[asterisk-users] Anyone use the Linksys phones?
Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. I have read through the literature, but would like some real world feedback. Thanks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] IAX2 + port translation
Dr. Michael J. Chudobiak wrote: Hi all, I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they work, but sometimes the caller just gets dead air or disconnects. IAX2 debugs show HANGUP and INVALID codes in these cases, rather than a proper RINGING transaction. My firewall is doing NAT, and changing the source port from 4569 to something else - my IAX2 provider suggested this might be a problem. Is it? Should this work: steerpike*CLI iax2 show registry Host UsernamePerceived Refresh State 64.26.157.230:45698886708729 64.26.155.62:14353 60 Reg 64.26.157.230:45696134827945 64.26.155.62:14353 60 Reg 64.26.157.230:45696136866597 64.26.155.62:14353 60 Reg 64.26.157.230:45696136866675 64.26.155.62:14353 60 Reg There are four DIDs, and all are registered to an odd port (14353). Is this OK? (I am using a Sonicwall TZ170 with Enable Consistent NAT on). - Mike If memory serves me properly what you are showing looks correct. You server is registering to your provider on port 4569 as it should. Their server is seeing you register from 64.26.155.62 and using the prt 14353 which is the port that your firewall has given that outgoing connection. Possibly that the firewall is removing that connection port after some time and your provider cannot get back to your box? Try setting the reregistration time lower than 60 and see if it helps. ___ --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] PCI Problems
Sean Cook wrote: OK... maybe I got a little anxious and ran out and bought a Tyan GX28 with dual Opteron (dual core) processors. (It is a nice server ;) ) I did neglect to find out that you can not manually set the IRQ's on this motherboard. I am now stuck sharing an IRQ with the ethernet controller and no foreseeable end to my dilemma. I have a Digium TE210P and zttest consistently runs at 99.97% which as you guessed, is giving rather unpleasing sound quality. My options as I see it are: 1. Buy a new server 2. Buy a sangoma A102U I am looking for practical suggestions from those of you out there who have had a similar experience that may aid me in making this decision. Thank you, Sean ___ Have you tried changing the PCI slot and resetting the bios config? Robert ___ --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] List of transcoding combinations
Is there a list or matrix somewhere that shows what codec can be transcoded? I am playing with different allowed codecs between my asterisk box and some of my providers testing voice quality and bandwidth usage on my cable connection, and I occassionally run into an issue where asterisk cannot convert between two codecs. For instance G.723 and ULAW will not work together through asterisk. Would like to have a matrix of some sort where I know ahead of time what combinations I can and cannot use. 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
[Asterisk-Users] openSUSE 10.0 and zaptel init script
Hi all, I just installed openSUSE 10.0 on a spare machine to try and do some development work. I did a checkout on libpri, zaptel, and asterisk and everything compiled and installed perfectly. My issue is with the zaptel script placed in the rc.d directory to automatically initializ the zaptel modules. When running zaptel start, I get an error that /etc/rc.d/functions does not exist. I have searched the server and found the functions script in two different locations but neither of them work with the init script. Is there a package that actually adds the proper functions script to rc.d that I am missing? If I look on a debian machine, I see the functions script under /etc/init.d where it should be. TIA Robert ___ --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 Jitter Buffer for 1.2.5
On Tue, 14 Mar 2006 14:32:02 +0100 Olle E Johansson [EMAIL PROTECTED] wrote: 14 mar 2006 kl. 13.35 skrev Matt: Right saw that. But I'm trying to get away from using CVS-HEAD :) We all are. Every developer have switched from CVS to Subversion :-) This is not the development branch, but the release branch code, which we use to create the 1.2.x releases. The jitterbuffer itself is *not* release branch code, it's very much development. Please test it. The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). /O Olle, Pardon this dumb question please, but where are these test located. I looked under http://svn.digium.com and do not see them. I am not fluent in where everything is located and would like to do some testing on some of the other items such as the sip jitterbuffer. It will only be minimal but I would like to help where I can. Robert ___ --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 Jitter Buffer for 1.2.5
On Tue, 14 Mar 2006 13:44:57 -0500 Matt [EMAIL PROTECTED] wrote: http://svn.digium.com/view/asterisk/team/oej/jitterbuffer-1.2/ Thank you I was looking directly under asterisk and not team. :-) Robert ___ --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] Oh this is bad.... bindaddr and rtp traffic
On Tue, 7 Mar 2006 09:12:25 -0700 Douglas Garstang [EMAIL PROTECTED] wrote: I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp0788 0.0.0.0:50600.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A 'rtp debug' shows it's receiving the RTP packets, it just seems it does nothing with them. Anyone seen this? Doug. I thought all RTP was controlled through rtp.conf and only the SIP traffic was controlled through SIP.conf. I am not sure what settings, beside the RTP port range, you can out into the rtp.conf though. Robert ___ --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] [OT] List messages and end user outages
Sorry, this is off topic to asterisk itself, but is about the list server. I had a power failure lastnight at home, where my email server resides, and my network was down for about 20 minutes, that was after 45 minutes of uptime on UPS. Since power was restored, around 9:45 PM EST on 2/16, I have not received a single post from the users, biz, or dev lists. Normally when this has happened in the past, it has taken 24 hours for the list server to start sending to my email server again. My question is why so long? I am on other lists and it might take an hour or so for the messages to start showing up, but why 24 hours for a 20 minute loss of contact with my email server? Robert P.S. - If there is somewhere else this question should be directed, that would be constructive, please feel free to let me 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] Newbie question
On Wed, 15 Feb 2006 08:59:22 -0800 (PST) housi mueller [EMAIL PROTECTED] wrote: Hi there, I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive. I dont now which card to take. Please tell me what you think about. I appreciate all suggestions. Thanks in advance Housi Mueller My personal preference would be to go with the E1/T1 now. It would give you expansion opportunities in the future between the Asterisk and the Panasonic, allow you to be all digital between, and finally if you ever decided to ever get rid of the Panasonic, you could pull a T1 from the telco straight into the Asterisk box. Spend a little more now and save in the future. Just my $.02 Robert ___ --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] Help with bad audio using MPC..
I sent the below message out last Friday when the list seemed to be having issues. Never got any responses and not sure if it just no one knows or if it did not get through. Please don't flog me too bad for reposting... :-) Hi all, I am having some audio quality issues with a provider under sip. The issue I am having is that the audio seems to be acting like a simplex connection. I have tested my setup with a second provider and the audio quality to them is great. Checked network type issues, latency, packet loss, etc. and all seems to be ok. What I did find was a difference in the RTP debugs. Here is a capture from both providers: RTP Debug from Teliax SIP connection w/ good audio: Sent RTP packet to 208.139.204.228:10102 (type 0, seq 9473, ts 135520, len 160) Sent RTP packet to 208.139.204.228:10102 (type 0, seq 9474, ts 135680, len 160) Got RTP packet from 208.139.204.228:10102 (type 0, seq 4467, ts 149600, len 160) Got RTP packet from 208.139.204.228:10102 (type 0, seq 4468, ts 149760, len 160) RTP Debug from MPC connection w/ bad audio: Sent RTP packet to 66.128.8.234:61414 (type 0, seq 3506, ts 51040, len 160) Sent RTP packet to 66.128.8.234:61414 (type 0, seq 3507, ts 51200, len 160) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23701, ts 52480, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23702, ts 52560, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23703, ts 52640, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23704, ts 52720, len 80) Notice that the lengths are different in the MPC packet capture. I am getting two packets from them to every one of mine. I was askied by them to set my packet size to 20ms but do not know where to do that or if it can be done. They also stated that the packet size should be negotiated in the SIP INVITE and 200 OK messages. Can someone point me in the right direction? Even just what to look for here. I am currently running version 1.2.2, but had the same issues with 1.09 and 1.2. Thanks, Robert ___ --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] Help with poor audio using SIP
Hi all, I am having some audio quality issues with a provider under sip. The issue I am having is that the audio seems to be acting like a simplex connection. I have tested my setup with a second provider and the audio quality to them is great. Checked network type issues, latency, packet loss, etc. and all seems to be ok. What I did find was a difference in the RTP debugs. Here is a capture from both providers: RTP Debug from Teliax SIP connection w/ good audio: Sent RTP packet to 208.139.204.228:10102 (type 0, seq 9473, ts 135520, len 160) Sent RTP packet to 208.139.204.228:10102 (type 0, seq 9474, ts 135680, len 160) Got RTP packet from 208.139.204.228:10102 (type 0, seq 4467, ts 149600, len 160) Got RTP packet from 208.139.204.228:10102 (type 0, seq 4468, ts 149760, len 160) RTP Debug from MPC connection w/ bad audio: Sent RTP packet to 66.128.8.234:61414 (type 0, seq 3506, ts 51040, len 160) Sent RTP packet to 66.128.8.234:61414 (type 0, seq 3507, ts 51200, len 160) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23701, ts 52480, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23702, ts 52560, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23703, ts 52640, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23704, ts 52720, len 80) Notice that the lengths are different in the MPC packet capture. I am getting two packets from them to every one of mine. I was askied by them to set my packet size to 20ms but do not know where to do that or if it can be done. They also stated that the packet size should be negotiated in the SIP INVITE and 200 OK messages. Can someone point me in the right direction? Even just what to look for here. I am currently running version 1.2.2, but had the same issues with 1.09 and 1.2. Thanks, Robert ___ --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] video development
On Wed, 11 Jan 2006 15:38:04 +0100 Matt Riddell (IT) [EMAIL PROTECTED] wrote: I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SEREyeBeam ) greetings and thanks in advance. Asterisk already does this. We provide Video IVR creation for customers. All you have to do is have an audio file and video file that are the same length and then play the audio file, the video file will be played with the audio. H264 support was added to Asterisk about 3 days ago. H263+ has been in for a while. -- Cheers, Matt Riddell As a noob that might be interested in this also, how well does this work with the seperate audio and video files and keeping them in sync? I just keep flashing back to the old days of trying to do stereo with music using two C64's.. :-) Robert ___ --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] Nested MySQL Commands
On Wed, 11 Jan 2006 11:39:20 -0700 Douglas Garstang [EMAIL PROTECTED] wrote: Peter, Too slow! We're going to potentially be doing several MySQL lookups for routing even the most basic of calls, and if every one of those queries has to make a call out to an AGI script, it would become a performance problem. Douglas. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nested MySQL Commands On 11/01/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. When things start to get that complicated, I reckon it's time for AGI Peter Has anyone yet played with MySQL version 5?? My understanding is that is now includes stored procedures. Wonder if that will help things with Asterisk? Send a query over to the MySql server with only the required parameters and have it do all the processing for you and only returns the results. I know that is a nice feature od Microsoft Sql. But have not had a chance to read up on the performance of the new version of MySql... Robert ___ --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] CDR MySQL
On Tue, 13 Dec 2005 02:25:50 +0100 Patrick [EMAIL PROTECTED] wrote: On Mon, 2005-12-12 at 22:08 -0300, Juanjo Portela wrote: My cdr_mysql.conf is the same I was using for version.1.0.9 and it is as follow [global] hostname=localhost dbname=dbasterisk password=dbpassword user=dbuser userfield=1 Any ideas? Any ideas about what? The weather? Specify your question in more detail so people don't have to guess what you are asking. Regards, Patrick Maybe if people weren't so quick to jump to conclusions, they would see that some did actually ask a question before and were now supplying the answer. http://lists.digium.com/pipermail/asterisk-users/2005-December/138183.html ___ --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] Porting a phone number to a voip provider
On Fri, 09 Dec 2005 00:36:18 -0500 Matthew matthew@zeut.net wrote: Hello, has anyone taken their cell phone number and ported it over to a voip provider? If so, what voip provider and what was your experience? Matt Matt, I have done this. I had a cell number with ATT Wireless and first ported it to Broadvox Direct. There service was ok but ended up not fitting my needs as trying to run their Mediatrix box into my Asterisk box was just not working too well. I have since ported it away from Broadvox Direct to a Voicepulse Connect Account. I am running it straight into Asterisk now over an IAX connection and it has been working fine for me. Not a lot of calls come in on it, so I cannot really tell you what the real up time is on it. I just know that no one has ever told me that they tried on that number and could not get me. Robert ___ --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] IAX and Firewall
On Fri, 18 Nov 2005 14:20:45 -0700 Joseph [EMAIL PROTECTED] wrote: Do I have to have IAX2 port (udp 4569) open when receiving calls from a registered server. My asterisk shows that it is registered with teliax server but the calls to my asterisk are being dropped? -- #Joseph I don't believe so. By registering with the remote server, you are giving them the NAT port to get back into your server with. All communications will take place on that port. THe only time you would need to open up the firewall and direct the port to your server would be if you have a user on the outside that is registering back into your Asterisk box. I know someone will correct me if I am wrong, but I believe that is the way it works. You have to forward ports for SIP because of the way the RTP stream is setup. Robert Webb ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fresh checkout Zaptel will not compile?
Hi all.. I just setup a test box with Debian running kernel 2.6. Went to CVS and did a checkout of the new beta 2 release using the command: cvs checkout -r v1-2-0-beta2 zaptel libpri asterisk asterisk-addons asterisk-sounds. I then compiled libpri fine and moved on to zaptel. Did a make clean then make install and get the following error: /bin/sh: line 1: [: argument expected make -C SUBDIRS=/usr/src/zaptel modules make: *** SUBDIRS=/usr/src/zaptel: No such file or directory. Stop. make: *** [linux26] Error 2 hecate:/usr/src/zaptel# I am not up to speed on make or its errors, but it looks like to me that it is complaining about /usr/src/zaptel not being there or that modules is missing. AS you can see from the last line there is a /usr/src/zaptel directory. Or is it something with my 2.6 kernel and a modules directory or something. Have never gotten this error before, that is why I am asking for help. Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fresh checkout Zaptel will not compile?
On Tue, 01 Nov 2005 10:18:45 -0500 Paul Zimm [EMAIL PROTECTED] wrote: I then compiled libpri fine and moved on to zaptel. Did a make clean then make install and get the following error:How about `make linux26' ? I am not up to speed on make or its errors, but it looks like to me that it is complaining about /usr/src/zaptel not being there or that modules is missing. AS you can see from the last line there is a /usr/src/zaptel directory. Or is it something with my 2.6 kernel and a modules directory or something. Have never gotten this error before, that is why I am asking for help. Robertdo you have kernel sources installed? I just dealt with these same issue on a recent install on debian. You don't need the kernel sources, but you do need the kernel headers for the kernel image you're running. You need to have a sym-link named /lib/modules/`uname -r`/build/ which links to your kernel header directory. Zaptel wouldn't compile properly until I removed the 2.6 kernel source files, because I had the 2.6 version of the kernel sources and my kernel image and headers were 2.6.11. Marv Horst dicovers the kernel source That was it. I had recently upgraded from 2.4 to 2.6.8 and thought I had added in to install the headers but had not. I did the source but not the headers. Seems to be compiling fine now. Thanks for the brain jog... :-) Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware setup question
I have just a quick setup question about how some of you have hardware setup. Basically, for a system that has an average volumes of calls in an office setting, are you using one or two network cards. I am just wondering if it owuld be any advantage to having one NIC for the extensions and one NIC for your trunks. Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Goiax.com DID not working anymore?
On Fri, 21 Oct 2005 10:25:59 -0400 Paul [EMAIL PROTECTED] wrote: Kanuri, Seshu (Company IT) wrote: [EMAIL PROTECTED] wrote It's a free service. It belongs on this list. Olle is right. Even if it is a free service it does not belong here. This forum is for any Asterisk related user issues, not some DID issue of one of a hundred such service providers. Take it off this list. Now that makes 2 of you who are wrong. Goiax.com is providing a valuable free service to asterisk users. For one thing it enables users to do some free testing of PSTN-asterisk setup. I believe the posters to this thread are likely 100% asterisk users so what is so bad about using the asterisk users mailing list for discussion? There are lots of unwarranted posts to all the lists from the totally clueless. Why don't you pick on them instead? No, this belongs on the asterisk-biz list as this is an issue of business practice not an operational issue of the Asterisk software itself. The -users list is for those that are having issues with getting Asterisk up and running or trying to figure out how to do certain software realated tasks or scripting. Can you not comprehend the difference?? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Goiax.com DID not working anymore?
Just tested mine and it is working fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Blake KroneSent: Thursday, October 20, 2005 5:32 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Goiax.com DID not working anymore? I've been using my goiax.com DID for a few days now and it is no longer working. I get the number or code you dialed can not be found. I haven't touched any configs or anything on the asterisk box since it was working last night.Anyone else having problems using the DID from goiax?Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM Card FXO Question
I have a TDM card with one FXO and one FXS. I am trying to make sure I understand correctly the TX and RX Gain in the Zapata.conf correctly. If I have a phone cord plugged into an FXO port tied into a POTS line and boost the TXGain, am I correct in thinking that the audio going back to the phone company is boosted by X percentage?? TIA, Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM11B pinout
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gary Smith Sent: Monday, September 05, 2005 5:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TDM11B pinout Hi Asterisk Community, I have a development with a TDM11B in it. I am trying to connect this to a exchange line as well as a UK telephone and are looking for some pinout information for the FXS port. Is it the centre 2 pins that at the tip and ring. I have been digging around the Digium site but cannot seem to pick up this info. Any help appreciated. Thanks -- Gary Gary, A standard RJ11 telephone connector will work fine with the ports on the back of the TDM card. I am assuming that in the UK, you use the same connector as we do in the States.. :-) Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoringswitch= Realtime/[EMAIL PROTECTED] altime_ext
On Wed, 24 Aug 2005 14:47:25 -0400 Araba, Michael [EMAIL PROTECTED] wrote: Thanks John, You are my savior. This is such a great relief. Apparently realtime will not use either '127.0.0.1' or 'localhost' to connect to the database. I had to use the actual IP address attached to the NIC before it worked. SNIP You claim it is an Asterisk issue, did you by any chance make sure that database was allowing connections on 127.0.0.1 and localhost and not just the actual IP?? Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoringswitch= Realtime/[EMAIL PROTECTED] altime_ext
On Wed, 24 Aug 2005 15:25:15 -0400 John Novack [EMAIL PROTECTED] wrote: In my case, mysql is set to any host So, yes, it does seem to be an Asterisk issue And my buddy is pretty savvy with mysql, Linux and databases on Unix/Linux, having worked for a large IT company for some 20 years. John Novack P.S. Robert- Something wrong with your mail clock? You responded to a message hours before it was sent! Sorry, was not trying to insult your expertise.. Just sometimes things can get overlooked. Thanks for the heads up.. Something went awry with my email server today and the ntp client went screwy. Should be fixed now. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't get G729 working after buying a license.
SNIP Ok, I figured it out, * was not using the config under the [router] context in the config file. Once I enabled g729 in [general] it worked. So the question is why does * ignore this config for the 192.168.77.254 endpoint? in sip.conf: [router] type=friend context=default host=192.168.77.254 dtmfmode=info disallow=all allow=g729 nat=no canreinvite=yes qualify=yes Maybe double and triple check that the router context is actually being used. SOunds like it isn't. I have gotten caught in this situation before. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial through IAX to FWD
On Wed, 27 Jul 2005 18:07:23 +0200 Walid Azab [EMAIL PROTECTED] wrote: Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have = 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XX instead of having to dial 011*1800XXX? Thanks Are you using [EMAIL PROTECTED] or setting up the configs yourself?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A TDM issue..
On Tue, 26 Jul 2005 10:24:20 -0400 Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 26 July 2005 09:43, chouck wrote: I assure you I have read the asterisk handbook many times. The immediate=yes is for picking up a phone on an fxs and having it immediately dial an extension. I am looking for someone to dial an extension and have it immediately pick up the phone on an fxs port. I misread your message then, I apologize. immediate=yes means it immeditely enters the dialplan at the given context's 's' extension. It can dial an extension, execute an AGI, make your coffee... whatever you have in the dialplan. Now to answer your question though -- how do you intend for Asterisk to physically pick up some telephone somewhere? Are you wanting extension 5, for example, to not ring a telephone on an FXS port but have the phone automatically answer? The phone needs to have auto-answer capability... Asterisk can't make something answer a line, it can only ring the FXS port and wait for the connected device to answer... -A. Andrew, I just went back and read his original post. The port is interfacing with an intercomm system that does answer immediately. So what you have given him should work fine. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap channel configuration problem
On Mon, 25 Jul 2005 15:44:07 +0200 Alexis F. [EMAIL PROTECTED] wrote: Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I am sure someone will correct me if I am worng, but I just looked on Digium's sit to verify, and the Green moduls are FXS modules for connecting analog phone sets. NOT to connect to the PSTN. For the PSTN you would need the FXO module which is red. Hopefully you have not had any incoming calls, as that will blow up the FXS module and make it unusable I make modprobe zaptel modprobe wctdm without error return. Configuration files : /etc/zaptel.conf fxsks=1 loadzone = it defaultzone=it /etc/asterisk/zapata.conf [channels] language=en context=from-sip signalling=fxs_ks /etc/asterisk/extensions.conf TRUNK=Zap/g2 TRUNKMSD=1 DIALOUTANALOG=Zap/1 You configs above regarding signalling would be correct IF you had the FXO module and not the FXS module. SNIP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly 3rd party - it hangs on Initialising and exits with error
On Wed, 20 Jul 2005 18:00:24 +0200 Robert Rozman [EMAIL PROTECTED] wrote: Hi, I spot weird behaviour of latest Firefly 3rd party on my laptop. Sometimes it comes to state that it won't start (hangs on Initializing ) and it again works after system restart... Didn't yet figured out how to recreate it. Any similar experience ? Also - how can I force Firefly to make outgoing calls (also sip or iax calls) through Asterisk ? I'd like to make outgoing iax calls through Asterisk or other registered pbx so I can correct caller id, register outgoing call and other things Any advice ? Yeah... Try the web site for the writers of Firefly: http://www.freshtel.net/ This is an Asterisk USer list. Not a Firefly list. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly 3rd party - it hangs on Initialising and exits with error
SNIP . And please note that in general members of the list dislike List Police even more than they do off-topic posters. B. Cool... I will be sure to ask any question I have now and expect not to get Policed by anyone on this list. Sounds like this is the list for the support of ALL things Voip. Flame away all... Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Question: Type of card
On Thu, 7 Jul 2005 10:49:32 -0700 Dan Adams [EMAIL PROTECTED] wrote: Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan The link below gives you great information on the card you need. Look espicially close to the box with all the writing in it just below the URL to www.asterisk.org that starts with For interconnection with digital and analog telephony equipment http://www.voip-info.org/wiki-Asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simpletelecom dead?
On Mon, 4 Jul 2005 20:56:30 -0400 Jimmy Smith [EMAIL PROTECTED] wrote: 6 beyond-the-network.LosAngeles.savvis.net (208.173.57.30) 33.966 ms 34.143 ms 33.841 ms 7 * * * hangs there... savvis invoice paid ? beyond-the-network a black hole ? On 7/4/05, Gary Reuter [EMAIL PROTECTED] wrote: Hmmm Can't place calls... Can't access website... Neither of the 3 nameservers answer anything... Anyone heard/know something to explain all this? My guess is that they are out of business. I just tried calling the telephone number in Nevada listed on their domain registration and it states the number has been disconnected. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new Asterisk@home installation
On Tue, 5 Jul 2005 22:07:11 +0800 Ian Bert Tusil [EMAIL PROTECTED] wrote: I've just Installed [EMAIL PROTECTED] i browsed it's built-in AMP. it prompts for a login if you click on asterisk management portal. i tried user:[EMAIL PROTECTED] pass:password and user:admin pass:password but it didnt get through. do you the default login for it? thnx, ian Try this from the [EMAIL PROTECTED] handbook. The [EMAIL PROTECTED] site will tell you abut [EMAIL PROTECTED], not the Asterisk users list... http://asteriskathome.sourceforge.net/handbook/index.html#Section_3 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simpletelecom dead?
On Tue, 05 Jul 2005 11:26:39 -0700 Bruce Ferrell [EMAIL PROTECTED] wrote: I've gotten word from their Marketing VP. They are doing some kind of massive move and expect to be down until Thursday Sounds like their Marketing VP needs to get a clue and let customers know what is going on. If a company I paid money to did this, I would yank all my accounts.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: TDM11B Dev Kit PCI + Asterisk CVS Head
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith Caldwell Sent: Saturday, July 02, 2005 8:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: TDM11B Dev Kit PCI + Asterisk CVS Head Ok, after hours of research I finally found the problem. I found a document from digium at http://www.digium.com/asterisk_handbook/zapata.conf.html which states that everything above the channel=x statement applies to that interface which seems a little backwards to me. After reconfiguring I have context=internal signalling=fxo_ks callerid=Keith 100 channel=1 context=pstn-in signalling=fxs_ks callerid=asrecieved channel=4 Just in case anyone else has the same problem. Keith AS soo many people on this list have stated soo many times, the wiki and Google is your friend. Just by searching Zapata.conf on the wiki in the Google search box, found this in 10 seconds: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] passing through MWI info from SBC
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jon Radon Sent: Saturday, July 02, 2005 10:49 AM To: andrew matthews; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] passing through MWI info from SBC Woah woah woah.. why not just disable SBC voicemail and have asterisk handle it? I don't understand why you would go to such great lengths when you can just have Asterisk deal with it. Because in most cases, the POTS provider will not disable voicemail on a per number basis as it is a part of the 'package'. So there really isn't an option to do that. The other issue is that with call waiting, if you do not answer the call there is no way to have Asterisk handle the voicemail. So here is where you would still need the pass through in order for the subscriber to know there was a message. For me, I just got a $5 per month DID and forwarded all my POTS call to it. I get up to two simultaneous incoming calls that Asterisk handles completely. It also includes a voicemail system where if my connection goes down or I exceed those two calls, then if someone leave a message, it gets emailed to me. The other benefit is that I still have my POTS line and use it for all my local outgoing calls and use a 1.3 cent per minute provider for all LD calls and I still save money by not spending that extra $20 a month for unlimited LD on my POTS line. Has worked great so far. Only down side is if my DID connection drops, I have no way to call into the house. But, I just use the cell to call the wife in that case. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Friday, July 01, 2005 11:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP/[EMAIL PROTECTED] (asterisk at home) custom incoming routing A few things to followup on my earlier post: 1.) Definitely put the [tdm-in] context in the file /etc/asterisk/ extensions_custom.conf. That way your changes will not get overwritten. 2.) I am still unable to make call waiting on the incoming ZAP line work, b/c I have not thought up a good way to make this happen. hasn't anyone done this before? 3.) When setting up the ZAP trunk, I found it usefull to put w in the dial prefix field to force the system to wait for the dial tone. If I didn't do that, I could not call out on the ZAP Channel. Tom Tom, Actually, it is the extensions_custom.conf that DOES get overwritten. Unles things have completely changed since version 1.0. You should be using the extensions.conf to place all your permamnant changes. The _custom was there to tell you that those are the custom settings for [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: passing through MWI info from SBC
On Fri, 01 Jul 2005 11:10:27 -0700 Chris A. Icide [EMAIL PROTECTED] wrote: John Novack wrote: Mike Myers wrote: snip Wow, this is a serious problem for me. I don't need to actually check the voicemail itself from Asterisk, just to be able to tell that there is voicemail waiting. Are you saying there is no way in Asterisk to do this? Is that true for using Digium hardware as well as FXO ports on a SIP ATA? Vonage VM doesn't matter to me, since I'll turn it off and use Asterisk for that functionality, but determining SBC's VM status is very important. My whole wife's family (multiple households) uses it. In the past, if one family tried to switch to a non SBC provider, they always returned in less than a week because of lack of VM interoperation. So my wife will put the kibosh on the whole Asterisk project unless I can light the MWI light when SBC VM is waiting. Since the cheapest analog phones can do this, I don't think she's going to understand that these $200 Polycom phones can't... :-( Is there no way around this? Thanks, Mike Here is what I would do. Install a TDM04 card with a couple fxos. Connect the analog phones that your wife will be using to the tdm card. In zapata.conf, set those phones to immediate=yes, and when you get an event on the fxo port, connect it to the fxs port with the stutter tone. This way, when she picks up the phone, it will immediately connect her to the sbc provided dial tone, and she can hear the stutter or lack thereof. When a call comes inbout however, you can still route it as you want. Not a perfect solution, since the phones she will be using are forced to use SBC, but the best solution I can think of. -Chris Or, could you use something like the zap_barge option tied into a routine that monitors for the FSK and then when it is received, it then runs the routine that is already in place to set the MWI for the FXS ports. Crude, I know, but the only way I can think of to pass it. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to do very simple Zaptel Config. NO LUCK!
See messages inline...
On Thu, 30 Jun 2005 09:48:51 -0700 (PDT)
David [EMAIL PROTECTED] wrote:
Hi,
I am trying to do the world's most simple install.
I have a Wildcard TDM400P with 3 ports: 1 FXS on port
1 and 2 FXOs on ports 3 and 4. (i'm not using port 3
for now, put want it for expansion purposes)
I simply (to start with) am looking to have the FXS
phone ring when an FX0 port is dialed. I would also
like to be able to place outgoing calls on the FXS
through the FXO. Right now, I'm not interested in SIP
or IAX... thats for me to handle later!
I have the following *3* files:
***/etc/zaptel.conf***
# Zaptel conf
fxoks=1
fxsks=3-4
loadzone=us
defaultzone=us
Looks as it should...
***/etc/asterisk/extensions.conf***
[globals]
RECEPTIONIST=Zap/1
LOCALTRUNK=Zap/4
[incoming]
exten = s,1,Answer()
exten = s,2,Dial($(RECEPTIONIST))
[internal]
[outgoing]
ignorepat = 9
exten =_9NXXNXX,1,Dial(${LOCALTRUNK}/${EXTEN:1})
exten =_9NXXNXX,2,Playback(invalid)
exten = _9NXXNXX,3,Hangup
Looks ok at initial glance.. But I did not take the time
to really think about what you have..
***/etc/asterisk/zapata.conf***
language=en
context=default
switchtype=national
signalling=fxo_ks
channel = 1
signalling=fxs_ks
channel = 3
channel = 4
Here is where your issue is..
The channel= line signifies the end of any config
information for the channel(s) put here. And if you have a
config option for an earlier channels and either want it
to be different of not exist in a later channel, you must
make that know by adding that option for the current
channel config. So by what you have above, all three
channels are using the default context. Try using what I
have wriotten below:
***/etc/asterisk/zapata.conf***
language=en
context=outgoing
signalling=fxo_ks
channel = 1
signalling=fxs_ks
context=incoming
channel = 3,4
The above will direct any incoming calls on channel 3 or 4
to the incoming context in your dial plan. It will also
direct any one picking up a phone handset to get dial tone
and direct what they dial to the outgoing context.
And by your dial plan above, they must use 10 digit
dialing.
Hope this helps.
Thats my entire setup. I dont get any dialtone on the
fxs and the fxo doesnt pick up the phone when it
rings. Any ideas what I'm doing wrong? THANKS in
advance for your help.
David
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RE: [Asterisk-Users] Gizmo: Skype done right?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hank Sent: Thursday, June 30, 2005 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Gizmo: Skype done right? they claim to have a windows download but I can't get the program. also they give no instructions on how to get it connected to asterisk Which brings us to the question... Why is this being said to be good for Asterisk?? I did download it and load it on my computer. But there are NO options for connecting to anything or anyone else but a Gizmo account. So just how is this good for the open source VoIP community and Asterisk?? Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax Problems
On Wed, 29 Jun 2005 08:15:20 -0400 Chris Mason (Lists) [EMAIL PROTECTED] wrote: An ethereal trace indicates the IP address is active, but it is not responding to iax packets (registration). So, either their asterisk app has failed or they have folded their tent as well. I am sure it's just a crashed server, wait an hour and let the support people deal with it. -- Chris Mason NetConcepts The server is up as IAXPing generates responses from voip-teliax.com and voip-co2.teliax.com Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hidecallerid on analog line
On Wed, 29 Jun 2005 13:56:00 -0700 (PDT) chawki hammoud [EMAIL PROTECTED] wrote: Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; It would have to be done through who ever provides your POTS service. They provide the caller ID to who you are calling. Some have the option to block it. Asterisk cannot be configured to do this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt
On Mon, 27 Jun 2005 15:27:22 -0400 Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Monday 27 June 2005 14:31, Michael Di Martino wrote: If this list spent at least half the time on helping other asterisk admins as it does on trivial things like LiveVoips bankruptcy it just might be a great list. As it stands now this list is kind of useless. Most request for assistance with asterisk problems go unresolved of unanswered. Do you have some proof of this? I find the list rather helpful on the whole, with interjections of other (sometimes very OT) subjects inbetween. -A. I think he is just frustrated because only two people have replied to his question about his IAXy device not working after having repeated his same question a dozen times in new threads.. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Di Martino Sent: Monday, June 27, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. In a Google search box: site:lists.digium.com What you are searching for ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: iaxy over the public cloud
I am trying to get an iaxy device to connect to my asterisk box over the public cloud however It fails register and I cannot figure out why. Below is my iax.conf, iaxy setup file and out from iax2 debug. My iax.conf [u7403] type=friend accountcode=iaxy host=dynamic secret=u7403p context=from-iaxy disallow=all allow=ulaw callerid=my iaxy 7403 trunk=no notify=yes Mi iaxy setup file [EMAIL PROTECTED] iaxyprov]# cat iaxy.conf.7402 ; ; IAXY Provisioning description ; dhcp ;ip: 216.207.244.130 ;netmask: 255.255.255.192 ;gateway: 216.207.244.129 codec: ulaw ;codec: adpcm server: 207.251.84.198 ;altserver: 192.168.0.2 user: u7402 pass: u7402p SNIP Well, the most notable thing I see immediately is that in your Asterisk config you are using [u7403] and in your IAXy config you are using u7402. Might try getting those to match first. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail
On Fri, 24 Jun 2005 09:27:45 -0400 Michael Di Martino [EMAIL PROTECTED] wrote: Ok I have added the timeout value but it still does not pick. However jus to test voicemail function I comment out the first line and voice does pick up. What could be wrong. exten = 7403,1,Dial(IAX2/u7403/1/5) exten = 7403,2,Voicemail(u7403) exten = 7403,102,Voicemail(b7403) exten = 7403,103,Hangup AS someone esle suggested, go to www.voip-info.org and read about the dial command. It is obvious you haven't as your syntax in for the commands above are incorrect. Between the extension and the timeout, you use a , not a /. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK
On Fri, 24 Jun 2005 13:10:13 -0400 (EDT) [EMAIL PROTECTED] wrote: On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote: We are on a real world... Every cyber cafe has its own little hacker/cracker that is sniffing out... A simple ethereal capture could give me a bank pin number... It is REALLY trivial! And this is different exactly HOW with inband DTMF?? They can do the EXACT same thing! If you want security don't use VOIP unless it's encrypted and/or over a VPN. It's really that simple. Ok, point me on HOW may I get DTMF inband with ethereal. Andrew, I'm just looking for the most quality/security solution to use Asterisk with G729, ok?! I think this is good for all of us. Thanks. Denis. People, could you PLEASE check first as to who your respons is going to. This double posting that has started recently is getting VERY annoying. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip-info.org
On Wed, 15 Jun 2005 14:37:42 -0400 Huddleston, Robert [EMAIL PROTECTED] wrote: Site down again?? Voip-info.org? or maybe really slow? Up here for me at 15:00 EDT... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATTN: Keith
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, June 11, 2005 11:58 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ATTN: Keith On Saturday 11 June 2005 11:35, Tracy Phillips wrote: That is *precisely* why the RFC is worded should -- it is optional. If the RFC said must then it is required. RFCs are worded very carefully as a general rule. I am just glad everyone doesn't have that attitude about RFCs. I'm not sure I understand -- I'm not making this up, RFCs use must and should very carefully. The latter is a guideline, and the former is a rule. I'm trying to find the link describing this but it's eluding me at the moment. I think this is a VERY good thing; RFCs are like the laws of the internet; they should not be open to interpretation since they define the protocols used to interoperate. -A. Andrew, Did some looking for you. It is contained in RFC 2119, Key words for use in RFCs to Indicate Requirement Levels. Here is an excerpt: Abstract In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. Authors who follow these guidelines should incorporate this phrase near the beginning of their document: The key words MUST, MUST NOT, REQUIRED, SHALL, SHALL NOT, SHOULD, SHOULD NOT, RECOMMENDED, MAY, and OPTIONAL in this document are to be interpreted as described in RFC 2119. Note that the force of these words is modified by the requirement level of the document in which they are used. 1. MUST This word, or the terms REQUIRED or SHALL, mean that the definition is an absolute requirement of the specification. 2. MUST NOT This phrase, or the phrase SHALL NOT, mean that the definition is an absolute prohibition of the specification. 3. SHOULD This word, or the adjective RECOMMENDED, mean that there may exist valid reasons in particular circumstances to ignore a particular item, but the full implications must be understood and carefully weighed before choosing a different course. 4. SHOULD NOT This phrase, or the phrase NOT RECOMMENDED mean that there may exist valid reasons in particular circumstances when the particular behavior is acceptable or even useful, but the full implications should be understood and the case carefully weighed before implementing any behavior described with this label. 5. MAY This word, or the adjective OPTIONAL, mean that an item is truly optional. One vendor may choose to include the item because a particular marketplace requires it or because the vendor feels that it enhances the product while another vendor may omit the same item. An implementation which does not include a particular option MUST be prepared to interoperate with another implementation which does include the option, though perhaps with reduced functionality. In the same vein an implementation which does include a particular option MUST be prepared to interoperate with another implementation which does not include the option (except, of course, for the feature the option provides.) 6. Guidance in the use of these Imperatives Imperatives of the type defined in this memo must be used with care and sparingly. In particular, they MUST only be used where it is actually required for interoperation or to limit behavior which has potential for causing harm (e.g., limiting retransmisssions) For example, they must not be used to try to impose a particular method on implementors where the method is not required for interoperability. So here you are absolutely correct. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Issue with IAXy in Canada?
On Monday 06 June 2005 22:20, Obaid Siddiqui wrote: I tested IAXy with my asterisk server in US, using both DSL. It was working fine. As someone else stated, first try and do a trace route from your friends end to your * box. Once it is confirmed that he can even reach your IP, try running the iax ping tool found here: http://www.voip-info.org/tiki-index.php?page=IAX It is called IAX Ping tool 1.01 for windows. That will confirm that the two ends can talk IAX. Robert Webb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiPSupply Dot Com: Epilogue
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Wolfe Sent: Friday, May 27, 2005 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue Maybe I should my pictures in with me and supermodels. :-) Cheers, -Scott SNIP Only if you have your clothes on and they don't... ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home - mysql login
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Quintin Sent: Thursday, May 26, 2005 1:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] [EMAIL PROTECTED] - mysql login Hi all,whats the root password for [EMAIL PROTECTED] db, to login from the consel? Thx Q That would be a question that should be directed toward the [EMAIL PROTECTED] forum. http://sourceforge.net/forum/?group_id=123387 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages
And to point out additional info, the backlight for the entire phone flashes when the mailbox it is programmed to monitor has a message. MUCH easier to see than a little flashing red light. Robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, May 26, 2005 3:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages Why does it need to be near the garage? Isn't she house trained? And what's wrong with the grandstream bt101? You can program the message key with the access number and code to access the voicemail? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kellner, Peter Sent: Thursday, 26 May 2005 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Looking for inexpensive phone to use with Asteriskwith message light and a button that will let me play new messages I'm wanting to have a phone at home next to the garage door that when my bride comes home, she can see that there is a new message, push a button and have the messages played to her. Otherwise, she will not let me install asterisk on my home line. Can someone suggest relatively inexpensive hardware that will do this for me (us)? Thanks, -Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium FXS modules too fragile?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Wednesday, May 25, 2005 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Digium FXS modules too fragile? SOME people also puzzle over the fact that you can't boil eggs on an electric guitar. Of course you can. Ever heard of Jimi Hendrix? I think he fried his mellon, not egg.. ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rings - How to set number
It is called search the wiki!!! http://www.voip-info.org/wiki-Asterisk+Zap+channels But can only be done for ZAP channels.. On a side note. When are you guys going to fix your QWEST peering out of Richmond??? I would really like to be able to use my Asterisk box during business hours. But a latency jump from 12 ms to over 300 ms on your QWEST link just makes it rather difficult some days.. As of yesterday: Tracing route to coco.nw3c.org [141.153.107.250] over a maximum of 30 hops: 1 *** Request timed out. 221 ms17 ms46 ms 66-173-233-1.serial.cavtel.net [66.173.233.1] 316 ms21 ms13 ms eth6-0-0-100m.core-02.rich.va.cavtel.net [64.83.47.177] 4 404 ms 398 ms 417 ms dca-edge-04.inet.qwest.net [65.125.14.145] 5 431 ms 436 ms 441 ms dca-core-03.inet.qwest.net [205.171.9.97] 6 455 ms 477 ms 469 ms 205.171.209.114 7 485 ms 487 ms 400 ms dcx-edge-02.inet.qwest.net [205.171.251.22] 8 405 ms 383 ms 380 ms 208.46.127.254 9 424 ms 439 ms 453 ms so-7-3-0-0.BB-RTR2.RES.verizon-gni.net [130.81.10.93] 10 412 ms 435 ms 263 ms so-5-0-0-0.BB-RTR2.PHIL.verizon-gni.net [130.81.7.246] 11 317 ms 286 ms 297 ms so-7-0-0-0.BB-RTR1.PHIL.verizon-gni.net [130.81.19.54] 12 312 ms 332 ms 351 ms 130.81.12.58 13 330 ms 340 ms 318 ms a5-0-0-732.g-rtr1.clrk.verizon-gni.net [130.81.5.226] 14 338 ms 352 ms 315 ms 141.153.95.70 15 310 ms 327 ms 324 ms coco.nw3c.org [141.153.107.250] Trace complete. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Huddleston, Robert Sent: Tuesday, May 24, 2005 10:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Rings - How to set number Now for the fun one - change ring pattern?? Like distinctive ringing? Is this supported by asterisk or the end-point -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Umair Bari Sent: Tuesday, May 24, 2005 10:20 PM To: Tim P; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Rings - How to set number exten = 1234,1,Dial(SIP/1234,Number_of_Sec_for_Ringing,tr) Tim P wrote: Maybe this marks me as a real newb but where do I set the number of rings that a phone has before it sends it to voicemail? Also for some odd reason when I ring an extension attached to my sipura 2100 ATA it takes it about 12 seconds to start ringing after I dial it (sits there with dead air on the calling phone). Any idea on these, am I missing some simple configuration switch for either? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New IAXy from Digium
I was just browsing Digium's web site and noticed they are taking orders for the new IAXy. Has anyone purchased and tested one of these yet?? I have thought about buying one for testing, but want to make sure it isn't going to be a flop like the last one. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outbound dialing issue with FXO
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Wednesday, May 18, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Outbound dialing issue with FXO We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a you must first dial a 1 or 0 message back from telco when dialing out standard POTS lines. Try adding a couple of w's in your dial plan to each of the dialed numbers. Two seem to do the best and give about a second before the dial starts to allow for the dial tone to kick in. RObert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voipjet anyone?
On Fri, 13 May 2005 07:59:09 -0700 Wiley Siler [EMAIL PROTECTED] wrote: Almost positive iLBC is not allowed Use uLaw... They do allow for iLBC. From their FAQ page: Codecs. Carriers with primarily business customers should use the G.711 codec when sending VoIP traffic to VoipJet. This ensures that all calls are of the highest sound quality and free from compression degradation. You've paid for VoipJet Tier 1 telco termination and G.711 lets you fully enjoy it. And don't forget some other codecs can add significantly to a call's end-to-end latency, too. If you need to save bandwidth (admittedly very expensive in some parts of the world) then the iLBC codec bundled with Asterisk makes an excellent choice. It's free to use and takes one-fouth the bandwidth of G.711. Another important reason to try iLBC is if there is jitter and packet loss on your network's connection to us, because G.711 really needs ideal conditions to work well. Finally, GSM potentially uses even less bandwidth and CPU processing power than iLBC. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Amp extensions script
Hi, Is there a script in amp for adding the extensions? And can it be modified? When adding a new extension it rewrites all of the information it the context blowing out my additions. You my want to try the AMP forum. Since they are the producers of AMP, they may have a little better info. http://sourceforge.net/forum/?group_id=121515 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line
I was wondering if there was a way to have incoming calls to my PSTN line be transferred to a voip line? I would like to make it so that as soon as the pstn call is recieved it will switch the call to the voip line, thus freeing up the pstn line to get more calls. Kind of like roaming. Tom Why not just call forward everything to your Voip line and then run it through *. Most all providers allow for at least two incoming calls at a time. You would then have your PSTN line free for outgoing only and tie it into a group with your Voip and save some outgoing VoIP minutes. Robert P.S. - This does work very well. It is what I am using at home with my PSTN and myphonecompany.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with TDM400P card
Guys I have a problem getting a TDM400P card to go. It has 4 FXS ports (green modules) and I get this error: [EMAIL PROTECTED] root]# ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Slaves: 06) 6 channels configured. ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? My zaptel.conf reads: [EMAIL PROTECTED] root]# more /etc/zaptel.conf fxsks=1 fxsks=2 fxoks=3-6 loadzone=us defaultzone=us And my rc.local loads: /sbin/modprobe zaptel /sbin/modprobe wcfxo /sbin/modprobe wctdm The 2 100p cards load perfectly but the TDM is not. Any ideas? Could you post the contents of dmesg that are relavant when you load the modules?? Just want to make sure that things are actually loading in the order you have your zapatel.conf set for. It sounds like the cards are not loading in the same order you have the channels configed for. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with TDM400P card
Zttool shows nothing inside thebox. I tried removing the x100 cards, moving the tdm card around, disabled all usb and unnecessary stuff still, kudzu when booting up shows the card and the card shows up on /etc/sysconfig/hwconf but I wonder why it shows 2 of these and I only have 1 tdm400p card with 1 module If I remember correctly, when I installed [EMAIL PROTECTED] and it did its reboot, the TDM was removed from kudzu as it loaded the linux zaptel and you want to load the zaptel obtained from Digium. Try removing it permanantly from kudzu then try loading your modules. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with TDM400P card
-Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Friday, April 29, 2005 1:50 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Problems with TDM400P card How do I remove it from kudzu? I am looking for that now... Sorry it has taken so long to respond, I had some errands to run. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More TDM questions....
Ok, So I am trying to still figure out my ringing issues. This time I grabbed the butt set I own and hooked it into my pots line. With the butt set in monitor mode, I called the pots line so I could actual hear the AC ring. It was a low frequency ringing sound like I am accustomed to. I then hooked same butt set to the TDM and initiated ringing from a SIP extension. I heard the data for the caller ID come across then I heard a much higher pitched ringing sound. Almost like, even though the TDM setup has been verified at 20 HZ, it is ringing at a much higher frequency. Any ideas?? If someone wants, I will try and do a recording of both sounds so you can hear for yourself.. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] start asterisk
On Thu, 28 Apr 2005 16:01:44 + Luz Lopez [EMAIL PROTECTED] wrote: Hi All. I have installed Asterisk on linux Redhat version 9, I follow step by ssstep the installation, my card digium is TDM400P, whith modprobe wcfxs I have load this module. My vonfiguration files are in /etc/asterisk, the file /etc/zaptel.conf hace the folloeing lines: fxoks=1 #fxsks=4 loadzone=us defaultzone=us whit command ztcfg -vv say: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. Nut when I start the asterisk the following message appear Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: manager.c:1478 in init_manager: Unable to open management configuration manager.conf. Call management disabled. Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_agent.c:809 in read_agent_config: No agent configuration found -- agent support disabled Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_mgcp.c:3948 in reload_config: Unable to load config mgcp.conf, MGCP disabled Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_iax2.c:6839 in set_config: Unable to load config iax.conf Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: iax2-provision.c:496 in iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. Apr 28 10:54:53 localhost asterisk[2557]: NOTICE[2557]: chan_skinny.c:2541 in reload_config: Unable to load config skinny.conf, Skinny disabled Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: chan_oss.c:1016 in load_module: XXX I don't work right with non-full duplex sound cards XXX Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: chan_oss.c:257 in sound_thread: Read error on sound device: Resource temporarily unavailable Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_zap.c:6220 in mkintf: Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling Apr 28 10:54:53 localhost asterisk[2557]: ERROR[2557]: chan_zap.c:9155 in setup_zap: Unable to register channel '1' Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: loader.c:345 in ast_load_resource: chan_zap.so: load_module failed, returning -1 Apr 28 10:54:53 localhost asterisk[2557]: WARNING[2557]: loader.c:440 in load_modules: Loading module chan_zap.so failed! Somebody can give me suggestions? Thanks in Advanced, Regards. Did you put the correct settings in zapata.conf as per the wiki?? http://www.voip-info.org/wiki-Asterisk+config+zapata.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving Incoming Calls not working properly on TDM400P with 4 FXO modules
On Wed, 27 Apr 2005 11:24:24 -0600 Andrew Elchuk [EMAIL PROTECTED] wrote: Hi, I have two of the above installed into a server running Asterisk on Debian Linux. Currently, only two phone lines are connected to the system. I had both phone lines plugged into the one card, and it worked fine for dialing out on them, but when receiving incoming calls, only the line plugged into port 1 would answer. I then tried plugging the other line into ports 3 and 4 on the first wildcard and they were no go, and then tried port 1 on the other card and it worked fine for dialing out and answering?? I checked and the cards are not sharing an IRQ with anything else, and in wcfxs.c, #define AUDIO_RINGCHECK 1 is already commented out, as other forums mentioned it could be a problem. What gives here? zaptel.conf fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 loadzone=us defaultzone=us zapata.conf [trunkgroups] [channels] language=en context=main-menu signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 musiconhold=default callerid=asreceived signalling=fxs_ks channel = 1-8 Thanks. How about posting the appropriate lines from dmesg to make sure that all the channels were recognized by the zaptel driver. Then we can go from there. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RJ45 to RJ11?
On Wed, 27 Apr 2005 13:02:56 -0500 (CDT) Paul Shiflet [EMAIL PROTECTED] wrote: I just received my TDM400 card from digium with 2 fxo and 2 fxs interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS phones. How do i interface my POTS phones with this; can i just crimp an RJ45 connection on the end of the phone cord? Paul Better yet... Just plug in the RJ11 and it will work perfect. THe little retaing clip will center it in the RJ45 connector. The pin layouts from the middle out are the same. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
SNIP #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' I am unfamiliar with the Cisco configs but I am just comparing your error message to what you have in the config to make this suggestion. In the error it has user=phone and in your config commented out there is #user_info: phone. What if you tried uncommenting that line and putting in username? It could be that when thatline is commented out, it uses phone by default. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb Sent: Saturday, April 23, 2005 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; List Receiver Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device SNIP #user_info: phone # SIP Configuration File (stop) When the phone tries to register, all I get in the Asterisk console is this: Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804 handle_request_register: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '24.18.147.95' I am unfamiliar with the Cisco configs but I am just comparing your error message to what you have in the config to make this suggestion. In the error it has user=phone and in your config commented out there is #user_info: phone. What if you tried uncommenting that line and putting in username? It could be that when thatline is commented out, it uses phone by default. Robert Actually after getting into the Cisco site it looks like you want a value of none for that. Configures the user= parameter in the REGISTER message. Valid values are: * none-No value is inserted. * phone-The value user=phone is inserted in the To, From, and Contact Headers for REGISTER. * ip-The value user=ip is inserted in the To, From, and Contact Headers for REGISTER. The default value is none. It says the default value is none but you may want to hard code it as it looks like that is not what it is doing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't make my PRI dial out
On Fri, 22 Apr 2005 10:37:32 -0400 Mark Phillips [EMAIL PROTECTED] wrote: I have a full PRI installed on my * machine. I can get inbound calls just fine but can't make outbound ones. Zaptel.conf says; span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf says language=en context=default switchtype=4ess pridialplan=unknown signalling=pri_cpe channel=1-23 echocancel=yes group=1 Your zapata.conf should look like this: language=en context=default switchtype=4ess pridialplan=unknown signalling=pri_cpe echocancel=yes group=1 channel=1-23 You need to move the echocancel and the group above the channel line. The channel line definitions must be above and not below. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't make my PRI dial out
On Fri, 22 Apr 2005 11:48:18 -0400 Mark Phillips [EMAIL PROTECTED] wrote: Nothing happens. I get the same (non)error. I get plenty of output when receiving a call however. Mark Andrew Kohlsmith wrote: On April 22, 2005 10:41 am, Robert Webb wrote: Your zapata.conf should look like this: language=en context=default switchtype=4ess pridialplan=unknown signalling=pri_cpe echocancel=yes group=1 channel=1-23 You need to move the echocancel and the group above the channel line. The channel line definitions must be above and not below. You're right, but that's not his problem. Cause code 0 is no cause code at all; I'd turn on pri debug span 1 output and see what's coming up there. -A. I am grasping at straws here, but have you tried it without the pridialplan command?? According to the wiki, this really doesn't need to be there. pridialplan: Sets an option required for some (rare) switches that require a dialplan parameter to be passed. This option is ignored by most PRI switches. It may be necessary on a few pieces of hardware. Valid options are: unknown, local, private, national, and international. This option can almost always be left unchanged from the default. Default: national. pridialplan=local ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi problem with dialout
SNIP == DISCONNECT_IND PLCI=0x101 REASON=0x3481 == No one is available to answer at this time How changing from CAPI to a zaphfc card will correct this error I don't know, and problably neither does the person who suggested it. REASON 0x3481 is Unallocated (unassigned) number. = Wrong number. -- Dave Cotton [EMAIL PROTECTED] Just as a shot in the dark, but does the telco maybe require 10 digit dialing for ISDN?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error in starting asterisk
On Fri, 22 Apr 2005 01:26:45 +0800 Nathaniel Angelo A. Torres (247talk) [EMAIL PROTECTED] wrote: Hi, here's the content of my Zapata.conf [channels] language=en context=default signalling=em_w faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=0 channel=1-15,17-30 I really don't know what to load into these values. I wanted to use E1R2 of wcte11xp Please help me out. Thanks. Angelo SNIP Nathaniel Angelo A. Torres (247talk) wrote: Hi, I'm receiving this error, please help me solve this. Apr 22 00:12:26 VERBOSE[3735]: == Parsing '/etc/asterisk/phone.conf': Apr 22 00:12:26 VERBOSE[3735]: == Parsing '/etc/asterisk/phone.conf': Found Apr 22 00:12:26 VERBOSE[3735]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Apr 22 00:12:26 VERBOSE[3735]: [chan_zap.so]Apr 22 00:12:26 VERBOSE[3735]: [chan_zap.so] = (Zapata Telephony) Apr 22 00:12:26 VERBOSE[3735]: == Parsing '/etc/asterisk/zapata.conf': Apr 22 00:12:26 VERBOSE[3735]: == Parsing '/etc/asterisk/zapata.conf': Found Apr 22 00:12:26 WARNING[3735]: parse error: No category context for line 10 of zapata.conf Apr 22 00:12:26 ERROR[3735]: Unable to load config zapata.conf Apr 22 00:12:26 WARNING[3735]: chan_zap.so: load_module failed, returning -1 Apr 22 00:12:26 VERBOSE[3735]: == Unregistered channel type 'Tor' Apr 22 00:12:26 VERBOSE[3735]: == Unregistered channel type 'Zap' Apr 22 00:12:26 WARNING[3735]: Loading module chan_zap.so failed! I'm trying to setup E1 R2 for the digium wcte11xp Thanks. Angelo Angelo, try commenting out or deleteling the echotraining cmmand in your config file. From what I can tell on the Wiki, that command is for the X100P card only. And per your error, if I am counting correctly, it states that there is no category context on line 10. I am assuming that line 10 in your config file is the echotraining command. Open your zapata.conf and verisfy that line 10 is indeed the echotraining command. If not, please post back waht is at line 10. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wait in Dial String
On Wed, 20 Apr 2005 10:24:37 -0500
Josiah Bryan [EMAIL PROTECTED] wrote:
On Wednesday 20 April 2005 10:29 am, David Choo wrote:
Dear All,
My boss has placed a requirement for me to forward all
our IDD calls
through a partner's IDD service, which requires us to
call a 8 digit
number, wait for 1 sec, before we send in the foreign
number we're trying
to call.
As I can't find anything on getting the PBX to wait, i'm
only removing the
1st 3 digits (900) and sending in an extra 1 to simulate
the wait. It
works, but not all the time. Is there anyway that I can
place a wait
command here? I'm tried placing w / p but both don't
works. Would like to
seek your kind assistance!
exten = _9001.,1,Dial(Zap/g1/64919669,,D(${EXTEN:3}),)
exten = _9001.,n,Hangup()
Try 'w',
E.g. for my old bridge to BizFon, I had to dial 9, wait,
then the number:
exten = _NX,1,Dial(Zap/g1/9w${EXTEN})
Just put the 'w' between the numbers that you want it to
'wait' at.
-josiah
And as an added tidbit... If I remeber correctly, each w
is about a 1/2 second. So to get a second pause you would
need ww in the string.
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Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Wed, 20 Apr 2005 18:33:44 + Jaime Blanco [EMAIL PROTECTED] wrote: Hi, I just installed the asterisk and the X100P card. I can receive calls from PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone I can dial the demo extension on asterisk pbx. The issue is as soon as I try to dial out 92714756 or another number I received the following message: *CLI -- Executing Dial(SIP/1001-2b93, Zap/g2/2714756) in new stack Apr 20 02:27:40 NOTICE[245776]: app_dial.c:536 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time -- Executing Congestion(SIP/1001-2b93, ) in new stack == Spawn extension (from-sip, 92714756, 2) exited non-zero on 'SIP/1001-2b93' You are getting this because your dial plan is trying to send the connection to ZAP/g2 which is any zap channel in group number 2. If you look in your zapata.conf below, you do not even have a group defined. Zapata.conf is: [channels] callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 immediate=no context=default signalling=fxs_ks channel=1 SNIP Trying cleaning up your extensions.conf so it is a little more readable. I understand that you may just be getting started, but it is really difficult to try and decipher your extensions.conf file the way it is. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Revision question.
See inline responses... On Mon, 18 Apr 2005 10:43:30 -0400 Ian Pattison [EMAIL PROTECTED] wrote: I don't know how everyone else is doing but my woes are continuing. Hardware: Digium TDM400P (REV G according to the silk screening on the board) 2xFX0, 2xFXS purchased in August/September 2004 Dell Precision 420 (PIII-733, 512MB RAM nothing fancy but not doing too much either) Software: Zaptel, Libpri and Asterisk (v1-0) downloaded and re-compiled from CVS today (April 17) SuSE 9.1 (Kernel 2.6.4-52-default) configured as a life-support system for Asterisk only... no other apps running. Here's are my issues: 1. dmesg reports the card as Revision E/F although Rev G visually confirmed (see below) Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device :03:05.0 PCI: Sharing IRQ 11 with :00:1f.3 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) 2. Low ringing voltage still (~44V AC). I have used the boostringer=1 option when loading wcfxs, did I miss something at compile time? The reason you are still seeing low ring voltage is due to the fact that the module is not using the boostringer when it is being loaded. If it were, you would see PCI: Found IRQ 14 for device 00:08.0 Freshmaker version: 71 Freshmaker passed register test Boosting ringinger on slot 1 (89V peak) Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 0 (United States / North America) Notice the line in there about bossting the ringer to 89V peak. I am not seeing that in yours. 3. Rogue on-hook 109V AC voltage (11V AC off-hook) on both FXS ports. I have conformed that it is being generated by the card itself. I repeat, it is not being induced on the wire. After finding it a the wall jack I was able to sample the same 109V AC at the card itself with no cables attached. No clue... 4. Random calls dropped on the FXO ports from both FXS and SIP clients. The drop is usually preceded by a 2-3 second buzzing sound on the line. This occurs with both incoming and outgoing calls. It should be noted that the card is sharing an IRQ with another device (the USB controller to be exact). No matter what slot the card is inserted in it ends up sharing an IRQ. To that end I made sure it was sharing with an unused device (no USB devices attached). Are you using the USB for anything?? If not, turn it off in your BIOS if you have the option and don't even let it load. Looking for help here... Thanks, Ian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to specify channel 1: No such device
On Mon, 18 Apr 2005 11:54:09 -0600 Rich Adamson [EMAIL PROTECTED] wrote: Inline... Hi, I did not find any useful information to configure a Wildcard TDM400P with a FXO card. I've tried everithing, I tried configure it using the cvs and the information from digium page, I tried to configure it using debian packages, I tried to configure with kernels 2.4.30 and 2.6.11, I even switched the mother board (I tried 3 motherboards). I tanks in advace any help you could give me. Best Regards, Gregorio Toscano [EMAIL PROTECTED] The erros are: Apr 15 16:08:37 WARNING[1468]: chan_zap.c:850 zt_open: Unable to specify channel 1: No such device Apr 15 16:08:37 ERROR[1468]: chan_zap.c:6458 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Apr 15 16:08:37 ERROR[1468]: chan_zap.c:9558 setup_zap: Unable to register channel '1' My configuration files are: lsmod Module Size Used byNot tainted wctdm 97248 0 (unused) zaptel214784 0 [wctdm] dmesg (final): Module 0: Not installed Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) najay:/etc# cat zaptel.conf loadzone = us defaultzone = us fxs_ks=1 ^^ that should be fxsks (might also try fxsks=3 since your only module is #3. I don't remember how these are numbered for sure. Don't forget to run 'ztcfg -vv' after the modprobes. That should tell you which channel the fxo module is on. SNIP In actuality, the modules start at 1 and not 0. So you would need fxsks=4 in your zaptel.conf. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problem connecting multiple boxes via IAX2
Senerio multiple * boxes connecting to a central * box with T1 card via IAX2. 1box 1 abd 2 work fine all the time box 3 - after approx 10-15 minutes with no calls - central box with T1 card fails to deliver incoming calls to box 3. Connectivity is good, * exten-2-exten good Ok, I am completely confused about your setup. I thought IAX was used over an IP connection. How do you have it setup by by connecting with a T1 card?? I thought the T1 cards were for incoming voice from the telco or a channel bank. Not used for IP... Signed, Confused?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Revision question.
On Thu, 14 Apr 2005 08:14:37 -0600 Rich Adamson [EMAIL PROTECTED] wrote: I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. I did some more testing today. I called Digium on 4/12 and they suggested some things to try, like different motherboard, switching pci slots, etc.. I did everything they asked, except for the mother switch as I do not have a different one to put in the system at this time. So, after all that, my ringing issue still persists. Too some measurements from bot the card and my POTS line in both the on-hook state and ringing state. I uses a digital multi-meter to make the measurements on both. Here are the results TDM400P Before slot change: On hook idle: 43.8 Volts DC 0Volts AC Ringing: 0Volts DC 56.4 Volts AC After slot change: On hook idle: 48.7 Volts DC 0Volts AC Ringing: 0Volts DC 65.5 Volts AC We can only assume the above represents a fxs module on the card. Correct? I would find it hard to believe that changing slots would cause the on hook DC voltage to change from 43v to 48v. That smells like a funcky voltmeter. Slots should have nothing to do with DC voltage unless the module is simply bad. The AC (ringing) voltage is reasonable, but again it should not have changed simply because of a slot change; again, questionable voltmeter. On my POTS line: On hook idle: 43.8 Volts DC .013 Volts AC Ringing: 50.5 Volts DC 93.9 Volts AC The on hook DC voltage from all US telco's will factually be in the 48v to 52v range. If their central office equipment produced 43 volts, they would have alarms going off all over the place. Their alarms would trigger somewhere in the 46 to 48 volt range. So, that measurement implies the voltmeter is not accurate. The AC (ringing) voltage is well within acceptable telco limits and can range from about 70v to upwards of 105v. Could it bee that from the phone company they retain the DC offset voltage while applying a ring frequency and as it appears on the TDM it shuts off the DC offset when ringing starts. Could this be the issue with those of us in the U.S. having ringing issues with the TDM's?? Doubtful that is an issue. The reason for saying that is the chipset used on the fxo fxs modules was manufactured by Silicon Labs, and those same chipsets are used in other telephony equipment worldwide. Silicon Labs is known for good to excellent products. If their chipsets didn't function correctly, there would have been a large uprising a couple of years ago when those chips were first produced. That hasn't happened, and they don't have a lengthy chip revision history. Asterisk code does not have any control over adding/removing the DC component during ringing, so that's not an issue either. Doubtful that adding/removal the DC component would have any impact on normal telephone sets, however there certainly could be funcky sets that don't like that DC removal. Given the number of postings relative to the TDM card lately, I don't remember exactly what your ringing issue was. Could you remind us without deleting the significant parts of the above? Even though it is long, I will leave everything intact. I have had a few issues with dropped calls when using the FXS to FXO connection. Not sure what the issue is with that. THe main issue I have is with the ringing on the FXS card. I have three differnt brands of phones and all three do the same thing. I might get two or three calls in where everything works fine. But then the next one will cause intermittent ringing one all phones and no data for caller id. I have tried every combination of the phones I have that is possible. From only one of each type hooked directly to the FXS card to hooking the card to my internal house wiring and using various combos of the phones connected. It almost acts like the phones are requiring just a hair more ring voltage to work properly. That is why I was testing the voltage levels. I will try and grab a different meter to test with. The system is a PIII 933MHZ, VIA chipset and has a 500 watt power supply in it. So I don not think it is a power issue from the computer itself. The reason I asked about the DC offset during ringing, is that on the telco side, I noticed that the offset remained even when ringing voltage was applied. On the TDM, it does not. In the manual for the chipset that someone sent me, there is the option to apply a DC offset voltage during ringing. Additionally, the telco side gives the 93 Volts AC when ringing where the TDM is
Re: [Asterisk-Users] Ring two extensions at the same time
On Thu, 14 Apr 2005 07:19:50 -0700 Sean Kennedy [EMAIL PROTECTED] wrote: G.Marshall wrote: Hello, I can not find anything on this, so it may not be possible. I would like to dial one number which then rings at least two extensions at the same time. Not a hunt group, but ringing at the same time as if they were plugged into the same physical port. Does anyone know if this can be done, and if so how? Many thanks, Spencer I know you can do Dial(SIP/101SIP/102) and the like, but you specify you do not want this ( not a hunt group ). How do you want the call to be handled when someone picks up a phone that's ringing? Sean Actually, that is what the is for. It rings all those phones at the same time and not in a hunt group. Using it myself in a dialplan now to ring a zap channel, a sip phone and an outside cell phone. All ring simutaneously and when one phone is answered, all the others quit ringing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Revision question.
On Thu, 14 Apr 2005 10:59:11 -0600 Rich Adamson [EMAIL PROTECTED] wrote: I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. I did some more testing today. I called Digium on 4/12 and they suggested some things to try, like different motherboard, switching pci slots, etc.. I did everything they asked, except for the mother switch as I do not have a different one to put in the system at this time. So, after all that, my ringing issue still persists. Too some measurements from bot the card and my POTS line in both the on-hook state and ringing state. I uses a digital multi-meter to make the measurements on both. Here are the results TDM400P Before slot change: On hook idle: 43.8 Volts DC 0Volts AC Ringing: 0Volts DC 56.4 Volts AC After slot change: On hook idle: 48.7 Volts DC 0Volts AC Ringing: 0Volts DC 65.5 Volts AC We can only assume the above represents a fxs module on the card. Correct? I would find it hard to believe that changing slots would cause the on hook DC voltage to change from 43v to 48v. That smells like a funcky voltmeter. Slots should have nothing to do with DC voltage unless the module is simply bad. The AC (ringing) voltage is reasonable, but again it should not have changed simply because of a slot change; again, questionable voltmeter. On my POTS line: On hook idle: 43.8 Volts DC .013 Volts AC Ringing: 50.5 Volts DC 93.9 Volts AC The on hook DC voltage from all US telco's will factually be in the 48v to 52v range. If their central office equipment produced 43 volts, they would have alarms going off all over the place. Their alarms would trigger somewhere in the 46 to 48 volt range. So, that measurement implies the voltmeter is not accurate. The AC (ringing) voltage is well within acceptable telco limits and can range from about 70v to upwards of 105v. Could it bee that from the phone company they retain the DC offset voltage while applying a ring frequency and as it appears on the TDM it shuts off the DC offset when ringing starts. Could this be the issue with those of us in the U.S. having ringing issues with the TDM's?? Doubtful that is an issue. The reason for saying that is the chipset used on the fxo fxs modules was manufactured by Silicon Labs, and those same chipsets are used in other telephony equipment worldwide. Silicon Labs is known for good to excellent products. If their chipsets didn't function correctly, there would have been a large uprising a couple of years ago when those chips were first produced. That hasn't happened, and they don't have a lengthy chip revision history. Asterisk code does not have any control over adding/removing the DC component during ringing, so that's not an issue either. Doubtful that adding/removal the DC component would have any impact on normal telephone sets, however there certainly could be funcky sets that don't like that DC removal. Given the number of postings relative to the TDM card lately, I don't remember exactly what your ringing issue was. Could you remind us without deleting the significant parts of the above? Even though it is long, I will leave everything intact. I have had a few issues with dropped calls when using the FXS to FXO connection. Not sure what the issue is with that. THe main issue I have is with the ringing on the FXS card. I have three differnt brands of phones and all three do the same thing. I might get two or three calls in where everything works fine. But then the next one will cause intermittent ringing one all phones and no data for caller id. I have tried every combination of the phones I have that is possible. From only one of each type hooked directly to the FXS card to hooking the card to my internal house wiring and using various combos of the phones connected. It almost acts like the phones are requiring just a hair more ring voltage to work properly. That is why I was testing the voltage levels. I will try and grab a different meter to test with. There is a compile-time option to increase the ring voltage. I don't recall the specifics, but its likely in wctdm.c or an associated header file. (As you probably can tell, I don't use the fxs modules on my TDM card.) Yes, I do know about the compile time option and it is enabled. Well, at least the ZAPTEL driver is saying it is. I will reload the driver without the
Re: [Asterisk-Users] TDM400P Revision question.
On Thu, 14 Apr 2005 12:45:44 -0400 Ian Pattison [EMAIL PROTECTED] wrote: My specific issue has to do with ringing on my FXS ports. A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE 2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't get a ring from it). Normally I'd assume that it's a low power issue on the FXS port but with a phone rated at 0.1 REM? I do have some strange voltages though ON-Hook: ~48V DC, 107V AC (this really concerns me...) Off-hook: ~6V DC, ~12VAC (where the hell is this AC component coming from???) Ring: 0V DC, ~45V AC Suffice it to say that electrically this is completely out to lunch... I'd like to throw an oscilloscope on the line to see what's what but I'm having trouble finding one. Thanks, Ian The one test I did not look at was off hook scenario. I will try that tonight. I am also going to call a friend of mine now and see if he has a line tester I can borrow to accurately measure the voltages. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line Presence:
On Thu, 14 Apr 2005 11:42:34 -0700 Sean Kennedy [EMAIL PROTECTED] wrote: Hi all With the recent thread on line presence in asterisk, can anybody tell me if there is a phone out there that supports this? Say I have 20 extensions: Is there any way, hardware based, for me to see the activity on those lines. And for a bonus, is there any way for me to interact with them? Thank you. Sean Is this what you are looking for??? http://www.grandstream.com/y-gxp2000.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P Revision question.
On Mon, 11 Apr 2005 10:54:30 -0400 Robert Webb [EMAIL PROTECTED] wrote: Good morning all.. I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. I did some more testing today. I called Digium on 4/12 and they suggested some things to try, like different motherboard, switching pci slots, etc.. I did everything they asked, except for the mother switch as I do not have a different one to put in the system at this time. So, after all that, my ringing issue still persists. Too some measurements from bot the card and my POTS line in both the on-hook state and ringing state. I uses a digital multi-meter to make the measurements on both. Here are the results TDM400P Before slot change: On hook idle: 43.8 Volts DC 0Volts AC Ringing: 0Volts DC 56.4 Volts AC After slot change: On hook idle: 48.7 Volts DC 0Volts AC Ringing: 0Volts DC 65.5 Volts AC On my POTS line: On hook idle: 43.8 Volts DC .013 Volts AC Ringing: 50.5 Volts DC 93.9 Volts AC Could it bee that from the phone company they retain the DC offset voltage while applying a ring frequency and as it appears on the TDM it shuts off the DC offset when ringing starts. Could this be the issue with those of us in the U.S. having ringing issues with the TDM's?? Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars
On Tue, 12 Apr 2005 15:04:26 -0400 David Brodbeck [EMAIL PROTECTED] wrote: -Original Message- From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED] I don't think the GPL obliges you to give credit to anybody. In fact, I think that's a key difference between the GPL and the BSD license. Actually, the GPL does require credit to be given, at least in the sense that the source code is modified and not necessarily in the advertisement as I understand it. Please take a look at the Developers archive as there is an ongoing discussion about another distro that is for Windows that is also dealing with the GPL issue. I am not completely sure that this falls under the exact same category, but I believe it is really close. Just in this case it would probably deal more with AMP than Asterisk directly. But not having this sellers code, I connot confirm nor deny that this is the case. Can anyone chime in on whether or not this seller must provide the source code for what he is selling?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overwriting config file problem
On Tue, 12 Apr 2005 14:05:06 -0500 mr. barker [EMAIL PROTECTED] wrote: I am using [EMAIL PROTECTED] When I manually add anything to the extensions_additional.conf file it gets rewritten when I add an extension using the web interface I am trying to include the monitor function .. I got that working however it gets deleted when I add something using the web interface I see that it can include = ext-local-custom is this the file that should be used to add custom scripting ? If so where would it be located? It would be located in the /etc/asterisk directory. It is not created by default, that I can see, so you will need to create one and add your cusom config in it. Or else just modify the extensions.conf file and add it there then do an include of your custom section. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Good morning all.. I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Revision question.
Sorry for the initial no subject line. Was in a hurry when I typed this and somehow missed putting it in. Please accept my apologies On Mon, 11 Apr 2005 10:54:30 -0400 Robert Webb [EMAIL PROTECTED] wrote: Good morning all.. I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: Monday, April 11, 2005 5:35 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars Lol, just posted a question to the list that should keep away any bidders. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Charles Osstyn Sent: Monday, April 11, 2005 4:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars Is this ok to sell this on Ebay when they are using open source software? http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItemitem=5766004579 Hoping to have helped, SNIP GARBAGE While I agree that it is ok to sell packages such as this, I do have to inquire as to whether the proper credit has been given for all of the modifications in branding and such from the [EMAIL PROTECTED], AMP, and other projects that this company has used. I have asked for proof of it as I am leary that they have when in there advertisement for the product they clearly state: We have also built in a Web GUI for call accounting records (databased in MySQL). Also builtin is the AsteriskGUI Operator Panel. This panel will show actual realtime call activity, allow an operator or administrator to disconnect calls, create conferences, or initiate calls internally or externally by the click of the mouse from any web browser, internal to your office, or with a change to your router, even external management as well. Making it seem that they are the creators of everything except for the * server. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P Revision
I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. I'm not sure when they came out with the Rev H one. If you look back at the archives over the last year, you'll see several people that have had problems and several more that have not had any problems at all. There does not seem to be any common ground for those that have had problems. Gut feeling (and some rather general comments) tend to suggest the issue is associated with the pci bus, and possibly something to do with the TigerJet pci controller on the card. Best guess is that it has something to do with pci bus timing issues and that probably is somewhat dependent on the exact motherboard in use. Someone posted a note a few weeks ago that essentially said, if your tdm card goes out to lunch (every week or two), dump the tdm registers, and if their all zero's (or 0xff's forget which), then the card should be replaced. The Rev H card _does_ have some additional components on it close to the TigerJet chip, and the fxo modules are now marked as x100 (which they were not marked on the originals). So, something in the design has changed. Hopefully, its an improvement. :) I won't know for another two weeks or so. Best bet is to call digium support and let them walk you through it. It only took about 30 minutes for me last week, and after I described my problem they offered to RMA it without saying anything more, and without logging into my system. Must have been pretty obvious/familiar. OK, I just physically checked my card and the card is stamped with REV H but zaptel insists that it is a REV E/F. Could this be causing some inconsistent issues I have been having. Like voice quality and calls from the FXS port just all of a sudden dropping for no reason?? What can I do to insure that zaptel sees it correctly from now on?? Are there really any code differences that make a difference in how the card is detected?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars
Robert Webb wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: Monday, April 11, 2005 5:35 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars Lol, just posted a question to the list that should keep away any bidders. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Charles Osstyn Sent: Monday, April 11, 2005 4:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars Is this ok to sell this on Ebay when they are using open source software? http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItemitem=5766004579 Hoping to have helped, SNIP GARBAGE While I agree that it is ok to sell packages such as this, I do have to inquire as to whether the proper credit has been given for all of the modifications in branding and such from the [EMAIL PROTECTED], AMP, and other projects that this company has used. I have asked for proof of it as I am leary that they have when in there advertisement for the product they clearly state: We have also built in a Web GUI for call accounting records (databased in MySQL). Also builtin is the AsteriskGUI Operator Panel. This panel will show actual realtime call activity, allow an operator or administrator to disconnect calls, create conferences, or initiate calls internally or externally by the click of the mouse from any web browser, internal to your office, or with a change to your router, even external management as well. Making it seem that they are the creators of everything except for the * server. As not native English speaker I would not see that built in means that they made it, ... for me it just says installed and setup. (compare it with a workshop for your car, if they build into your car a stereo, than they certainly did not create the stereo) All in all, I would not see any problem with selling it on E-bay. ONLY that the picture of Asterisk CDR has been changed to their name, but if they still give the credit to the original it is still ok. The buyer decides if he accepts the price or not. I saw another company, who made fun with the credit!!! The offered a 14 page credit list. Pick what is really inside bye Ronald I see where you are coming from. However, when I read or hear someone say We have built in something into a system, to me that implies they designed it and are taking the credit for it. Yes, that have given a bulleted list of items included in the package. But no where in their advertisement nor in the screenshots that I can see, do they give credit to anyone other than Asterisk for the package. And the entire thing appears to be completely built off of the [EMAIL PROTECTED] project that is certainly a lot easier to setup than what they claim in that writing. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [Asterisk-Dev] Re: Livevoip IAX DTMF troubles
SNIP If you look at a 'iax2 debug' log you will see things like: Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF Subclass: 6 Timestamp: 15832ms SCall: 2 DCall: 00167 [217.160.244.186:4569] which seem to indicate the codes are making to my local asterisk box, or at least are not making it to the IVR system. (I pressed a six) If I change to sipmedia or broadvoice (adding them above) and then dial in via them (both SIP rather than IAX) it all works correctly. thoughts? Cross posted on purpose (since this was posted to -dev and some folks on -users may have an interest). To bring some level of closure to the above and document the actual findings that resulted from my analysis of the OP's problem, the issue with the above is: - LiveVoip (Level3) was not sending the dtmf in iax2 packets, rather the tones were arriving inband. (I used both Ethereal and iax2 debug to verify.) - Since the OP was using iax2 with g711 to LiveVoIP, the tones were arriving at his * box via inband audio, and given the debug shown above (Tx-Frame), * interpreted the inband dtmf and actually sent the tone back to LiveVoip in an outbound iax2 control packet. LiveVoip has acknowledged the problem and is working to resolve it. Its not an asterisk issue. Since LiveVoip indicated the problem exists for about 5% of their DID's, the user could probably ask for a different DID, possibly change to an 800 number, possibly change protocol from iax to sip where dtmf inband is supported, wait for a livevoip fix, etc, etc. Rich Not meaning to be completely off topic here, as I am not completely up to speed on all the protocols, but could this issue that LiveVoIP has acknowledged also be related to the ringback issue with IAX everyone has had?? Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Squeaking / chirping on ZAP Digium TDM400P
On Fri, 1 Apr 2005 16:42:54 -0500 Kellner, Peter [EMAIL PROTECTED] wrote: I've got an asterisk server 1.07 with a Digium TMD400P (2fxo;2fxs). I have it configured to answer an incoming line and transfer to one of the 2 fxs's and it works. I have noticed that on incoming calls I get intermittent squeaks and chirps on the line that I don't get if I plug the incoming line to a PSTN. I'm the only conversation on this hardware and it is a 2.2Ghz P4 with 512Meg RAM. Any ideas on what or how to look for this problem? Thanks, -Peter Could be an interrupt conflict. Do a cat /proc/interrupts on a linux command line and see if the TDM is sharing an interrupt with another device. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk-1.0.7 Build - Serious issues
On Thu, 31 Mar 2005 10:27:24 -0800 hank smith [EMAIL PROTECTED] wrote: isn't [EMAIL PROTECTED] included in 1.07? of asterisk? also I checked the asterisk.org site and saw 1.06 but not the latest when was it put up on asterisk.org? Huh??? Last time I checked, [EMAIL PROTECTED] was an install created by someone else. [EMAIL PROTECTED] is a self installing package that includes Asterisk. Not the other way around.. DO some more reading.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk @ home
On Wed, 30 Mar 2005 08:29:39 -0500 Matt [EMAIL PROTECTED] wrote: Hi, What happened to asterisk @ home 0.7 that the dialout-default macro no longer works? ___ EVERYONE This is NOT the [EMAIL PROTECTED] list group. Please go to: http://sourceforge.net/forum/?group_id=123387 To get help for [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home 0.7 released
On Tue, 29 Mar 2005 12:55:41 -0600 Jeffrey Sharpe [EMAIL PROTECTED] wrote: Thank you! Jeffrey Please do a little searching of the list next time. I just answered this same question about 4 days ago!!! Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing Volume
On Tue, 29 Mar 2005 12:30:31 -0800 Noah Silverman [EMAIL PROTECTED] wrote: hi, We are using PTSN lines connected through the Digium FXO modules for our incomming lines When a caller calls in, the prompts play back at a really high volume. They are a bit distored and fuzzy since they are so loud. Can anybody give me some suggestions?? Thanks, -N Look in the zapata.conf and in the section for the PSTN channels add txgain=-5 or some other variation. This will reduce the volume by a percentage on the transmit out of the FXO port. See www.voip-info.org for the full zapata.conf setup.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP-1.10.007 Released!
On Mon, 28 Mar 2005 12:24:00 -0500 steve szmidt [EMAIL PROTECTED] wrote: On Monday 28 March 2005 12:19, Jon Walsh wrote: How does one downlaod the upgrade only is there the ability to do so from the software or do you need to re-burn an iso or is the iso an upgrade version or the whole install over again? Jonathan You can do it manually, or through a script like this one: wget szmidt.org/asterisk/asterisk-update.sh There's a line which let's you specify a specific release like 1-0-7. You can get both the developer version or stable. Don't forget to chmod 700 /usr/local/sbin/asterisk-update.sh -- Steve Szmidt Steve, I believe you have confused AMP-1.10.007 with ASterisk 1.0.7. This was an inquiry about AMP and not Asterisk. Two different beasts. Jon, The AMP program is an addon to ASterisk. It is not in ISO form. If you go to the AMP web site, http://amp.coalescentsystems.ca/ and download the tar file, you can uncompress it and do an upgrade. The upgrade instructions were included in the original AMP announcement... Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
