 settings - FXO/FXS in/out can be juggled to help. I also found out after adding a Budgetone 200 that I had much less echo problem going through it and the spa3000 FXO - vs. using the local analog phones on the spa3000 fxs port to FXO port. So some of the answer might be to get rid of as much (or all) local analog as you can. I plan to buy more hard sip phones and do that here eventually. This is ultimately more flexible as each extension has it's own number and they can dial each other as well as dial more then one place simutaneously. The big problem is that SIP phones are generally ugly and black and not styled for home use. Doug On Sun, 5 Nov 2006, James Harper wrote: In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset -- (FXS)SPA3000(FXO) -- PAP2 And set the Line1 dialplan on the SPA3k to '(:@gw0S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with my impedance settings set to the Australian standard of 220+820||120nf, and the PSTN and PAP2 echo cancellers enabled (or not, and all combinations of) I get local echo as soon as I pick up the handset (I hear my voice bounced back to me). Surely this shouldn't be??? There is no hybrid involved at all! If anyone on this list with a SPA3k (that doesn't have any local echo problems on the PSTN port) and an ATA with a FXS port, could they please try the above setup and post the results (including SPA3k hardware and firmware versions, and the ATA used)? I wonder if there is a problem with some versions of the SPA3k where there is some sort of inbalance on the PSTN port that causes echo right there rather than further down the line? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible have multiple ip numbers for an extension?
On Mon, 2006-11-06 at 21:31 +0800, William Kenworthy wrote: Is it possible have multiple concurrent ip numbers for a single extension? How? I am using a laptop that I move around various local and remote networks so the IP numbers it uses varies. As I am on extension '205', I want be found wherever I have a connection by anyone dialling 205. Using an openvpn tunnel works well, but there are a couple of networks I dont want to tunnel (because it would mean tunneling a tunnel trough a tunnel(!) as they are already secure. Using asterisk 1.2.13 on gentoo BillK Bill, Does it matter? It appears that only the extension and password (secret) are necessary to register. Have you tried varying the IP? Firewall and router issues will still exist, however. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3k wired to PAP2 for echo testing
I'm in the US and had bad echo problems with the SPA3K and the latest firmware. I was under the impression that the echo was due to my long cable run to the CO ~15000'. Changing the impedance (900 ohms) would help for a while, but after a few days the echo came back. If I rebooted the SPA3K the echo would go away for a while, but always came back. Assuming a software problem, I back-revved to an earlier F/W version. This seemed to help, but was not a cure. It did not pass muster with my wife. BTW: I did not experience echo on SIP calls through my ITSP or locally w/in my network. I've seen some chatter about a Global option helping, but never tried it. I gave up and switched to a TDM11B. There was also some talk about having the earpiece volume up too high such that the phone's microphone picked up the sidetone and caused echo. I did have better results when the phone's volume was turned down, but I the SPA3k echo problem was never cured Bob... James Harper wrote: In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset -- (FXS)SPA3000(FXO) -- PAP2 And set the Line1 dialplan on the SPA3k to '(:@gw0S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with my impedance settings set to the Australian standard of 220+820||120nf, and the PSTN and PAP2 echo cancellers enabled (or not, and all combinations of) I get local echo as soon as I pick up the handset (I hear my voice bounced back to me). Surely this shouldn't be??? There is no hybrid involved at all! If anyone on this list with a SPA3k (that doesn't have any local echo problems on the PSTN port) and an ATA with a FXS port, could they please try the above setup and post the results (including SPA3k hardware and firmware versions, and the ATA used)? I wonder if there is a problem with some versions of the SPA3k where there is some sort of inbalance on the PSTN port that causes echo right there rather than further down the line? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAPtel channel dance
On Thu, 2006-11-02 at 11:32 +0200, Tzafrir Cohen wrote:
On Thu, Nov 02, 2006 at 10:10:01AM +0100, Florian Hars wrote:
Zaptel installs an /etc/modprobe.d/zaptel and an
/etc/{defaults,sysconfig}/zaptel that list the modules in a different
order, so If you happen to have a TDM2400P and a TDM[124]xxP, all channels
change their numbers if you do a /etc/initd/zaptel restart. This is
slightly confusing.
The order of the channels is the order in which the spans register to
Zaptel, which is basically the order in which the modules load.
On Debian, load the modules through /etc/modules . Otherwise they will
be loaded through hotplug/udev in an unpredictable order (by the order
of PCI slots) which may or may not be the order that you like.
Gentoo has an equivalent file, whose name I forgot.
Redhats seem to lack such a mechanism, and I'm not sure whther or not
those cards do get hotplugged/coldplugged. Thus the tsrange need to load
them in the zaptel startup script.
Anyway, the order in which you happened to load them right now is not
guaranteed to be the order in which you load them next time unless you
explicitly
(I'd file a bug if there were a bug tracking system
that allowed users to submit bugs).
Users are surely allowed. Just register.
Also, bug reports to xpp/genzaptelconf are welcomed. It should be able
to write such module loading lists that should provide predictable order
in both Debian and Redhats.
For Redhat, Fedora, CentOS and other derivatives:
You can play tricks in /etc/modprobe.conf using the install directive.
The man page for modprobe.conf gives an example.
You could also force their loading and presumably their order in initrd
or rc.modules which runs as part of rc.sysinit.
rc.modules is the cleanest approach (IMHO), as initrd gets rebuilt by
some updates (e.g. kernel).
Bob...
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Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
Switching is what you want. NAT is Network Address Translation that allows the router to map IP addresses between router interfaces. You may wish to verify that all of the ports on your network, if automatically negotiated, did what you want. Probably, 100Mb, Full-Duplex. If not then force the config. Improper negotiation tends to drop packets. Everything appears to work, but slowly. Depending on your network infrastructure, you may also look into QOS. Bob... On Wed, 2006-11-01 at 16:15 -0500, Zeeshan Zakaria wrote: All the phones already have the latest firmware. They keep updating themselves automatically. In my setup of Grandstream phones, all the computers of the network go through the phones, i.e. I am using the builtin phones as swithces. They all have 2 ethernet ports. Does this has to do anything with the voice quality, or do I need to change something in the phones' setup, like switching it from switch to router in basic settings? What is this NAT/Router setting anyways and how should it be setup? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAPtel channel dance
Tzafrir Cohen wrote: On Thu, Nov 02, 2006 at 06:34:03AM -0500, Bob Chiodini wrote: For Redhat, Fedora, CentOS and other derivatives: You can play tricks in /etc/modprobe.conf using the install directive. The man page for modprobe.conf gives an example. This is not the proper place: those are not real dependencies. You may actually want to load those modules separately one day. Probably not, but from what I've seen on some of the Fedora lists it's preferred over the rc.modules hack to force module loading. You could also force their loading and presumably their order in initrd or rc.modules which runs as part of rc.sysinit. Hmmm... sounds nice, however the text I read there is: # Load modules (for backward compatibility with VARs) if [ -f /etc/rc.modules ]; then /etc/rc.modules fi Is it guranateed to remain there? rc.modules has been there a long time... Whether it will stay, who knows. I have not seen anything about deprecation and it should not get touched by any upgrades. rc.modules is the cleanest approach (IMHO), as initrd gets rebuilt by some updates (e.g. kernel). And can't easily be re-run. Given Florian's latest post the problem seems to lie in the zaptel modules' build and/or install mechanisms. I have not dug into the Makefiles, but I would think the result of genmodconf and zaptel.sysconfig should be consistent. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Electric usage of a tdm400p
Erick, It looks like the 2.5 laptop drive requires 5 watts to spin up. Adding that to the 15 watts for the Digium card, leaves about 40 watts available for the MB. It's unlikely that the system will be producing ring voltages when the drive is spinning up. It depends on how conservative you may be with real-time power management, e.g. spinning drives down when not in use, etc. I did not easily find too many ITX MB power requirements, but the one I did find, required 45 watts (peak). In the worst case, ringing 5 RENs on each Digium port and spinning up the disk, you would be overtaxing the power supply. I doubt you will have 5 RENs on each port and all ringing, but you could. In ages past, hard drives were the most vulnerable to poor power regulation, but that may have changed. With the higher cost of 2.5 drives, I would not take any chances. Beefing up the power supply would also eliminate the need for manually managing power should you need a CDROM or more power hungry drive in the future. It's also one less concern when troubleshooting the system. As Moj has pointed out, problems can occur when working close to the edge. I, too, have experienced similar problems when power was limited and have had to, temporarily, resort to a bigger power supply to get a system installed. Then fell back to a smaller one in operation. Good luck. Bob... On Wed, 2006-10-18 at 08:49 -0800, Mojo with Horan Company, LLC wrote: I set up a similar system on an VIA Epia 5000, and I had issues when I included the CDROM in the mix. I had to use another ATX power supply to complete the install, but then once I removed the CDROM drive I had no power issues. I presume you could install the OS with the CDROM drive installed and the molex power connector REMOVED from the TDM card, then when the OS was installed and you had network connectivity, power down, remove the CDROM, add the power supply for the TDM card, then install zaptel etc. Or just try it and tell us what happens, low power won't break it in my experience. Your cdrom drive might have a lower power consumption than mine. Moj Erick Perez wrote: Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card. the mini-itx comes with a 60W DC to DC adapter (80W peak). So I need power to manage the hdd, motherboard,the tdm card. A disk cable can be made available, but is not present as a factory default. So My real concern is power. On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote: On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Electric usage of a tdm400p
On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a Voicemail Lamp device
Tom, The uniden TRU446 and the CLX465 both are supposed to detect stutter dial tone (SDT) from the phone company and light the MWI. When used with asterisk the SPA3000 can generate SDT. I'm not sure it can do so on its own. I gave up on the SPA 3000 due to echo problems. http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCU http://www0.epinions.com/content_70491344516 Hope that helps, a little. Bob... Tom Lynn wrote: I'm looking for an external device that can flash when there is new voicemail in a mailbox. I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system. Problem is, the Uniden system has it's own answering machine, which I don't want to use. But the message lamps are driven solely by the internal answering machine function. Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a Voicemail Lamp device
Tom, There are a couple of SIP based cordless phones out there. A little pricey, however. Such as: http://www.voipsupply.com/product_info.php?manufacturers_id=35products_id=923 It might be compatible with your existing cordless hand sets. Uniden seems pretty good about that. Or: http://www.voipsupply.com/product_info.php?manufacturers_id=40products_id=1007 Bob... Tom Lynn wrote: I can get stutter dialtone using my spa3000, but the uniden doesn't respond to it by lighting the lamp. All it sees is an incoming call from the spa. It looks to me that I'll either need an external MWI device or I'm going to have to replace the Uniden phones. On 10/14/06, *Tom Lynn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: My uniden phone is the TRU8885-3HS. On 10/14/06, *Bob Chiodini* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Tom, The uniden TRU446 and the CLX465 both are supposed to detect stutter dial tone (SDT) from the phone company and light the MWI. When used with asterisk the SPA3000 can generate SDT. I'm not sure it can do so on its own. I gave up on the SPA 3000 due to echo problems. http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCU http://www0.epinions.com/content_70491344516 http://www0.epinions.com/content_70491344516 Hope that helps, a little. Bob... Tom Lynn wrote: I'm looking for an external device that can flash when there is new voicemail in a mailbox. I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system. Problem is, the Uniden system has it's own answering machine, which I don't want to use. But the message lamps are driven solely by the internal answering machine function. Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?
On Thu, 2006-10-12 at 10:20 -0400, Jay R. Ashworth wrote: On Wed, Oct 11, 2006 at 12:07:50PM -0400, Bob Chiodini wrote: We had a power failure that took down the internet connection and local DNS server. My local Cisco phones could not register (IP addresses are hard-coded) and, because of the DNS failure I could not register with my SIP provider. I have not had a chance to sort through the logs, but I had to reset the Asterisk box, after the DNS server was restored. In my case, inbound and outbound PSTN calls (via a TDM11b) were failing. The local analog phone rang (on an inbound PSTN call), but did not recognize the analog answering machine taking the line off-hook. Once the caller hung up, the local (analog) phones would ring again, but no call was present, as reported by my wife. BTW: The Asterisk box is on UPS and did not go down and I do not have voicemail enabled for my local extensions. As a general rule, if you aren't already, you should have your Linux box running a local DNS server, to which everything in your net should be pointed, and that server *should have an authoritative zone for your local RFC 1918 network number, in both directions*. If it does not, then those reverse lookups that many programs generate, in trying to log names for connections instead of numbers, will go to the outside world before they bounce... or they'll time out when your uplink is dead. This may be (part of) your problem. Cheers, -- jra Jay, I agree. My caching nameserver runs on another Linux box in the network, that failed to reboot after yesterdays outage, due to my error. We had another power failure this morning :-( Florida Plunder and Loot strikes again! No adverse affects, this time. Heading to Wally World after work for a couple of UPS units. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?
On Wed, 2006-10-11 at 11:19 -0400, Jerry Geis wrote: I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said Registration error. Why would phones loose registration to asterisk when the internet connection and DNS was lost. All phones have hardcoded IP addresses not DNS names. Any ideas? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ditto here. I run Trixbox 1.1 with the latest updates. We had a power failure that took down the internet connection and local DNS server. My local Cisco phones could not register (IP addresses are hard-coded) and, because of the DNS failure I could not register with my SIP provider. I have not had a chance to sort through the logs, but I had to reset the Asterisk box, after the DNS server was restored. In my case, inbound and outbound PSTN calls (via a TDM11b) were failing. The local analog phone rang (on an inbound PSTN call), but did not recognize the analog answering machine taking the line off-hook. Once the caller hung up, the local (analog) phones would ring again, but no call was present, as reported by my wife. BTW: The Asterisk box is on UPS and did not go down and I do not have voicemail enabled for my local extensions. This sounds similar, possibly: http://lists.digium.com/pipermail/asterisk-users/2006-October/168910.html Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no callerid from PSTN using TDM2400P
On Wed, 2006-10-04 at 14:58 -0700, Naija Man wrote: Hello all, Asterisk 1.2.8 zaptel 1.2.6 Hardware: digium TDM2422P I have a fully configured asterisk system with POTS line for PSTN access. I am not receiving the callerid for incoming calls from the PSTN. I get the following error message. -- Starting simple switch on 'Zap/3-1' Oct 3 22:53:14 NOTICE[17948]: chan_zap.c:6061 ss_thread: Got event 18 (Ring Begin)... Oct 3 22:53:16 ERROR[17948]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-22) Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6091 ss_thread: CallerID feed failed: Success Oct 3 22:53:16 WARNING[17948]: chan_zap.c:6135 ss_thread: CallerID returned with error on channel 'Zap/3-1 I've been seeing similar problems, but they are intermittent. I'm in the US. It seems that restarting asterisk clears up the problem for a while, but it may be only coincidence. I'm running Trixbox 1.1.0, Asterisk 1.2.12.1, Zaptel 1.2.9.1. I did not have a problem with [EMAIL PROTECTED] v 2.8. H/W TDM11B. Still looking... Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID not getting passed?
On Mon, 2006-09-11 at 20:25 -0700, Christopher Corn wrote:
im having issues when routing calls from the outside with my new VSP.
this is what asterisk tells me when i try to make an incoming call, i
get the no service response when i call.
-- Executing GotoIf(SIP/christopher_corn-eddb, 1?from-trunk||
1) in new stack
-- Goto (from-trunk,s,1)
-- Executing NoOp(SIP/christopher_corn-eddb, No DID or CID
Match) in new stack
-- Executing Answer(SIP/christopher_corn-eddb, ) in new stack
-- Executing Wait(SIP/christopher_corn-eddb, 2) in new stack
-- Executing Playback(SIP/christopher_corn-eddb, ss-noservice)
in new stack
-- Playing 'ss-noservice' (language 'en')
my extensions additional.conf has this
[ext-did]
include = ext-did-custom
exten = 408335,1,Set(FROM_DID=4083354290)
exten = 408335,n,Goto(ext-local,103,1)
exten = s,1,Noop(No DID or CID Match)
exten = s,n,Answer
exten = s,n,Wait(2)
exten = s,n,Playback(ss-noservice)
exten = s,n,SayAlpha(${FROM_DID})
exten = _[*#X].,1,Set(FROM_DID=${EXTEN})
exten = _[*#X].,n,Noop(Received an unknown call with DID set to
${EXTEN})
exten = _[*#X].,n,Goto(ext-did,s,1)
; end of [ext-did]
i tried to replacing my number with my username, my phone number
without area code, using dashes, but nothing works.
is it because my vsp, axvoice.com doesn't pass did's?
any information is appreciated. thanks.
Try turning on SIP debug to see what you are getting from your provider.
Bob...
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Re: [asterisk-users] Asterisk and NAT ?
Does the Linksys know it should be using port 5070? It would seem to me that port forwarding would be required as the phones are behind a NAT'd firewall. How would asterisk know how to get there since it's not on the same subnet (outside the firewall). If the asterisk box has physical access to the NAT'd subnet, you could put in a second NIC connected behind the firewall. Bob... On Thu, 2006-09-07 at 23:57 -0400, William Piper wrote: Does the phone have stun settings? If so, try using stun.fwdnet.net and take out the port forwards and see if it works. bp On 9/7/06, Noc Phibee [EMAIL PROTECTED] wrote: yusuf a écrit : Hi, you dont have to/should'nt be using different SIP ports for each phone. Its completely not needed. Also, you dont have/need to port forward. Just open ports 5060 and 1000-2, on the box that asterisk is running, and on your NAT router. Dont port forward. Then in sip.conf [202] username=202 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne nat=yes canreinvite=no [200] username=200 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne nat=yes canreinvite=no then restart linksys and thomson, and you will see that they both register on asterisk cli. Now you will be able to call/receive on both. Thanks for your answer, but if i don't put a port forward, i have : 200/20083.167.122.119 D N 5060 UNREACHABLE On the thomson, i have SIP Unregister, it's a important option ? Thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
Nick, Anything helpful in the asterisk or system logs. Try bumping up the debug and verbose levels see what shows up on the console. Weird that it would work inbound and not outbound. Bob... On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote: Hey all, A previous annoyance with not being able to call out to my brother on FWD from my Asterisk system had me thinking that since I have my own PBX, and that system has it's own 1-to-1 static NAT to the internet, I should be able to act as the provider for him or any of my family, and have them as local extensions of my PBX, right? So I took my laptop to work (using the X-Lite SIP softphone) and watch my ACL logs on my router for any denies to my Asterisk box. As expected udp/5060, then once that was open, a series of randomish udp/1+ requests. My phone registered, and I tried to call one of the phones behind a PAP2. Worked first shot, and just as clear and responsive as it was when I was home. But, the phones at home could not call me, they when to voice mail. I had heard that SIP doesn't survive NAT all that well, and that IAX native phones do a better job. My question is, given my description of how I am set up and what I am trying to accomplish, should I be looking at SIP or is IAX a more robust choice? (I was hoping to get video working as well, h.263 I believe it is). Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?
Rushowr wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey all, I'm looking into setting up a system or two with either IMAP or ODBC storage of Voicemail messages and wanted to hear about your experiences, gather tips or warnings, etc, before I go diving too deep into it. Are either of those storage methods working reliably for any of you? What are some of the issues you had to deal with when setting it up? What's the performance like? You get the general idea... Quick stats on base test systems: Latest SVN trunk as of this morning Gentoo MySQL 5.0 Realtime sip and iax peers/users Realtime sip/iax/voicemail config LARGE dialplan Thanks in advance for any input, SKM I've got a small asterisk (trixbox) system running at home as well as a Cyrus IMAP server handling the family's email, not on the same machine, but that is certainly possible. The IMAP server has been running for years on several different versions of RedHat and Fedora. I've had no major problems. Once in a while the DB that IMAP uses needs conversion when there is a major version change, but it's pretty simple. Cyrus IMAP does not use mbox format. Each email is saved in an individual file. For instance my asterisk-users folder has 11430 files. My personal opinion is that the access to individual files, indexed via a DB is faster than the mbox method of saving all of the mail in a single file. Client IMAP support is very good in Thuderbird and Evolution on the Linux side and outlook, outlook express and thunderbird on the windows side. Cyrus IMAP is a little finicky to set up, but once it's all running it's pretty smooth. dovecot is another IMAP server that supports both maildir and mbox formats. I don't know anything else about it, other than it's the default for Fedora. Hope that helps. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
Elpidio, Glad to hear it. Depending on your config, you may need to allow the RTP ports through as well. I poked holes in my firewall for ports 1-2, probably overkill, though. Bob... Elpidio Ramos wrote: This helped a lot. It was the firewall. I got it configured right now. Thanks */Bob Chiodini [EMAIL PROTECTED]/* wrote: I think all anywhere should allow 5060. Try running service iptables stop (as root) to shutdown the firewall. See if 5060 then answers. I'm not running a firewall on my asterisk box so I'm not sure what the rule would need to be. service iptables start will restore the firewall. Bob... Elpidio Ramos wrote: Bob, I get the same answer you get when using netstat -an When I query the firewall rules I get this: Chain RH-Firewall-1-INPUT (2 references) target prot opt source destination ACCEPT all -- anywhere anywhere ACCEPT icmp -- anywhere anywhere icmp any ACCEPT ipv6-crypt-- anywhere anywhere ACCEPT ipv6-auth-- anywhere anywhere ACCEPT udp -- anywhere 224.0.0.251 udp dpt:5353 ACCEPT udp -- anywhere anywhere udp dpt:ipp ACCEPT all -- anywhere anywhere state RELATED,ESTABLISHED ACCEPT tcp -- anywhere anywhere state NEW tcp dpt:ssh ACCEPT tcp -- anywhere anywhere state NEW tcp dpt:http REJECT all -- anywhere anywhere reject-with icmp-host-prohibited I assume this indicates port 5060 is restricted? Elpidio */Bob Chiodini /* wrote: Elpidio, Is it truly not listening or is maybe a firewall blocking port 5060. What does netstat -an | grep 5060 tell you? I get this: netstat -an | grep 5060 udp 0 0 0.0.0.0:5060 0.0.0.0:* iptables -L will list any firewall restrictions. Bob... On Fri, 2006-09-01 at 09:12 -0700, Elpidio Ramos wrote: Does anyone knows what could be the cause for asterisk not listening in post 5060 if SIP interfaces is loaded with no problems? I am using Fedora Core 3. I have followed the instructions in several tutorials and tried several soft phones and the SIP interface seem to be dead. 1. When loading asterisk SIP load with no problem 2. When I activate the DEBUG for a peer, ip or sip in general, I don't get to see any messages when a connection is attempted from any soft phone. 3. The soft phones all report a timeout when trying to register. 4. Tried also to move to port 80 that I know is open but still the same problem. I will appreciate any help anyone can provide with this problem. Elpidio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *Elpidio Ramos* President RM International Services SA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax +52 (55) 1755-6601 Cell USA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740 Direct *Elpidio Ramos* President RM International Services SA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax +52 (55) 1755-6601 Cell USA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740 Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Linksys PAP2 ATA
Nick, I've used a SPA3000. There seems to be a later model from Linksys, hopefully it works better. I had some severe echo problems due to my distance from the CO. The SPA3000 never could seem to compensate. The older software worked better, but it never passed muster with the wife. Went to a Digium TDM11B, no problems. There are plenty of mini-howtos on the web to set up a SPA3000. If you are close to your CO and the price is right, give it a try. I think I read somewhere that at up to 7000 feet between the CO and the SPA acceptable results are possible. I'm at about 18000 feet. Bob... Nick Ellson wrote: Hi Corey, I spend 2 hours with REALLY bad docs on how to Unlock the PAP2 Vonage I got from Office Max. I bought it the second I saw a glimpse of an articaly that it could be turned back into an NA. Anyone want to try this? The nes ones one the shelf in my area had 3.1.3 code already, but if you put together a few seperate How-To's ya can get a really simple procedure and the files to clean out Vonage and I can say my kids LOVE the new phones in their rooms to play house with ;) (So yeah, end result was actually 30 seconds and two reboots on the unit and it's Vonage Free and happily on my Asterisk network) So now I am looking at the Linksys SPA3000 to use as a poor-man's FXO port. It looks like this is an easier task doc's wise. Anyone set this up? Nick If anyone wants the final steps I used on the PAP2 lemme know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Linksys PAP2 ATA
Nick, I know some adults that can have an entire conversation in the same amount of time. Does pressing the # key speed up dialing? If so look for a timer in the PAP config or tell the kids to press #. IIRC the spa3k had something similar, but never did much in-house dialing. $86 is a pretty good price. I paid more than that for the spa3000 6 months ago. Bob... Nick Ellson wrote: Hey Bob, I think the SPA31-2 is the new guy on the block. Only $10 more too mail order. $86 was the best I saw. So I have the PAP2 with two cheapy $4 wall phones mounted in the kids room, they are calling each other and my laptop.. Only issue so far is that to call one PAP2 from the other there is a 10 sec delay before the ringback/ring occurs.. and a 3 5 year old can have an entire conversation before the phone even rings. ;) Calling from my X-Lite soft phone to the PAP2 is nearly instant. But it does have my wife actually jazzed about having two more phones where she works in the house so she can join the fun.. Score! A free pass to buy more toys! Another PAP2 and a SPA3102 for me Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sipura SPA3000
Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
Elpidio, Is it truly not listening or is maybe a firewall blocking port 5060. What does netstat -an | grep 5060 tell you? I get this: netstat -an | grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* iptables -L will list any firewall restrictions. Bob... On Fri, 2006-09-01 at 09:12 -0700, Elpidio Ramos wrote: Does anyone knows what could be the cause for asterisk not listening in post 5060 if SIP interfaces is loaded with no problems? I am using Fedora Core 3. I have followed the instructions in several tutorials and tried several soft phones and the SIP interface seem to be dead. 1. When loading asterisk SIP load with no problem 2. When I activate the DEBUG for a peer, ip or sip in general, I don't get to see any messages when a connection is attempted from any soft phone. 3. The soft phones all report a timeout when trying to register. 4. Tried also to move to port 80 that I know is open but still the same problem. I will appreciate any help anyone can provide with this problem. Elpidio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
I think all anywhere should allow 5060. Try running service iptables stop (as root) to shutdown the firewall. See if 5060 then answers. I'm not running a firewall on my asterisk box so I'm not sure what the rule would need to be. service iptables start will restore the firewall. Bob... Elpidio Ramos wrote: Bob, I get the same answer you get when using netstat -an When I query the firewall rules I get this: Chain RH-Firewall-1-INPUT (2 references) target prot opt source destination ACCEPT all -- anywhere anywhere ACCEPT icmp -- anywhere anywhereicmp any ACCEPT ipv6-crypt-- anywhere anywhere ACCEPT ipv6-auth-- anywhere anywhere ACCEPT udp -- anywhere 224.0.0.251 udp dpt:5353 ACCEPT udp -- anywhere anywhereudp dpt:ipp ACCEPT all -- anywhere anywherestate RELATED,ESTABLISHED ACCEPT tcp -- anywhere anywherestate NEW tcp dpt:ssh ACCEPT tcp -- anywhere anywherestate NEW tcp dpt:http REJECT all -- anywhere anywhere reject-with icmp-host-prohibited I assume this indicates port 5060 is restricted? Elpidio */Bob Chiodini [EMAIL PROTECTED]/* wrote: Elpidio, Is it truly not listening or is maybe a firewall blocking port 5060. What does netstat -an | grep 5060 tell you? I get this: netstat -an | grep 5060 udp 0 0 0.0.0.0:5060 0.0.0.0:* iptables -L will list any firewall restrictions. Bob... On Fri, 2006-09-01 at 09:12 -0700, Elpidio Ramos wrote: Does anyone knows what could be the cause for asterisk not listening in post 5060 if SIP interfaces is loaded with no problems? I am using Fedora Core 3. I have followed the instructions in several tutorials and tried several soft phones and the SIP interface seem to be dead. 1. When loading asterisk SIP load with no problem 2. When I activate the DEBUG for a peer, ip or sip in general, I don't get to see any messages when a connection is attempted from any soft phone. 3. The soft phones all report a timeout when trying to register. 4. Tried also to move to port 80 that I know is open but still the same problem. I will appreciate any help anyone can provide with this problem. Elpidio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *Elpidio Ramos* President RM International Services SA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax +52 (55) 1755-6601 Cell USA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740 Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does anyone offer truly unlimited voip in the US
Steve, VoiceEclipse has a US unlimited plan for $20/month. Two inbound numbers that can be in different area codes. I have not figured out how to recognize which number the inbound call came in on, but, right now, that is not that important to me. Others have had other problems. Research is recommended. Bob... Steven M. Sawczyn wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Thanks for any suggestions, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940 dialplan.xml
Francisco Gonzalez Canales wrote: I forgot to mention the firmware: P0S3-08-2-00 On 7/12/06 4:53 PM, Francisco Gonzalez Canales [EMAIL PROTECTED] wrote: Hello, For some reason it seems like the phone is not getting the file dialplan.xml file loaded from the TFTP server. Dialplan.xml file: DIALTEMPLATE TEMPLATE MATCH=* TIMEOUT=5/ /DIALTEMPLATE I have telneted to the phone and used: 'show dialplan' And I get the following output: Dialplan is Dialplan version: EMPTY Any ideas? Best, Francisco Francisco, Maybe case sensitivity. My dialplan.xml is: DIALTEMPLATE TEMPLATE MATCH=* Timeout=5/ !-- Anything else -- /DIALTEMPLATE Timeout is mixed case. The filename is all lower case. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk server crashes daily
Al Lougher wrote: Hi - This is the first Linux server I have ever built with an installation of [EMAIL PROTECTED] 2.7 mailto:[EMAIL PROTECTED]. For development I have been running on VMWare on an XP box and sustained no crashes or reboots. After moving Asterisk to it's own server I am experiencing daily crashses (around 4am) and I'm not quite sure what the problem is, nor am I sure where exactly to look for logs of any errors prior and during the crash. During the crash there should be nothing running so I'm not sure why it crashes at this time (perhaps some system job that is running at this time?). My hardware is: AMD Athlon 64bit 3200 CPU, 1 gig memory, 100gb hd and a gigabit NIC card. The BIOS is set with defaults. Many thanks, Al. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. http://us.rd.yahoo.com/evt=42297/*http://advision.webevents.yahoo.com/handraisers Al, Sounds like a possible memory issue. Memtest might help diagnose. You'll probably need a bootable CD such as the Fedora rescue CD. Otherwise, take a look in /var/log for the system logs. the main log is messages. The asterisk logs are in /var/log/asterisk. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with router setup on new asterisk box
Al Lougher wrote: Hi - I hope someone out there can help. I've just built a new asterisk server running [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 2.7. and I'm having real difficulty setting up my cable modem for the internet connection. I have 1xcable modem and 1xnetgear router and 1xPCI nic card. I simply set the netgear up as 192.168.0.1 (which is default anyway) and plug the cable from the modem into the internet/input port on the router. Then I connect any port on the router to the PCI card on the asterisk box. Next I run netconfig and set the gateway to 192.168.0.1. Reboot everything, but I cannot ping any internet site. I can though ping the router. Any ideas? Thanks Al. Al, Did you set an IP address on your asterisk box? You could also let it get its address via DHCP. Probably the router defaults to being a DHCP server. That has the advantage of getting the proper DNS servers from your ISP and the proper default route. You did not specify who you cannot ping or whether it's by name or IP address. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This is getting really annoying - re: POSTFIX
Matt wrote: What on earth is going on with the list?!?! Some of my messages never make it... then days later I get something like this back: Final-Recipient: rfc822; asterisk-users@lists.digium.com Action: failed Status: 5.0.0 Diagnostic-Code: X-Postfix; mail forwarding loop for asterisk-users@lists.digium.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users And I thought it was just me, or maybe gmail. I've seen very little traffic since last Wednesday or so. Bob... Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... Jerry Jones wrote: use an fxo interface and 600ohm input on amp On Jun 11, 2006, at 9:53 AM, Thomas Kenyon wrote: Doug Lytle wrote: Thomas Kenyon wrote: I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is Does the connection use 2 screws for analog inputs? Yup. If this is the case, you could get a cheap Grand Stream BT102 and pull the speaker leads off and connect it to that box. The GS can be setup to auto answer. Doug I'm going to try the suggestion in the Bat Phone thread above, bringing one of the PAP2s out of retirement. Wish me luck. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
Thomas Kenyon wrote: John Novack wrote: Bob Chiodini wrote: You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... FXO ports do not generate loop current, they detect loop current from the Central office. Think of an FXO as a controllable switch and matching transformer. An FXO port will give Asterisk/Zaptel a red alarm without loop current, so that probably won't work for a PA system that doesn't supply battery. What's wrong with using the (usually) unused sound card built into many machines? John Novack Mostly it uses the wrong impedance, and I know I can probably get an impedance matching transformer, but I'm not allowed to spend any money that I don't need to. (Otherwise I'd have replaced the amp in the first place with one that didn't fall from the arc). Won't you still need to maintain the loop current to make the PAP2 look like the port is off-hook? (FXS BTW) I would think the impedance matching xformer falls int the need to category. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
Thomas Kenyon wrote: John Novack wrote: Bob Chiodini wrote: You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... FXO ports do not generate loop current, they detect loop current from the Central office. Think of an FXO as a controllable switch and matching transformer. An FXO port will give Asterisk/Zaptel a red alarm without loop current, so that probably won't work for a PA system that doesn't supply battery. What's wrong with using the (usually) unused sound card built into many machines? John Novack Mostly it uses the wrong impedance, and I know I can probably get an impedance matching transformer, but I'm not allowed to spend any money that I don't need to. (Otherwise I'd have replaced the amp in the first place with one that didn't fall from the arc). Won't you still need to maintain the loop current to make the PAP2 look like the port is off-hook? (FXS BTW) I would think the impedance matching xformer falls int the need to category. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] very slow network from GXP-2000 switch port
On Fri, 2006-06-02 at 12:01 +0200, Louis-David Mitterrand wrote: Hello, At a client site yesterday I installed a dozen GrandStream GXP-2000's with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX and phones: network access for users windoze PC's through the phone's switch port was unbearably slow, making it almost impossible to work. When plugging back PC's directly to the LAN speed was normal again. On my test setup with a single phone here a the office I don't have that problem. Is there a known issue with that firwmare version? Could the switch be playing foul? (a Netgear FSM-7326-PEU) This sounds like a speed/duplex autonegotiation problem. Check the PC's (and phone's) network speed and duplex configuration. 100 Mbit/half duplex or 10Mbit/full duplex tend to be problematic. If you ping from the PC to the upstream router and lose half, or more of the pings this is probably the problem. Try forcing 100 Mbit full duplex at the PC and/or the switch. Check the PC's network config before and after placing the phone in-line. If they are Windows-based PCs the config is somewhere in the network properties dialog. If needed, an example: http://ots.iit.edu/network/windowsxp_fullduplex.php For Linux, use ethtool. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio streaming points different with VRRP
On Thu, 2006-06-01 at 12:30 +0200, Shenen Shenen wrote: Hi!I've a question: I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connection goes on the second asterisk? I want this: I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the other asterisk but not the audio streaming...the callers are always pointed to asterisk1, but for the right run, the callers must point to the asterisk2 Is there some *.config file where I can put my vrid IP, so in automatic the asterisk1 and asterisk2 translate their IP to the vrid? The vrrp is right like I set it.Asterik1 is the master with 192.160.252.1 IP and vrid like 192.160.252.10 asterisk2 is the slave with 192.160.252.2.Via Ethreal they are ok,if .1 goes down, .2 goes up, but the audio streaming point always to 192.160.252.1. Help me please, 1 thanks I'm not an expert, but... Shouldn't the stream point to .10? It may not be a IP issue, but an ARP cache/MAC address problem. Please provide more details on your configuration, i.e. what's running VRRP (e.g. router, linux box, etc.) and what the network looks like around the redundant machines. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Mail Audio Progression
Good Morning, I've been trying to set up [EMAIL PROTECTED] and thing are going pretty well. I do have a question: When I *98 into voice mail I hear a message that says Asterisk mail then short pause then the word mailbox then a very long pause, then a request for a password. I believe some other audio should occur in the long pause between mailbox and the request for a password. Am I correct? What should I hear and how can I troubleshoot? From the log: May 24 22:08:06 VERBOSE[4862] logger.c: -- Executing VoiceMailMain(SIP/201-33cc, default) in new stack May 24 22:08:06 DEBUG[4862] channel.c: Scheduling timer at 160 sample intervals May 24 22:08:06 VERBOSE[4862] logger.c: -- Playing 'vm-login' (language 'en') May 24 22:08:08 DEBUG[4865] manager.c: Manager received command 'Command' May 24 22:08:08 DEBUG[4865] manager.c: Manager received command 'Command' May 24 22:08:09 DEBUG[4862] channel.c: Scheduling timer at 84 sample intervals May 24 22:08:09 DEBUG[4862] channel.c: Scheduling timer at 0 sample intervals May 24 22:08:09 DEBUG[4862] channel.c: Scheduling timer at 0 sample intervals May 24 22:08:19 DEBUG[4862] channel.c: Scheduling timer at 160 sample intervals May 24 22:08:19 VERBOSE[4862] logger.c: -- Playing 'vm-password' (language 'en') May 24 22:08:20 DEBUG[4862] channel.c: Scheduling timer at 63 sample intervals May 24 22:08:20 DEBUG[4862] channel.c: Scheduling timer at 0 sample intervals May 24 22:08:20 DEBUG[4862] channel.c: Scheduling timer at 0 sample intervals May 24 22:08:23 WARNING[4862] app_voicemail.c: Unable to read password May 24 22:08:23 DEBUG[4862] pbx.c: Extension *98, priority 5 returned normally even though call was hung up May 24 22:08:23 VERBOSE[4862] logger.c: -- Executing Macro(SIP/201-33cc, hangupcall) in new stack Any hints? Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Mail Audio Progression
I don't hear a request for my mailbox number. Should it say something like Enter mailbox number? Bob... Avi Miller wrote: On 25/05/2006, at 8:14 PM, Bob Chiodini wrote: message that says Asterisk mail then short pause then the word mailbox then a very long pause, then a request for a password. I Its asking you for your mailbox number at that point, then pausing to allow you to enter the mailbox number. When you don't, it assumes you mean the mailbox associated with the extension you're dialling in from. Hope that helps, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Mail Audio Progression
Avi, Got it! Thanks. The minimalistic approach :-) Bob... On Thu, 2006-05-25 at 21:07 +1000, Avi Miller wrote: On 25/05/2006, at 8:57 PM, Bob Chiodini wrote: I don't hear a request for my mailbox number. Should it say something like Enter mailbox number? I believe the prompt just goes Mailbox? -- its not great. But, there's no other prompts being played in your output. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit - EXPLAINED
Jürgen, The TAE jack sounds like a great idea. In my house all of the phone and data cabling is home-run to a punch-down block in a Comm closet. The single DSL/POTS filter is located there along with the modem router and a SPA-3000. Other than a nearby lightning strike destroying my filter, router and one NIC, it works pretty well. It's very similar to your diagram. Bob... On Wed, 2006-05-10 at 10:44 +0200, Juergen K. Zick wrote: Well, Bellsouth gave me a box of filters that have two RJ-11 jacks. One for the DSL modem and one for a phone. The instructions specified that every phone be connected to a filter. The DSL modem would then be connected to the DSL jack along with one of the phones. The modem should not be connected directly to a phone line. The point is to isolate (filter) the DSL signaling from the voice signaling as Juergen describes. However in the US the wiring is not, typically, home-run, but daisy chained, one wall plate to the next with no place to put a whole house filter. Telcos do not like customers in the demark at the cable entrance and customers can install their own DSL equipment. I did. Bellsouth mailed the equipment to me with a CD and a set of paper instructions. It's just easier to tell the DIY'r to filter everything. Yes, I was aware of that daisy chain problem. Nowadays, the same problems are appearing in German flats as well, as many outlets / jacks are mounted to the same line. And I agree that the TELCO advice for filtering everything is much simplier for the DIY people, _BUT_ it can lead to a mismatch of your phone line and additional reflections especially in the RF band where the DSL signals are being located. German phone jacks (TAE jacks) are different from RJ jacks and include switches. That mean that you can install the splitter into the _FIRST_ TAE jack connect your DSL modem to the splitter and the filtered POTS output is automatically being sent into the the rest of the daisy chain ... The proposed filtering works _ONLY_ when your line to the DSLAM is quite short and you have not a high attenuation on it. --Jürgen But slowly, we are getting completely off-topic on this list. I doubt that changing to static IP will solve to decribed problem, because it is a line mismatch problem on the physical layer of the connection. And these will not go away unless you change the wiring ! Hadar, I would suggest to try my wiring first before you take other action to buy something. Also, while testing the line with BellSouth, I would ask for BERT-tests in the ATM-layer loop of your DSL.connection while your father has no phone talk on the POTS side and then with a running phone talk on on his phone. Bob... On Tue, 2006-05-09 at 20:41 +0200, Juergen K. Zick wrote: Well, to avoid a misunderstanding see the following drawing: /---DSL-MODEM-HT-PC H| | +--+ Inet-PHONE from BellSouth (DSL over POTS) ---| SPLITTER | +--+ L| \-- | | answering POTS machine phone (maybe you have to reformat it into COURIER font) It's depending on the calling in your father`s flat but on the incoming line you should have only _ONE_ device, the SPLITTER !!! maybe your dad can try that ... All DSL connections in Germany are build up like that and I have not seen any that did _not_ work with this cabling ... --Jürgen Juergen K. Zick wrote: HI, well, that was what I expected in my posting yesterday. For me, your wiring looks strange. Here in Germany, we have spiltters connected to the incoming line which have two outputs: A high pass filter output for the DSL signal and a low pass output with DC pass-through for the POTS signal. the DSL output is being connected to the DSL-modem and the POTS output will feed your internal POTS wiring. The only jack that has both a phone and the DSL connector indeed has a splitter on it, provided by Bellsouth. Therefore, there is _NO_ filter needed on each POTS outlet, because there is nothing to be filtered out on your internal line anymore. You may be correct. I am definitely _not_ familiar enough with DSL. However, 5 years ago, I had a DSL line in my apartment, and I was specifically told by the installation tech that I needed a filter on _any_ jack that had a real phone connected to it. That may not have been necessary, or perhaps isn't necessary any longer, or perhaps varies by
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
Thanks for the welcome. I can't say too much about how BS does DHCP, my address is fixed. I would not expect the address to change too often. When I had a DHCP address I don't think it changed when the phone rang. I've got a couple of these filters laying around. BS hands them out like candy when you move. I could put one in an envelope and send it to your father, or deliver if he's near me. I'm in Brevard county. Bob... On Tue, 2006-05-09 at 08:55 -0400, Hadar Pedhazur wrote: Bob Chiodini wrote: I'm a Bellsouth DSL user in FL too. Here, the filter has a DSL/modem jack and a POTS jack. So if a phone and modem share the same wall plate the filter does the split. Interesting, I'm pretty sure that when they installed it in his apartment, they put in the Y cable, so it's definitely a supported Bellsouth configuration. I don't think connecting the DSL modem directly the loop is wise. That's assuming that the filter actually filters something on the DSL port and that the modem does not have a built-in filter. My modem is a Westell. http://en.wikipedia.org/wiki/DSL_filter Thanks. One other possibility, the ringing is causing packet loss (UDP) that the HT486 is not handling very well. Normal TCP traffic would generally recover. The streaming audio test should confirm the loss. Does the HT486 have any kind of logging? I don't know about the logging, but you might be correct with regard to the packet loss. Rich Adamson conjectured that's it's a firewall issue, and it certainly feels like that. Last night, it occurred to me that perhaps an answered POTS line causes the modem to request a new DHCP lease, meaning, it changes it's IP address. If that were the case, it would explain the behavior I'm seeing, namely that we can continue to hear him, because the HT can still find the remote end, but the remote end can no longer find him... I don't know how easily I can verify that (remotely, I'm not sure I can talk my Dad through that one ;-), but perhaps I can prove that theory one way or another... I'm still testing VOIP (read newbie) and have not run across this scenario. I'll add it to my list of things to test. Welcome! Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit - EXPLAINED
Bellsouth gave me a box of filters that have two RJ-11 jacks. One for the DSL modem and one for a phone. The instructions specified that every phone be connected to a filter. The DSL modem would then be connected to the DSL jack along with one of the phones. The modem should not be connected directly to a phone line. The point is to isolate (filter) the DSL signaling from the voice signaling as Juergen describes. However in the US the wiring is not, typically, home-run, but daisy chained, one wall plate to the next with no place to put a whole house filter. Telcos do not like customers in the demark at the cable entrance and customers can install their own DSL equipment. I did. Bellsouth mailed the equipment to me with a CD and a set of paper instructions. It's just easier to tell the DIY'r to filter everything. Bob... On Tue, 2006-05-09 at 20:41 +0200, Juergen K. Zick wrote: Well, to avoid a misunderstanding see the following drawing: /---DSL-MODEM-HT-PC H| | +--+ Inet-PHONE from BellSouth (DSL over POTS) ---| SPLITTER | +--+ L| \-- | | answeringPOTS machine phone (maybe you have to reformat it into COURIER font) It's depending on the calling in your father`s flat but on the incoming line you should have only _ONE_ device, the SPLITTER !!! maybe your dad can try that ... All DSL connections in Germany are build up like that and I have not seen any that did _not_ work with this cabling ... --Jürgen Juergen K. Zick wrote: HI, well, that was what I expected in my posting yesterday. For me, your wiring looks strange. Here in Germany, we have spiltters connected to the incoming line which have two outputs: A high pass filter output for the DSL signal and a low pass output with DC pass-through for the POTS signal. the DSL output is being connected to the DSL-modem and the POTS output will feed your internal POTS wiring. The only jack that has both a phone and the DSL connector indeed has a splitter on it, provided by Bellsouth. Therefore, there is _NO_ filter needed on each POTS outlet, because there is nothing to be filtered out on your internal line anymore. You may be correct. I am definitely _not_ familiar enough with DSL. However, 5 years ago, I had a DSL line in my apartment, and I was specifically told by the installation tech that I needed a filter on _any_ jack that had a real phone connected to it. That may not have been necessary, or perhaps isn't necessary any longer, or perhaps varies by provider, but that's what I was told at the time, and that's what I did (with no problems). The filters on the phone jacks that didn't have the modem connected were not splitters, just single filters. Seen from my German wiring knowlegde, your cabling is wrong and causes the interruptions on the DSL service. That's definitely possible, just not my personal (single point!) experience. Don`t you have something like a spiltter available ? It should be the _ONLY_ filter on your incoming line and then the DSL-modem and the POTS phone should be connected to it ... OK, it would be easy for him to remove the other filters temporarily and test again. Thanks! --Jürgen Replying to my own post (and my most recent follow-up). I have now confirmed 100% that the DSL modem gets a _new_ IP address every time his real phone gets answered, or hung up! This (of course) disrupts the audio coming from to him, since the sending machine (Asterisk in my case), no longer has the correct IP address to send to him. I lowered his registration from the default 1 hour to 1 minute, so after we're disconnected, I can see that he's re-registering with a new IP address, each and every time :-(. I told him to call Bellsouth and ask about a Static IP address, but I don't know if they offer it, or how much they charge. While this one isn't solved, it's at least explained. Thanks to everyone who responded! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit
I'm a Bellsouth DSL user in FL too. Here, the filter has a DSL/modem jack and a POTS jack. So if a phone and modem share the same wall plate the filter does the split. I don't think connecting the DSL modem directly the loop is wise. That's assuming that the filter actually filters something on the DSL port and that the modem does not have a built-in filter. My modem is a Westell. http://en.wikipedia.org/wiki/DSL_filter One other possibility, the ringing is causing packet loss (UDP) that the HT486 is not handling very well. Normal TCP traffic would generally recover. The streaming audio test should confirm the loss. Does the HT486 have any kind of logging? I'm still testing VOIP (read newbie) and have not run across this scenario. I'll add it to my list of things to test. Bob... On Mon, 2006-05-08 at 15:23 -0400, Hadar Pedhazur wrote: Juergen K. Zick wrote: Well, I have no idea how DSL lines are connected in the US but what happens to a normal Internet connection when the phone is being picked up ? Test scenario could be that your Dad is listening to an Internet radio station or other audio stream and then being called Great idea! It's possible that there is a hiccup when the phone gets picked up, which a streaming audio connection might feel as well, in which case Bellsouth would have to acknowledge the problem ;-). Thanks, I'll have him test that. BTW, how are the real phones and the answering machine being connected ? Is the HT in front of them in the POTS line ? They are separated. The answering machine is in another room, connected to a normal phone jack, using a DSL filter to assure it doesn't get the noise of the DSL line. The HT is connected to the DSL modem, and there are no POTS lines connected to the FXO back-up port on the HT. In other words, the HT has only the WAN (to DSL) and LAN (to PC) ports connected, and an analog phone (a GE 5.8ghz handset) connected to it's FXS port. The only other possible connection problem (which I think I tested and rejected as a problem two months ago) is that there is a Y splitter coming out of the jack with a filter for the real phone, and no filter for the DSL modem (which is in front of the HT). I am reasonably sure that I had him remove the Y cable, and plug the DSL modem directly into the jack, and it still failed. However, I'll retry that again too now :-). Thanks Juergen! --Juergen I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone connected to it, and he even replaced one or two of them (when I thought that was the problem). I sent him a HandyTone GS-486 (HT), configured to connect back to my Asterisk server. He only has a single computer in his apartment, so it's connected into the HT, and the HT is connected into the DSL modem. He can make and receive calls on the HT, and the quality is excellent. If he's speaking via the HT (meaning a VoIP-only call) and the real phone rings, everything continues fine (temporarily). If the real phone is answered, either by a person, or by the answering machine (which is in another room, connected to a filter on another jack), then the audio on the Asterisk conversation becomes _one way_. My father can be heard _perfectly_ by the remote side of the conversation, but he can hear nothing. When the POTS line is hung up, then both sides of the VoIP call go dead (audio-wise). Of course, he can now redial a VoIP call, and both sides work perfectly... At first, I couldn't imagine that it was anything other than a bad filter, but other than replacing the filter (which didn't help), nothing else stops working. He can continue to use the Internet connection on his PC just fine, and I can continue to hear him speak over the VoIP connection with no problems either, so the Internet connection has not been lost. I have to admit to being completely clueless as to what to even look for, so _any_ advice as to things to test for would be appreciated. As I said at the top, I can reproduce this 100% of the time, so I can easily setup any debugging environment in advance, and trigger the problem at will, etc. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options
