Re: [Asterisk-Users] list down?
No problems here. 27 min behind according to your post time. Dean Collins wrote: List doesnt seem to be posting out still active here http://lists.digium.com/pipermail/asterisk-users/2005-June/date.html but not being received by email (time warner is the isp but other emails coming in every few minutes as per normal). Cheers, Dean ___ 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] Tools for effectively manage Asterisk
[EMAIL PROTECTED] wrote: Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards standardizing installed modules, functionalities, tools etc. The wall we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to us); it also suffers from its prerequisites: apache, mysql, php... too much things that should not go in a pbx We tried IPSwitchboard, but it seems only good as a monitor, not as a configuration tool (are we correct or are we missing something?) At this point we are thinking that we better abandon the idea of GUI tools and that we must go on the road of vi editing of .conf files. For now, that is your best option. We would like to understand what other people are using for asterisk management, and to get some suggestion from the community. Someone recently had a thread going about requirements for an Asterisk GUI (not sure what scale they were talking about). I suggested, and do strongly suggest, that Asterisk simply be instrumented for WBEM management and leave the rest out of it. This would allow Asterisk to leverage the many powerful management console applications available, and require the least amount of extra software running on an Asterisk server. WBEM is mature, industry-standard, widely supported, and open. The actual management or client application then can use a standard, open interface, and anyone can develop any app they want to meet their particular needs. http://www.dmtf.org/standards/wbem/ http://www.openwbem.org/ My personal preference for management of an Asterisk installation (note I mean installation and not just server -- the distinction is important), is a drag-n-drop GUI with click-to-edit on system objects and links between, developed from the perspective of the user instead of the perspective of the PBX or the developer. For example, a user or manager does not see extensions, they see phones. Greg ___ 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] AreskiCC - DOES IT REALLY WORK??????
I'm skeptical of any application that relies on register_globals to be onthat's just sloppy, lazy coding, period. John Dunham wrote: Set the global vars for PHP config to on. edit the /etc/php.ini We had the same problem and figured it out. John Dunham GXC Corp. -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] *Sent:* Tuesday, May 31, 2005 10:27 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] AreskiCC - DOES IT REALLY WORK?? Hi all, I am quite disappointed at the application AreskiCC. I have installed everything following the instructions but the thing doesnt want to work. First of all, when I start the index.php page, any name/password logs in. After the login it takes me to a page with a single option LOGOUT We are monitoring the database and it seems like the application doesnt connect to it. Does anybody in this have made this work? Can someone help me please?? Thanks, Robson ___ 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] What does Asterisk need in the way of a GUI?
Please do yourself and everyone involved a favor and use an open and widely used management standard such as WBEM. Start at www.openwbem.org and/or http://sblim.sourceforge.net/ for a jumpstart on WBEM. I had started on this already, but got sidetracked on other projects and never got back to it. I'll be happy to answer any questions you might have on the subject, to the best of my ability. Greg Mitchel Constantin wrote: We are two programmers who are passionate for Asterisk and we will be dedicating the next three months towards programming for Asterisk and would like to get some input from everyone on what they feel Asterisk is lacking or needs based on what is not currently a part of it or available through third parties. Hopefully, by asking up front we won't be wasting our time on something nobody wants or needs. Specifically I am asking in the way of GUI's (web-based or not), not in backend programming as Mark and others have that well under control! Thank you for your suggestions, Mitchel Tom ___ 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] RJ45 to RJ11?
Connect the POTS pair to pins 4 and 5 in the RJ45, and you should be fine (I say this not having looked at the TDM400 specs, but from the perspective of standard wiring practice and the assumption that Mark et al followed same). Greg Paul Shiflet 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 ___ 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] Telephone line installation.
Since you are not setting up an actual PBX in the true sense of the term, the hunting has to be done by the telco if you want to take more than one call at a time on the same number. Cincinnati Bell here locally calls their standard SMB phone service Centrex; Sprint would probably call theirs something else. Greg Manjit Riat wrote: Who is doing the hunting on your main phone number? Or do you not have a main phone number in this install? I guess hunting is only done by the telco right? (or are there any other options to that) ___ 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] Telephone line installation.
Their repsonsibility ends at the demarc. In a house this is the box where either exists an RJ11 phone jack, or a pair of screw terminals. If you have multiple lines coming into a business, you probably will have a 66- or 110-style punchdown block where they terminate their lines (in some cases they might just hand you a multipair cable and wish you luck), and you can connect whatever you want to your side of the block. The tool for 66 and 110 blocks is very inexpensive and available at any Home Depot in the electrical aisle (data/telecom section), as are the RJ11 telephone jacks and wallplates you might need (or the RJ11 plugs and crimp tool, both also available at Home Depot). Total cost for the tools and materials for 6-7 lines (assuming mounting the lines near the punch block): about $40 (you don't need the top-of-the-line contractors' versions of the tools to do just a few of these). Or, you can pay some guy in a hardhat about $80/hour plus/including whatever minimum charge they have for internal wiring, and then about 120% markup on the parts, something they would be very happy to charge you for. ;) Another option is a fractional T1 and just voice channels (in my area, with TWTC, monthly it's about $38/channel plus $80 for the local loop, which for 6-7 business lines might actually be cheaper than 6-7 POTS business lines), and a single T100P card...and you don't get the myriad problems reported on this list involving the TDM cards. Who is doing the hunting on your main phone number? Or do you not have a main phone number in this install? Greg Manjit Riat wrote: We are going to be doing an asterisk install with 5-7 lines. So we are looking to get two TDM04B cards. Now I believe when you get your telco(Sprint, etc.) to install the lines they basically just leave the wires without jacks. Am I right? If so, then can we ask them to install the jacks or would we have to do them ourselves? ___ 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] Cisco phones config over internet
There should not be any, except for the occasional rekeying. Greg On another note... When I observe the activity LEDs for a VPN circuit, it seems to be going full blast without and activity at either end point. Can anyone discuss the bandwidth overhead of a VPN circuit? ___ 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] SER vs Asterisk for SIP
Because SER does not process the RTP stream, it just directs it around. Greg Vikram Rangnekar wrote: Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP 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] POTS Lines
Many people have recommended the Sipura products, and others have recommended a channel bank connected the a PRI card in asterisk box. Does anyone know if it is possible to get Caller ID information into asterisk if using anything other than the TDM WildCard products (using pots lines?) I am interested in moving away from the WildCards but my clients require simple things like Caller ID. Caller ID on POTS lines is transmitted inband, so any of the above can handle incoming POTS CallerID. It gets a bit more complicated with ISDN PRIs and BRIs. Outbound POTS CallerID typically is at the mercy of the telco. HTH Greg ___ 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] OT: Linux routing with T100P problems
What can I use to find out why packets destined for the outside world (via 65.78.109.2) are not being routed? Check with your ISP and make sure they have you set up correctly. I have had issues in the past with that. Fact is, if you can ping the far end, *and packets are returned*, then the problem is not in your setup. If packets were not being returned I would say that ARP was misconfigured for your T1 interface, but since you are getting packets backit's not your problem. Call your ISP. BTW, I can ping the above IP from my machine just fine, so the rest of the world sees your T1 as well. Greg ___ 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: webmin or X admin.
Many web admin interfaces, of varying featureset and completeness, exist for Asterisk. See the Wiki for more: http://www.voip-info.org/wiki-Asterisk+GUI Greg German Aracil Boned wrote: Hello to all I'm new in the list. My name is German. I look very pretty asterisk software, but I don't know if it have any X or web tool to configure or admin. thank to all Best regards German Aracil Boned ___ 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] OT: Integrated Access T1 voice problems -is this possible?
information is from SBC (Southwestern Bell). They claim that if they bring us one T1 and we want to split it 50/50 voice/data (TDM voice and IP data) (not necessarily 50/50 but any combination of voice + data channels), there could be voice quality issues on the TDM voice side when the data portion was pushed (downloading a big file - their words!). It made no sense to me whatsoever, and I think that they are Complete hogwash. I have an IBL from Time Warner Telecom with 4 voice and 12 data and have never had a single problem remotely resembling what they are telling you. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: four wildcards in a single pc
It would be nice for Mark to comment on this design flaw ...? Why so quick to assume it's a flaw? Perhaps it's a compromise. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to connect * to Adtran 600?
http://www.voip-info.org/wiki-Asterisk+Channel+Bank Digium T100P, T1 cable to Adtran T1 port, extensions to Adtran FXS interfaces. Follow the instructions on the Wiki for configuring the T100P both in /etc/asterisk/zapata.conf and /etc/zaptel.conf. Configure the ports on the Adtran per the Adtran manual and you should be off and running. Greg Robert Augustyn wrote: Hi, I have been looking on that unit to be used as source of fxs ports. Now I am not sure how I can get * box talking to it? Thanks for advice. robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to connect * to Adtran 600?
Depends on how many FXO you have. If you have lots, you can use another channel bank with FXO cards in it and another T100P (or a multiport card like a TE4xxP). I haven't looked at the Adtran, but I would imagine that it takes FXO cards as well as FXS. If not, a Carrier Access ADIT 600 does, and works just fine with Asterisk. If you only have a couple of FXO lines, then you can try a TDM400P. Greg Robert Augustyn wrote: Gregory, Thanks for your input I had gove through a lot on that web site and I am still missing someting, I think ...I am not sure where and how I get FXOs configured? If I use T100P to connect to Adtran where am I going to connect the T1 comming from my Telco? Thanks in advance. robert --- Gregory Junker [EMAIL PROTECTED] wrote: http://www.voip-info.org/wiki-Asterisk+Channel+Bank Digium T100P, T1 cable to Adtran T1 port, extensions to Adtran FXS interfaces. Follow the instructions on the Wiki for configuring the T100P both in /etc/asterisk/zapata.conf and /etc/zaptel.conf. Configure the ports on the Adtran per the Adtran manual and you should be off and running. Greg Robert Augustyn wrote: Hi, I have been looking on that unit to be used as source of fxs ports. Now I am not sure how I can get * box talking to it? Thanks for advice. robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux basics
Wow...I am not sure you are ready for the leap you are about to take, if you have no prior Linux experience. At any rate... Best thing for you to do is go to the Redhat/Fedora site, download the FC ISO images, burn them to CD, and install Fedora just like you would install Windows. Boot from CD, follow the instructions, and accept the defaults wherever possible. Redhat has plenty of documentation on their site for installing their distribution. It's fairly painless. I would suggest you install a GUI (KDE or Gnome) as well, and use the GUI text editors to edit text files; VI is about as steep a learning curve there is. Greg Jim Guy wrote: Hello, I am just starting to research Asterisk and I would like to install it on a PC to try out. I have looked around quite a bit but I haven't found much information on the Linux part. I know you need to put Linux on the PC first but what version or flavor of Linux do you recommend? I contacted Red Hat and they had not heard of Asterisk and they said Asterisk is not certified for Red Hat. Are there any Linux installation instructions that you would recommend? If there are any other getting started suggestions, I sure would appreciate it. Thanks, Jim Guy Billings, MT Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard http://us.rd.yahoo.com/mail_us/taglines/spamguard/*http://promotions.yahoo.com/new_mail/static/protection.html. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime Bottom of the page, he just posted the link. Greg Darren Wiebe wrote: I saw that thanks... I was looking for the fields for iax and sip friends, and extensions. If nobody has a list convenient, I will snoop some more and see what I come up with. I thought I saw one once but I have been unable to find it since. Probably mental block on my part. :-( Darren Wiebe [EMAIL PROTECTED] Matthew Boehm wrote: What do you mean? For voicemail, I provided the fields in my instructions. Matthew - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 09, 2004 8:49 PM Subject: Re: [Asterisk-Users] MySQL Does anybody have a list of fields that should be added to tables for a basic setup? I would gladly write a perl script to add them if somebody has a list. Darren Wiebe [EMAIL PROTECTED] Matthew Boehm wrote: Sure. (I really need to write a wiki on this.) You have two choices here before we start. You can use RealTime one of 2 ways: ODBC or direct MySQL. Currently these are the only two supported methods. Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm going to instruct on how to use/install it. The RealTime MySQL driver can be found inside asterisk-addons. Just do the standard make, make install. Now copy asterisk-addons/configs/res_mysql.conf.sample to /etc/asterisk/res_mysql.conf (or whereever your conf dir is). Edit the res_mysql.conf to your liking. Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime config stuff. If you want voicemail, add this line: voicemail = mysql,asterisk,voicemail_users This basically says Please use the RealTime MySQL driver, the database asterisk and the table voicemail_users and bind that to the voicemail family. You can change to your liking: voicemail = driver,databasename,tablename Now go into your mysql server and make the following table: CREATE TABLE `voicemail_users` ( `uniqueid` int(11) NOT NULL auto_increment, `customer_id` int(11) NOT NULL default '0', `context` varchar(50) NOT NULL default '', `mailbox` int(5) NOT NULL default '0', `password` int(4) NOT NULL default '0', `fullname` varchar(50) NOT NULL default '', `email` varchar(50) NOT NULL default '', `pager` varchar(50) NOT NULL default '', `options` varchar(100) NOT NULL default '', `stamp` timestamp(14) NOT NULL, PRIMARY KEY (`uniqueid`) ) TYPE=MyISAM; Put in some rows. Restart asterisk and it should work. Please let me know if it works/doesn't work. -Matthew - Original Message - From: VCI Help Desk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 09, 2004 4:44 PM Subject: [Asterisk-Users] MySQL Does anyone have any instructions for setting up MySQL with the latest CVS? I upgraded from an older version this week and none of the MySQL works now and I believe it's due to the newer Realtime Architecture. I can't find any instructions that explain it very well anywhere. Any help would be appreciated. Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] possible OT - ADIT 600 question
Say I get an ADIT 600 with two FXS8A and one FXO8A from ebay. a. Is it good for Asterisk? A quick look in the Wiki or even a Google search (saving Critch the trouble here) shows that the Adit 600 (with a T100P) is an excellent match for Asterisk: http://www.google.com/search?q=asterisk+adit+600sourceid=mozilla-searchstart=0start=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official http://www.voip-info.org/tiki-print.php?page=Asterisk+hardware b. How do I connect the extensions and lines to it? Do I need a special jack? Can I get that jack in every corner? How do you mean? RJ11? RJ45? You have to install these, and run cable from them to your punchdown block -- which is where the Adit connects to your extensions. c. where can I find help for configuring it? The Carrier Access site is a good place to start. The Wiki covers topics regarding channel banks and Asterisk. d. what kind of backup does it have? Does it need to be reconfigured after a power outage? Carrier Access has these details for the asking. If you don't mind my saying, unless you are pursuing a career in telecom, your questions here might better be served by hiring an Asterisk or telecom contractor to do this for you. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conjuring Kevin Walsh (was: four wildcards in a single pc)
Digium cards need 1000 interupts per card per second due to the lack of onboard buffer. The buffer was left off of the design to keep the design simple and therefore inexpensive. All the cards present 8 bits of data per channel during that interupt and as all telephony is 8000 bits per channel per second 8000/8 = 1000 service needs per second. An interupt is the way hardware requests service. A channel is 64000 bits per second or 8000 8-bit samples. The Digium cards transfer 8 samples or 64 bits per interrupt per channel. This is a good example for the newbies of the list as to why proper formatting and list ettiquitte is important. I made a mistake, it was easy enough for someone to come around behind me and correct the message. We all can make mistakes. Oh whatever, get off it already. Any minimally intelligent amoeba Would have understood the correction regardless of where it occurred in his post. This case is hardly the poster child for bottom-posting vs top-posting. Do you really want to start this nonsense up again? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] possible OT - ADIT 600 question
You can shop around electrical contractors if you want; if you are just interested in installing the jacks and pulling the cable, they all can do that. There are also specialized telecom contractors that do it as well. Either way, it's basic work that the average contractor can do. Business Yellow Pages might even be enough. Greg Shoval Tomer wrote: Many thanks to both you and Greg. I'm currently looking for a consultant in New Jersey that can do the cabling and maybe the ADIT config. We can handle Aserisk pretty decently, but that telco stuff is way too much for us. Know anybody in NJ that can do that for a reasonable fee? Regards, Shoval -Original Message- From: Jonathan Moore [mailto:[EMAIL PROTECTED] Sent: Thursday, December 09, 2004 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] possible OT - ADIT 600 question Also note that if you are going to be doing a lot of work with Adits, that Carrier Access offers free courses on how to setup their channel banks. I went to a day long session on the 600 and found it pretty decent. I had already setup a few of these so I had figured out a lot of the hard stuff, but there is some good info on troubleshooting and error indecations. I second the post below. Adit 600s are a good choice for Asterisk. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Gregory Junker [EMAIL PROTECTED]: Say I get an ADIT 600 with two FXS8A and one FXO8A from ebay. a. Is it good for Asterisk? A quick look in the Wiki or even a Google search (saving Critch the trouble here) shows that the Adit 600 (with a T100P) is an excellent match for Asterisk: http://www.google.com/search?q=asterisk+adit+600sourceid=mozilla- searchstart=0start=0ie=utf-8oe=utf-8client=firefox- arls=org.mozilla:en-US:official http://www.voip-info.org/tiki-print.php?page=Asterisk+hardware b. How do I connect the extensions and lines to it? Do I need a special jack? Can I get that jack in every corner? How do you mean? RJ11? RJ45? You have to install these, and run cable from them to your punchdown block -- which is where the Adit connects to your extensions. c. where can I find help for configuring it? The Carrier Access site is a good place to start. The Wiki covers topics regarding channel banks and Asterisk. d. what kind of backup does it have? Does it need to be reconfigured after a power outage? Carrier Access has these details for the asking. If you don't mind my saying, unless you are pursuing a career in telecom, your questions here might better be served by hiring an Asterisk or telecom contractor to do this for you. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setting the Call Forward Number in Zap?
Is it possible to re-route incoming call on Zap channel of TDM400 FXO card to completely different and external telelephone number via some magic telephone command or signal? So, the Asterisk Zap channel would be cleared off of this call? Like in a scenario when person calls in via PSTN via a Zap channel and listens to IVR menu of Asterisk. Then (s)he presses an extension # and then this call gets redirected to an extenal telephone number outside of Asterisk. And the call to Asterisk is ended. Or I am dreaming out loud? You are describing ISDN features, by which you can signal the CO to route the call elsewhere. You'll need a BRI or PRI to pull this off. POTS lines and basic channelized T1s cannot do this that I am aware of (perhaps POTS lines with *XX features, but I could not tell you for sure). Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there any digital phones that run on asteriskyet?
No, actually he wanted to be able to plug them into a Zap card and have them work, not convert their protocol to/from SIP and so forth. *sniff* I miss my ATT 7406 ;) Greg Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: Are there any digital phones that run on asterisk yet? I'm talking about non-IP phones here... Possibly what you'll want is something that'll convert proprietary digital formats into standards-based formats. These folks have some interesting ideas: http://www.citel.com/products/docs/SIPProdSht_5.19.04.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receptionist Phone
I have a customer interested in an * system, however she wants to ensure that the receptionist phone will display who is on the phone and who is not. It is an office of 10 people, and there are 12 PRI channels available. She is an older lady and does not want to use a web interface. Any suggestions? In other words, she wants to look at a device that indicates hook status of various extensions. I am guessing also that web interface extends to computer interface of any kind. Assuming the above, then why are they interested in Asterisk? If they like the ability to trunk between offices, for example, using inexpensive public Internet connections, Asterisk might have a place in this scenario, but from what you have said here, Asterisk is not the solution for their needs. Square pegs, round holes. They need a basic key system with a receptionist console. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Pocket PC over cell phone connection?
Anyone tried using a pocket pc with sjphone or x-ten over a cell phone connection? Uhh, good luck. Latency, lack of bandwidth... Nice idea, but I would stick with the cell phone when you're on the road. Or wait for WiMax service offering rollouts sometime in 2005. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dont write me again
How about following the very easy-to-understand UNSUBSCRIBE procedure outlined at the bottom of every message from this list? (Oh gawd, I sound like Critchfield now :p ) Greg To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dont write me again
Ohh noo, now you know that it doesn't take someone being mean or even mean spirited to get annoyed at the lack of effort some people exhibit. LOL I know, I could just as easily have phrased it as bkw did. I feel soiled now :-\ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Manager
Not yet. It's under development. Greg Alex Brecher wrote: Is there anything open source out there that has the same or better feature set than Asterisk PBX Manager ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter Sent: Tuesday, November 30, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Manager I have this, it comes as a webmin module. I also got it with the intention of bundling it for clients. It costs $300 , and the license is tied to the NIC. While it wont do EVERYTHING, it will probably be sufficient for the user to set up extensions/phones/menus/voicemail/conferences. One thing that I am not happy with, is that it allows raw editing of the conf files. Gawd help us if a user gets into that lot. I emailed Third lane, and they replied staright away with an address where I could download an evaluation. I'd publish the url here, but there must be a reason why they don't show it on their web site. Oh, and by the way (this from a beginner), I found it by searching on the WIKI Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Time
How can I even tell if there's been a compilation problem? The last line in any make-based build will tell you of an error if one occurred. At this point, type asterisk and then asterisk -r at a command line. The first one starts Asterisk and detaches it as a daemon (background process), and the second attaches to the running server as a CLI (command-line interface). At this point it will depend on your particular situation, in terms of getting it to do what you have in mind. The Wiki is helpful, and searching the archives for similar questions you may have is also helpful. That link provided in another reply works too. Or you can just ask, and hope Steve Critchfield doesn't see your question (JUST KIDDING!!! :p) Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Compatible VSP Service in Ukraine?
If it's person-to-person, why not just use MSN Messenger (or similar) voice communication, or a gaming communications program like TeamSpeak or GameVoice? The codec technology is the same. Jeff Owen wrote: Im sure this might not be the correct place to ask and I have done a Google but I cant seem to find anything that says there is a VSP that will work with * in the Ukraine. I have a friend that lives in Kiev and basically want a phone number there to be able to talk to him and have him call me. If anyone has any information on it and they are willing to share please advise. Thanks, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small PBX setup
My question is would you guys setup an anolog system or VOIP for the phones. There is not a local VOIP provider in our area, so we can not port the 3 pots lines. I would use a 3-port FXO card (for the incoming lines) and a 4-port FXS card (for their existing phones) and just drop-in replace the old system with an Asterisk server. Are they using a proprietary system like a Partner or Merlin, or just 4 multi-line phones? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small PBX setup
They are using a proprietary system, it is not Partner or Merlin, but the phones will not work with other systems. If one of the requirements is to continue using their proprietary phones, you simply are left with getting the old system working. If they are amenable to different phones, you can find Partner systems on eBay for less than $2000 total (for four phones...hell we paid $4400 new 7 years ago for the one we have). If they are really flexible you can try to get them to go with Cisco 79xx, but it does not sound like they are ready to let go of the comfort of a key system, and in that case you are back to square one. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small PBX setup
For this reason alone I find it very hard to even consider a TDM400P card or two - I always suggest a channel bank (Adit600) and T100P, even if the density doesn't require it. I'd love to recommend the TDM card, perhaps this If I could find one with FXO modules I would suggest a used VINA Integrator for price. However, I have not had much luck along those lines. The ADIT 600 would be a good midrange choice, but if they yelped at $2000 for VM, then the total cost of a used ADIT plus the T100P plus an FXO card is closing in on $2000 again. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low Volume WAV Files in Email Attachments
Third, those complaining of low volume in emailed files are usually using a compressed format. In the uncompressed wav format, the volume is effectively doubled by shifting the audio data to the left one bit. This is done at the format level. Of course on playback via asterisk, it checks to see if it needs to shift the audio down and does so. So playback between asterisk recorded wav files should all sound the same on asterisk but isn't the same when played via a normal audio app. The complaints come mainly regarding the emailed attachements, which are WAV49 (MS-GSM) files, which (as far as I can tell) are just justified right and packed into 65 bytes per the IETF I-D. These files are not played back within Asterisk, and honestly, most of what you said above here is rubbish. I just spent more time than I ever cared to spend (including studying the actual GSM codec spec from the ETSI), learning more than I ever cared to learn about GSM (which, btw, if you are concerned about patents, is just as subject to them if you believe Philips' claims), and the difference between uncompressed WAV files (which also suffer from attenuated signal levels) and the GSM and/or MS-GSM files is far far more than just shifting the audio data to the left one bit. There is an issue surrounding the recording of data through Asterisk. That is inarguable. The problem is that no one seems to agree on where to begin looking, so no one has, really. I don't know the origin of the GSM files that make up the Comedian VM system prompts, but they do not suffer from this problem. However, GSM files generated by the VM system, at the least, have a signal attenuation problem to the point that the emailed attachments are unusable, and by most accounts, the phoned-in VM retrieval is barely useful to boot. Not only am I willing to try to track this down, I am furiously taken with the task, because it's a real issue that needs to be addressed, and I do understand that the actual devs have more important things to fix first. That's one of the nice things about open-source, eh? ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OS Choice ?
See the reply below yours. I would hazard a guess that Redhat and SuSE, followed by Debian, are probably the top three (RH and SuSE because of market share, and enterprise server distros thbey have). Greg Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? Best Regards, Alex Brecher -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: Sunday, November 28, 2004 12:37 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OS Choice ? You always have a choice.. Gentoo, Debian... and as always RedHat is NOT an OS. It's a Distro. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Brecher Sent: Sunday, November 28, 2004 11:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OS Choice ? Do I have any other options besides RH 9.0 ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com http://www.successfulhosting.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OS Choice ?
Do I have any other options besides RH 9.0 ? You always have a choice. Most distros provide some form of download for their media. RH/FC, regardless of version, is easiest IMO because of simple ISO image availability. If you really wanted, you could build up a Linux machine based only on a kernel, bootstrap a GCC build, and build everything else you need from there. I've done it before, and that's why I prefer to download ISO images, burn CDs, and install the distro. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low Volume WAV Files in Email Attachments
I'm taking a look at the functions involved to see what the issue is, if anyone cares. One question thowith all of the sound libraries on the planet, ones that write to far mroe open formats, such as MP3, which are just as acceptable as email attachments, and _have_ to be easier to deal with... But I digressI'll dive in and snoop around format_wav_gsm anyway. Greg Steve Prior wrote: Philippe Daoust wrote: I have read several posts regarding this problem but can't find one with a solution... I see the same issue: Voicemails picked up by handset have normal volume, but voicemail sent as a wav attachments in email are so low they are barely usable... Is there a way to fix the volume before they are emailed out? Thanks for any tips. Sign up and join the fun in bug #2023... http://bugs.digium.com/bug_view_page.php?bug_id=0002023 I'd also recommend emailing support at digium - not because you'll get anywhere, but to keep them aware that people are interested in getting this fixed. I've got the same problem and emailed support last week and got very little to encourage me that it's any kind of priority. What are you using for telephone hardware? Your situation is a little different in that you say the voicemails play back at decent volume over the phone - are you testing with voicemails left through a POTS line connection in all cases? I'm pretty new myself, but I've got C skills and am planning to get back to looking over the code myself soon. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
I would like you to name one PBX that does not support this behavior? Every system from Avaya including their Definity, Merlin Legend, Merlin Magix, Partner, and their new IP based PBXes support it, as do those from Mitel, Nortel, InteCom and every other system that I have ever used. A typical example is a manager/admin setup that works as follows: Partner is not a PBX, it is a key system. The Definity PBX does not directly provide key functionality. I can't speak to Merlin, not having used it myself. That said, Asterisk is a PBX like Definity, and should not support this. A FEP for Asterisk, that duplicates the functionality of a key system, should be developed, if it's in high enough demand. Like I said before, I am happy to spearhead the project development if anyone is interested. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
I'm not saying that it would compromise *'s 'PBXness'. But you are comparing products that have DECADES of development and maturity, building on basic features that * is just now getting stable, and that use proprietary hardware to accomplish these features. Kinda my point. I reiterate, if someone wants to help design and implement a separate project to accomplish the key functionality, then I would be happy to work on it. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk gui?
Just wondering how difficult it would be for AMP devs to develop a install wizard or a batch file that can automatically execute the install and download necessary dependencies... until then, I guess I'll be continuing to manually config my asteisk files This requirement is part of the project we are developing. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need some advice
what exactly do you call modules? is it hardware or software? sorry for knowing so little :) I looked on http://www.digium.com/index.php?menu=wildcard_tdm400p2 but I didn't really get it. I have to buy a TDM400 PCI card and then I need add other FXO or FXS cards to this PCI cards? so i need to buy several other cards for it (which wouldn't be a problem, I'm just a little confused)? They are small daughtercards that fit onto the card so that you can customize the number and type of ports the card provides. another newbie question, if i want to use a regular cordless phone, can I plug it into my TDM400 digium card (in addition to my 2 phones lines)? I guess Yes. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GUI
There are many current projects that perform various levels of administration assistance (besides at least two current threads in thist list on the subject ;). You can also find more at the Asterisk Wiki: http://www.voip-info.org Greg Michael Di Martino wrote: I am looking for a good Asterisk GUI to manage my server. Any Suggestions? Regards, Michael DiMartino Director of MIS *The tel^x Group, Inc.* 17 State St, 33^rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp and Asterisk
app_rxfax uses an incorrect structure parameter. Change callerid on line 83 (I think) to cid. Greg Eric Hall wrote: I did that [EMAIL PROTECTED] apps]# patch Makefile.patch patching file Makefile Hunk #1 succeeded at 52 with fuzz 2 (offset 11 lines). Hunk #2 succeeded at 88 with fuzz 2 (offset 19 lines). When back to the top-level and did a make I get this make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Tuesday, November 23, 2004 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Spandsp and Asterisk On Tue, 2004-11-23 at 09:00, Eric Hall wrote: Does anyone have an update patch file to get Spandsp installed? I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 I installed spandsp-0.0.2 when runnig the patch I get patching file Makefile Hunk #1 FAILED at 41. Hunk #2 FAILED at 69. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej Make sure you are trying to patch the Makefile in the apps directory, not the top-level Makefile. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Questions - IVR=Auto Attendant?
No. Auto-attendant is a subset of a class of applications that fall under IVR (interactive voice response). Greg Paul Rodan wrote: Are IVR and Auto Attendant interchangeable terms? They both do the Press 1 for thing. Sales is asking me how to word it and I've always used both terms interchangeably. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp and Asterisk
And this is after you did a make clean, at least in the apps directory? The part about overriding commands doesn't make sense to me... Greg Eric Hall wrote: Still getting errors make[1]: Entering directory `/usr/src/asterisk/apps' Makefile:103: warning: overriding commands for target `app_rxfax.so' Makefile:85: warning: ignoring old commands for target `app_rxfax.so' Makefile:106: warning: overriding commands for target `app_rxfax.o' Makefile:88: warning: ignoring old commands for target `app_rxfax.o' Makefile:109: warning: overriding commands for target `app_txfax.so' Makefile:91: warning: ignoring old commands for target `app_txfax.so' Makefile:112: warning: overriding commands for target `app_txfax.o' Makefile:94: warning: ignoring old commands for target `app_txfax.o' Makefile:115: warning: overriding commands for target `app_dtmftotext.so' Makefile:97: warning: ignoring old commands for target `app_dtmftotext.so' Makefile:118: warning: overriding commands for target `app_dtmftotexto' Makefile:100: warning: ignoring old commands for target `app_dtmftotexto' make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver Schmidt Sent: Tuesday, November 23, 2004 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Spandsp and Asterisk Eric Hall wrote: When back to the top-level and did a make I get this make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# I just fought a battle with spandsp/rxfax and won. My winning strategy can be found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20spandsp hth rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Gift for Mark Spencer
The horse is dead, guys. Let the city workers pick it up now. Thanks. Greg Adam Goryachev wrote: On Wed, 2004-11-24 at 14:17, Kristian Kielhofner wrote: While I usually refrain from the discussions such as this one, this comment has left me utterly disgusted. and while I managed top refrain from this for at least an hour! (ok, so I was eating lunch and don't like typing with sticky fingers :) What mailing list is this? Asterisk-Users. Why do we even HAVE Asterisk? Because Mark Spencer and the folks at Digium (and others) have been busting their butts to provide such a quality piece of software. You wouldn't even have a mailing list to complain on if it weren't for Mark. While this is all true, it is attempting to convert the aversion to spam to the excuse of It is for a good cause/non-profit/someone else who is in need/etc... Sure, I agree, Mark probably deserves more than he gets (I wouldn't know his personal/business financial position) as I know I (and most other small business owners) do. However, it still doesn't suggest that spam'ming people is the correct way to do this. In fact, it is very well known that it is NOT the correct way. Not only would I love to contribute to this, I think the whole idea of a hot tub and it being a surprise is wonderful (assuming Mark can keep his laptop out of the tub). Sure, I agree! Furthermore, how else are you supposed to contact the 8,000 or so people on this list (without posting to it directly, and thus blowing the surprise)? The Batlight from Batman? Well, this is a *very* good question. How are you supposed to contact the 500 people who want to buy a cheap rolex unless you send it to all 50 million email addresses you have on the CD you bought when someone sent their email advertising the CD to 50 million people... Sure, this was one method, however, I can't say it was the right method, nor can I say I know any better method though... I was looking for an excuse to pay more than the suggested $20, and I think I found it... I'm sure Mark will appreciate it. BTW, It would probably have avoided most of these emails if it had at least tried to claim some authority/credibility through being posted somewhere (hidden) on some well-known asterisk websites. eg, the wiki site (but not within the wiki itself where anyone can edit it), the asterisk website, the digium website, the spandsp site, etc... Anyway, what is done, is done, while we can all live and learn, we will never live if we are still discussing the past in 10 years time. Next. Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Gift for Mark Spencer
I think it's time for the list mod to step, no? End this thread? Greg Joe Greco wrote: You, bloody moron. Is not most email unsolicited. Are you familiar with the spam problem? Spam is unsolicited bulk e-mail. It is problematic for any number of reasons. A single unsolicited message may be unwanted, and that's an issue of some sort, but the real problem is when someone feels free to broadcast their message. At the expense of all the recipients. I never asked you to send an email, Are you the person in charge of telling people when they can send messages to the list? If not, then that's irrelevant. Your message is off topic, That's debatable. Your getting rude, Actually, my original reply was quite innocuous. When someone decided to quote it and say I was saying something other than what I was actually saying, I got a bit more explicit. Am I not allowed to correct a misquote? therefor *YOU* are a spammer. Test fails: bulk (UBE). I sent one message. Therefore not spam. By definition. Alternate test fails: commercial (UCE). I'm not selling anything. Therefore not spam. Also by definition. Sorry, I'm not a spammer. Heck, we do all sorts of anti-spam stuff here. You can try to figure out what my .sig means... ... JG ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Patching asterisk for spandsp
I was able to patch the apps/Makefile from the v1-0 branch (use -r v1-0 on the CVS command line) with Steve's patchfile, without issues. I included his patchfile for convenience. Which version of the source are you working with? Worst case, you can just look at the patch file too see what changes it makes (all it does is add build steps for the app_*xfax.c files), and edit the Makefile directly (and put up a patch to the list, out of courtesy, of course. ;) http://laughingmeme.org/archives/001753.html for doing patches, if you aren't familiar). However, app_rxfax.c does need to be changed; there is a patchfile for that too (the callerid member changed to cid in the Asterisk API). HTH Greg Eric Rees wrote: When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- app_rxfax.c 2004-11-22 15:28:42.0 -0500 +++ app_rxfax.c.new 2004-11-03 16:18:21.0 -0500 @@ -83,7 +83,7 @@ FaxReceived, Channel: %s\nExten: %s\nCallerID: %s\nRemoteStationID: %s\nLocalStationID: %s\nPagesTransferred: %i\nResolution: %i\nTransferRate: %i\nFileName: %s\n, chan-name, chan-exten, - chan-callerid, + chan-cid, far_ident, local_ident, t.pages_transferred, --- Makefile.orig 2004-10-02 02:14:37.029411336 +0800 +++ Makefile2004-09-26 23:47:43.0 +0800 @@ -41,10 +41,13 @@ APPS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so ; fi) APPS+=$(shell if [ -f /usr/local/include/zaptel.h ]; then echo app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so ; fi) APPS+=$(shell if [ -f /usr/include/osp/osp.h ]; then echo app_osplookup.so ; fi) +APPS+=$(shell if [ -f /usr/include/spandsp.h ]; then echo app_rxfax.so app_txfax.so ; fi) +APPS+=$(shell if [ -f /usr/local/include/spandsp.h ]; then echo app_rxfax.so app_txfax.so ; fi) + CFLAGS+=-fPIC ifeq ($(USE_POSTGRES_VM_INTERFACE),1) CFLAGS+=-DUSEPOSTGRESVM endif @@ -66,10 +69,16 @@ install: all for x in $(APPS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so +app_rxfax.so : app_rxfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + +app_txfax.so : app_txfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + app_voicemail.so : app_voicemail.o ifeq ($(USE_MYSQL_VM_INTERFACE),1) $(CC) $(SOLINK) -o $@ $(MLFLAGS) $ -L/usr/lib/mysql -lmysqlclient -lz else ifeq ($(USE_POSTGRES_VM_INTERFACE),1) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Patching asterisk for spandsp
Steve Prior wrote: I just ran into this last weekend. I believe that you are using a version of spandsp which is for an older version of Asterisk. The patch file is not part of the tarball; it's a separate download on the site. I had issues with non-1-0 CVS versions; the v1-0 branch worked fine. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Patching asterisk for spandsp
Just for sanity's sake, I went back and read the README on the site again, and it does say: Add the files rxfax.c, txfax.c and dtmftotext.c (the last one has nothing to do with the fax machine, but my makefile patch expects it to be present) You have to grab the dtmftotext.c file as well, which also is not part of the tarball. That could be the problem. Greg Gregory Junker wrote: Steve Prior wrote: I just ran into this last weekend. I believe that you are using a version of spandsp which is for an older version of Asterisk. The patch file is not part of the tarball; it's a separate download on the site. I had issues with non-1-0 CVS versions; the v1-0 branch worked fine. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
I went an nosed around the Bayonne site, and looked at their devel list archivesbased on historical trends, that project looks dormant (it seems to be duplicating what Asterisk does already -- and better). Other projects it links to also look either dormant or missing. I have seriously considered trying to track down the specifics of the Lucent ETR signalling, because I have a Partner key system I would like to use still. Yes, I absolutely could just dump Zap lines into the main system card, but then I lose the ability to signal into Asterisk (i.e. call transfers between the key phones and a remote VoIP user, etc). In short, it's really not meant to work that way and the solution comes up short on many levels. Getting * to speak ETR, however, is a completely different ballgame (although figuring out a powering scheme for the phones introduces another challenge). Greg Leo Ann Boon wrote: Gregory Junker wrote: Is there an open source key system, comparable to *? If there isn't , I'd be happy to work on developing one. It is clear that the need still exists for such a user interface paradigm. Bayonne is supposed to act as a key system, at least that's what was described on the web page. IMHO, it's probably a lot cheaper to re-use the old key system in tandem with Asterisk. A while ago, there was a discussion on this list about the feasilbility of re-using Toshiba key phone handsets with Asterisk. As it is, Toshiba has a Dialogic-style PCI card to support 16 digital handsets. Unfortunately, the cost of the card is US$2500 - much more than buying a brand new key phone. It would be pretty cool, if someone can reverse engineer the protocol(s) used by popular key phone systems. I think it's possible to use HFC in NT mode to drive those handsets. IIRC, most digital phones work along the lines of ISDN phones. Cheers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: changing configuration file
Are you doing this as root? You cannot edit anything in /etc unless you are root (superuser). Greg amna saleem wrote: -- Forwarded message -- From: amna saleem [EMAIL PROTECTED] Date: Thu, 18 Nov 2004 22:26:11 -0800 Subject: changing configuration file To: [EMAIL PROTECTED] hi! I am a beginner at Asterisk and Linux,I am trying to place a call using IAX ,but don`t really know how to chaneg the configuration file.I open the /etc/asterisks directory ,then open the iax.conf file from there but can`t edit it .Can anyone please help me reagarding this issue.How can a configuration file be changed or edited Does the same apply if I want to change the dial plan as well? Amna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk gui?
We are currently working on a WBEM-based management system for Asterisk. If you are familiar with Novell ZenWorks or Microsoft's MMC or the like, you know what I mean. Greg Jim Van Meggelen wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: November 22, 2004 3:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk gui? hello is there a gui that would allow me to configure everything from phones, to extentions, to voice mail to basicly everything that asterisk can do? THAT, my friend, is a tall order. Asterisk is in many ways more like a scripting language than a PBX; certainly in terms of its flexibility. What I mean by this is that the possibilities with Asterisk are so varied that even the most adaptable GUI will on some level have to impose a limitation on it. Perhaps rather than a GUI we should be wanting an IDE (as in Integrated Development Environment, not Intelligent Drive Electronics . . . bloody overlapping acronyms . . . but I digress . . . ). Even some basic syntax highlighting would improve the readability of extensions.conf immensely. Anyone know how to make THAT work in vim? I've hacked one together for UltraEdit that works reasonably well, but that's a Windows editor. I did go to www.voip-info.org and none of the guis I saw there do the trick and the ones that come close aren't downloadable just wanted to see status on this The GUIs that are out there consist of pre-defined interfaces to functions the designers deemed useful. None of them come close to harnessing the true potential of Asterisk (yet). Nevertheless, many of them are extremely interesting and show great promise. In the future, these GUIs may evolve in a manner similar to GNOME or KDE, where the most popular functions have been addressed in a manner acceptable to most users. But even the most comprehensive GUI couldn't hope to keep up with the rapid evolution of Asterisk. The folks at voxbox.ca very generously released their GUI creation to the community. It is known as AMP (Asterisk Management Portal) and is currently the one to watch. Give it six months to a year to build a solid developer community. The AMP list on Sourceforge is VERY active. The folks at Bicom Systems have done some very interesting stuff as well, but it's all closed up. Hard to even tell it's Asterisk. Regardless, for the time being the sage advice is to learn the conf files. There's no better way to properly grasp the staggering potential of Asterisk. Cheers, Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
You should always design an interface around a human being. A hard I could not agree more. Usability is my focus in any software system...including open-source, where it is typically the last thing considered. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
Not all over $500 - a quick search finds: For purposes of replacing a receptionist console with a touch screen (for example, replacing a 6x9 grid of buttons), that would be too small as well. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
Another strong possibility is that after a while, few operators would be willing to continue holding their arms in the air to operate a touch screen. Why would they be holding their arms in the air? You mount the touch panel in the same place at the same angle as the current console... Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
Is there an open source key system, comparable to *? If there isn't , I'd be happy to work on developing one. It is clear that the need still exists for such a user interface paradigm. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Examples of hardware implementations
Asterisk runs on any PC hardware that runs Linux, from that old PII sitting in the closet gathering dust to that 4-way Xeon blade server in a rack, and beyond, and all points in between. Digium has a line of PCI cards that work with Asterisk for T1/E1 lines, ISDN PRIs, analog POTS lines, etc. For more you can visit the Wiki at http://www.voip-info.org Look under the Asterisk section, they have several case studies in there. Greg Philip Trauring wrote: Can some people post some configurations they've implemented when deploying an * system for let's say 25-50 stations and maybe a larger 200 station system? I would assume some kind of chassis with some DSP boards and some kind of system board with a hard drive for running the system and storing the voice mails - obviously I'm interested in specific chassises and boards used and how they've fared. I'm interested to find out what kind of off-the-shelf chassises and boards are available out there and what people have found work well, what issues people have had with heat dissipation, etc. Thanks, Philip Trauring ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
Most customers don't want to be in a new era. They want something they are accustomed to. I don't need any more impediments to making money than I've already got. So if the customer wants a busy lamp, I am going to do my best to give it to them. I agree. This is why engineers do not make good salesmen. What is the size of the current line panel on her desk? I am thinking it might be worthwhile to produce an addon to Asterisk that drives a flat touchpanel that does the same thing as the current solution. Baby steps. If she can use the current mechanical switchboard then she can use this with no real retraining...plus you get the additional benefit of flexibility in configuration (if they end up needing more lines than the current panel supports, this is just a software change). Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
Me and another guy are working on LCD drivers etc for Linux. The thing Including touchscreen? Ideally someone would tell me how to make something either a) seamlessly convert serial/parallel/USB port to TCP and back at the other end, or b) point me to a resource on a simple chip with TCP support that will maybe print out 8-bit packets to an 8-bit pin out. Ideas? http://www.digi.com/products/terminalservers/index.jsp Works terribly well in my experience. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
$400-500 device here. Not very price competitive. I would like to see less than half that. What is the price point you are trying to hit? Any piece of a proprietary telecom system is by nature overpriced to begin with, and receptionist consoles certainly fit into that category. I agree that any touch screen ought to be able to do normal computer graphics. At this point, you are into normal LCD displays with touch capability, which I know retail over US$500 even for smaller ones. And at that point, you are back to doing a double-display on the receptionist computer, and in reality, you could directly run something like that FOP that everyone seems on about (if it fits your needs), since as I understand (never having programmed them myself) that touching a spot on a screen is the same as clicking a mouse there in terms of window-manager events. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: VOIP security on an IAX connection.
Ditto. There's another very clear advantage to OpenVPN over IPsec, and that's the fact that many firewalls are hard to run IPsec through, but OpenVPN, using a single ephemeral UDP link, will work just fine. I believe that the original poster is not concerned with getting it through a Linksys router at home, and that he has a highi degree of control over which hardware is in the trunk path. I could be wrong, but that's what it sounded like to me. I just tried to get it working last night, and I found it (OpenVPN) no easier as a VPN solution than OpenSWAN was, either in server setup and understanding, or client setup and use. My users and myself are running the XP SP2 and Win2K (updated) MS builtin client into the network through one of those hated Linksys routers, with no problems whatsoever. In the end, I decided that I'd rather stick with the open standards, than wait and hope that the OpenVPN proprietary software became a de-facto standard (isn't that what you all hated Microsoft for? But I digress...) For a single point-to-point link, like the poster requested, with Linux on both ends, there is no reason I can tell to go a proprietary route when IPSec works just fine and comes with the 2.6 kernel (or can be fitted on a 2.4 kernel just fine). Greg Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configuration needed
I have my SPA-3000 taking a PSTN line inbound and forwarding it to my Asterisk server after a few rings. I don't hear any dial tone when I do that kind of forwarding. I do it via the dial plan and I also tried it via CFwd SelX Caller/Dest. How are you attempting to do it? I am just starting in the configuration of it and didn't get to finish it yesterday; if I get time today I will get back to it with the suggestions in this thread. Thanks! Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Little off topic
Your actual question then is can the zaptel driver be connected with to a faxgetty? faxgetty expects a serial port, if I am not mistaken. So, can zaptel give me a pseudo-serial port I can use with faxgetty? Not having tried it myself, my expectation would be that it can not. Greg Eric Hall wrote: Here is what I was trying to do Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax server. My question is will a Wildcard T100P work in a Hylafax server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, November 19, 2004 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Little off topic Martin List-Petersen wrote: Citat Eric Wieling [EMAIL PROTECTED]: Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. Might be that i'm wrong on the unchannelized bit, but i don't see, where the analog port will help you ? The guy wants to do Hylafax directly on a T100P w/o Asterisk or Asterisk as middleware, which i don't see working. SpanDSP on the other side works well, but that is basically a softmodem emulation, something Hylafax can't do. I have not seen any applications for spandsp outside Asterisk, yet. *nod* I mist have missed the part about doing it all within Asterisk. I think I wrote that message before my 2nd cup of coffee. An analog port would allow you to plug a modem into the Asterisk box and run Hylafax using that. T-1- Asterisk - Analog - Modem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rtp codec error
I'll stop doing it when Walsh stops posting about it: http://www.faqs.org/rfcs/rfc1855.html (from the RFC) ...Don't wander off-topic, don't ramble and don't send mail or post messages solely to point out other people's errors in typing or spelling. These, more than any other behavior, mark you as an immature beginner. Please Kevin show us your Posting Police ID badge. If you cannot, and I doubt you can, then please do us all a favor and silently ignore posts you don't likeor at least do us the courtesy of taking your trolls off-list. Thanks. Kevin Walsh wrote: Daniel Eboa [EMAIL PROTECTED] wrote: Hello all, And that's as far as I read. You should try re-posting your question without the HTML, without the bold multi-coloured text and without the pointless, out-of-focus graphics. Having done that, perhaps someone will be inclined to read your article and might even be able to help. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to encript SIP comunications?
Linux 2.6 kernel includes IPSec directly, and ipsec-tools can be used to create a secure point-to-point link. OpenSWAN makes use of the kernel IPSec in 2.6, and makes it available in 2.2 and 2.4 kernels. IPSec can use shared keys or x509 certificates within or without a PKI for authentication. OpenVPN has been mentioned as another option, and it uses SSL/TLS for the encryption, and also supports PKI and PSK for auth. Both provide perfect-forward secrecy (PFS) which is important if your client wants past and future communications to remain impossible to decrypt, even with a compromised or subpoenaed private key. Any of the above can be used to encrypt a point-to-point link such as the one you describe. http://www.openswan.org http://www.openvpn.org Greg Linux Dominicana wrote: Hello everybody A given scenario: A client does want to have his own VoIP PBX with Asterisk running, but he ask me. How secure can be the communication among all subscribers? If there're sniffers on the middle or any other listening device on a given netowork. The client is not fictitial, but it main requirement is encription of all point to point comunications for given reasons. Any guidance, products, solutions implementation available and if works is much better. Suggestions are welcome Regards John Fach ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple asterisk process
And Steve provides yet another cordial, extremely helpful reply. Really, friend, does it do *that* much for your ego to step on people in public? If you can't be friendly, just ignore the damn email, no matter how many times the question has been asked. Greg Steven Critchfield wrote: On Fri, 2004-11-19 at 21:08 -0800, Hong Kim wrote: I'm running * on Redhat9 with E100P and ISDN PRI. When I executed asterisk, I could see about 25 asterisk processes. Did someone experienced this? Did you bother using google? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to encript SIP comunications?
http://www.openvpn.org sorry, this should have been http://openvpn.sourceforge.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple asterisk process
I *do* do the same for his posts. Every hundredth one or so, I feel it necessary to let the poor guy or gal who was unlucky enough to ask a simple question that Critch felt the need to answer, that we all were not like that. As a result, that person might even ask another question someday. Instead of being an ass about it, Steve could just as easily said: You can find the information you seek on Google.. It's only a few more words, and far more cordial. Greg Matt Riddell wrote: Gregory Junker wrote: And Steve provides yet another cordial, extremely helpful reply. Really, friend, does it do *that* much for your ego to step on people in public? If you can't be friendly, just ignore the damn email, no matter how many times the question has been asked. Maybe you could do the same for his posts. 8000 messages for first post (poster should have checked google - BTW its ok) 8000 messages for post telling him to search google 8000 messages for post telling Steven to ignore the mail 8000 messages for this post I am currently sending. 1. The first message shouldn't have been sent. (If everybody had ignored him he would have ended up at google anyway) 2. The 2nd, 3rd and 4th messages shouldn't have been sent (an extra 24,000 mails sent out) - and yes I know the 4th is my mail. 3. Maybe a new rule :-) Don't flame, but if someone does, don't respond to it. It will go away if you ignore it. Oh yeah, and sorry for polluting the net with this post...hopefully it'll be the last on the subject. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple asterisk process
This was addressed in a different thread, as I recall, regarding newbie posters, and it was decided, as far as I could tell, that no benefit would be had of such a thing. The feeling was that newbies should benefit from veteran experience too. Steven Critchfield wrote: On Sat, 2004-11-20 at 00:42 -0500, Gregory Junker wrote: And Steve provides yet another cordial, extremely helpful reply. Really, friend, does it do *that* much for your ego to step on people in public? If you can't be friendly, just ignore the damn email, no matter how many times the question has been asked. And what benefit is it to the list for someone who isn't going to be bothered to spend 2 minutes on a path of self enlightenment? 2 minutes might even be more than necessary for one who has spent any time on that path. The level of sophistication needed for running a asterisk box needs someone who at least exhibits more than a second grade education. What grade did your school actually quit spoon feeding you every fact and start asking you to use the tools in front of you to answer questions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple asterisk process
Add to it, my message wasn't a flame but rather a terse comment. Your Never said it was a flame. I said it was in a tone virutally guaranteed to make the guy consider you and everyone on the list to be a conceited jackass. The difference in your perception of your replies (the one I snipped included) and the way you actually come off in public, is the problem. You think you are being terse. You actually thought your post directed the guy to the answer repository. He probably did end up going to Google, but I'll bet he loses interest in Asterisk before long. I guess your work is done here then, right? If they guy isn't an expert, he has no hope of learning, huh? And they wonder why Linux doesn't catch on... Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP security on an IAX connection.
IPSec, especially with PFS, should be all you need. The 2.6 kernel comes with IPSec as part of the kernel, and suites such as OpenSWAN make it quite simple to set up secured links between two endpoints. Given that OpenSWAN is free, I don't see how it gets much more affordable. ;) Keep in mind that all IPSec does is encrypt the link. It does not do routing, it does not provide DHCP address, etc. L2TPD (for Windows clients) and other protocols do that through the encrypted tunnel. Look at the OpenSWAN site for more details: http://www.openswan.org Greg [EMAIL PROTECTED] wrote: Gentlemen and ladies of the Asterisk community. I am considering implementing asterisk based IAX solution for a business that handles a lot of sensitive data. Internal security will be no worse than before as they plan on connecting to their current PBX to handle switching. The asterisk boxes will just handle their trunks between the offices. Other than VPN with a few levels of encryption on the VPN any ideas on other good and affordable ways to implement security on the IAX links? Thanks. [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP security on an IAX connection.
Use iptables to secure your * box and allow traffic only from known servers/hosts. I would say that step one. When you do that then you can use a VPN to make sniffing more difficult. What link do you have between the offices? Is it public internet ? I gathered that he was mostly concerned with man-in-the-middle attacks. It doesn't really matter whether it's a leased line or public Internet at that point; it's not that difficult to break into a CO and tap a line, and if his security needs are high enough, that has to be considered as well. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP security on an IAX connection.
I use an OpenVPN tunnel as well, and IAX over that, and it works dandy. I highly recommend it. It's definately the easiest to configure, understand, and to get across diverse links. It is NAT-friendly, all UDP, etc. In my opinion, OpenVPN is to IPSEC as IAX is to SIP or H323. Does OpenVPN support PFS? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More than 20 FXS
Nahuel Pick up one or more Carrier Access ADIT 600 channel banks off of eBay (they always come loaded down with FXS channels), grab a quad-span Digium TE410P (or TE405P, depends on your PCI capabilities) -- 4 T1/E1 ports, that's 96/120 voice channels theoretical max -- and go to town. :) http://www.digium.com/index.php?menu=hardware_products All of that hardware works splendidly with Asterisk. You can peruse the Wiki for more information on configuring the beasts: http://www.voip-info.org Greg Nahuel Alejandro Ramos wrote: Hi everyone, Could someone tell me if I could make a Asterisk PBX + Digium hardware with more than 20 FXS? If I have a 5 PCI PC, I could only plug 5 TDM40B (4-port FXS) maximun. Is there a solution, a hardware in Digium or something that let me have a PBX with about 50 or 100 FXS and internal phone using a Asterisk PC. Thank you very much... Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More than 20 FXS
http://www.carrieraccess.com/dbfiles/marketing/card_specsheets_pdf/adit_quad_e1_service_card_spec_sheet.pdf That is the quad E1 card that goes into an ADIT 600. The ADIT 600 is a channel bank; it is what converts the voice channels on the E1 to FXS ports via 25-pair connectors on the back of the unit. Each ADIT 600 can support 48 FXO/FXS ports, via the 25-pair connectors on the back. From there you obtain either 66- or 110-style punchdown blocks, prewired with 25-pair tails terminated in the Amphenol (Centronics) connector (also can be found on eBay or at your local electric supply house), and terminate your voice lines from around the office to those punchdown blocks. Greg Nahuel Alejandro Ramos wrote: Thanks for the quick answer. I have seen this cards but how do I convert an E1 to 30 FXS plugs? Sorry about my newbie question about hardware. I have got working an Asterisk with Xlite and ATAs, but I need a lot of internals phones. Thank you again... Nahuel Ramos. On Thu, 18 Nov 2004 14:09:24 -0500, Gregory Junker [EMAIL PROTECTED] wrote: Nahuel Pick up one or more Carrier Access ADIT 600 channel banks off of eBay (they always come loaded down with FXS channels), grab a quad-span Digium TE410P (or TE405P, depends on your PCI capabilities) -- 4 T1/E1 ports, that's 96/120 voice channels theoretical max -- and go to town. :) http://www.digium.com/index.php?menu=hardware_products All of that hardware works splendidly with Asterisk. You can peruse the Wiki for more information on configuring the beasts: http://www.voip-info.org Greg Nahuel Alejandro Ramos wrote: Hi everyone, Could someone tell me if I could make a Asterisk PBX + Digium hardware with more than 20 FXS? If I have a 5 PCI PC, I could only plug 5 TDM40B (4-port FXS) maximun. Is there a solution, a hardware in Digium or something that let me have a PBX with about 50 or 100 FXS and internal phone using a Asterisk PC. Thank you very much... Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More than 20 FXS
You can find them on eBay, but they are few and far between. If you are in a hurry just contact Carrier Access and they would be more than happy to set you up. ;) Greg Bartosz Jozwiak wrote: Well I guess I can ask here. I am looking for FXO ADIT 600 cards. I can find only FXS cards. Bartek - Original Message - From: Nahuel Alejandro Ramos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Sent: Thursday, November 18, 2004 4:19 PM Subject: Re: [Asterisk-Users] More than 20 FXS Thanks for the quick answer. I have seen this cards but how do I convert an E1 to 30 FXS plugs? Sorry about my newbie question about hardware. I have got working an Asterisk with Xlite and ATAs, but I need a lot of internals phones. Thank you again... Nahuel Ramos. On Thu, 18 Nov 2004 14:09:24 -0500, Gregory Junker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Nahuel Pick up one or more Carrier Access ADIT 600 channel banks off of eBay (they always come loaded down with FXS channels), grab a quad-span Digium TE410P (or TE405P, depends on your PCI capabilities) -- 4 T1/E1 ports, that's 96/120 voice channels theoretical max -- and go to town. :) http://www.digium.com/index.php?menu=hardware_products All of that hardware works splendidly with Asterisk. You can peruse the Wiki for more information on configuring the beasts: http://www.voip-info.org Greg Nahuel Alejandro Ramos wrote: Hi everyone, Could someone tell me if I could make a Asterisk PBX + Digium hardware with more than 20 FXS? If I have a 5 PCI PC, I could only plug 5 TDM40B (4-port FXS) maximun. Is there a solution, a hardware in Digium or something that let me have a PBX with about 50 or 100 FXS and internal phone using a Asterisk PC. Thank you very much... Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP security on an IAX connection.
Perfect Forward Security? Yes, OpenVPN can easily be configured for dynamic re-keying at any specified interval and provides all the ciphers that the openssl library supports. I use and highly recommend it. Cool, I will definitely look into it; it wasn't too technically difficult getting OpenSWAN and the XP VPN client communicating, but it certainly could have been easier. The original poster, although I do not know his specifics, seemed like he would be interested in maintaining link security even with a man-in-the-middle attack and a compromised or subpoenaed private key. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Controlling Asterisk from PHP?
Does something like this exist? Dozens of different efforts are underway along these lines. http://www.voip-info.org/wiki-Asterisk+gui Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configuration needed
I want even less than that All I want to do is have the SPA-3000 configured so that it offers up its FXO and FXS ports to Asterisk -- nothing more, I want Asterisk to be the brains. 1) The SPA should hand incoming calls on the FXO to Asterisk. That's all I want. For an interim measure I would like to connect the SPA3K to the Extension ports of a Lucent Partner phone system, and have a few users be able to force call forwarding to select extensions and take calls into the Lucent no matter where they are in the world (VPN and XLite into Asterisk). The final implementation removes the Lucent altogether and uses a T100P and Asterisk as the only PBX on the premises, but first things first. One problem is that the SPA3K only uses two-stage dialing on the FXO -- VoIP2 path, which means any time someone calls the phone system and gets forwarded to a select SPA3K extension, they hear a dial tone. As far as I can tell, there is no way to disable that. You can have it execute a particular dialplan in the SPA3K but the caller gets to hear the digits as they are dialed into Asterisk. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded
Jeff Owen wrote: I have mine working so that all incoming calls are passed directly to * and no user heard any dial-tone or digits, even when the call goes back to the SPA-3k for the Line1 user. Share some config tips? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded
Good deal, I looked at the config site but wasn't sure. The SPA3K I have for testing is 2.09, not sure if that makes a difference. Thanks Greg Jeff Owen wrote: I used http://voxilla.com message board to get mine to work. Phoneboy there knows the spa-3k rather well and has it working with asterisk. He created a wizard to assist people with configuring the system to work together, give it a look see @ http://voxilla.com/spa3kasterisk.php. I just went thru the link on above for the spa3kasterisk.php and it matches my configuration. I'm also running the 2.0.10(GWf) software version on my SPA-3k's. I have been considering upgrading but haven't had a chance yet. -Jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker Sent: Thursday, November 18, 2004 9:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded Jeff Owen wrote: I have mine working so that all incoming calls are passed directly to * and no user heard any dial-tone or digits, even when the call goes back to the SPA-3k for the Line1 user. Share some config tips? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded
Excellent, thanks all. Greg Jeff Owen wrote: I read somewhere that the 2.10 firmware fixed the hearing of digit dialing. When I opened my unit, I upgraded it first then went to configure it so I don't' have any real experience with the hearing the digits. -Jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker Sent: Thursday, November 18, 2004 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded Good deal, I looked at the config site but wasn't sure. The SPA3K I have for testing is 2.09, not sure if that makes a difference. Thanks Greg Jeff Owen wrote: I used http://voxilla.com message board to get mine to work. Phoneboy there knows the spa-3k rather well and has it working with asterisk. He created a wizard to assist people with configuring the system to work together, give it a look see @ http://voxilla.com/spa3kasterisk.php. I just went thru the link on above for the spa3kasterisk.php and it matches my configuration. I'm also running the 2.0.10(GWf) software version on my SPA-3k's. I have been considering upgrading but haven't had a chance yet. -Jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker Sent: Thursday, November 18, 2004 9:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded Jeff Owen wrote: I have mine working so that all incoming calls are passed directly to * and no user heard any dial-tone or digits, even when the call goes back to the SPA-3k for the Line1 user. Share some config tips? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7970G VOIP phones
*puts on flame suit* Bob Willock wrote: I just bought a couple of these Cisco 7970G phones and it seems that they require a SIP image binary file to load when the phone boots and this file updates the firmware of the phone to run in SIP mode. The only problem is that Cisco seems to want to profit from the phone sales and then block you from using the phone by not allowing downloads of the binary files. What a scam!! Has anyone been successful in getting the SIP binary firmware update file? If not how did you get your phone working? Thanks!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7970G VOIP phones
*puts on flame suit* (bah ignore the other prematurely-sent reply) Seriously, though. It's not a scam, it's their business model (which is shared by many many companies). The software license is separate from the hardware. Always has been. You probably should have known that before buying the phone, and a little extra research on this list and the Wiki would have told you that. Just pay the $7 or $10 for the firmware license already, sheesh. Greg Bob Willock wrote: I just bought a couple of these Cisco 7970G phones and it seems that they require a SIP image binary file to load when the phone boots and this file updates the firmware of the phone to run in SIP mode. The only problem is that Cisco seems to want to profit from the phone sales and then block you from using the phone by not allowing downloads of the binary files. What a scam!! Has anyone been successful in getting the SIP binary firmware update file? If not how did you get your phone working? Thanks!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7970G VOIP phones
sorry wasn't paying too close attention too the model number, the other current reply addresses that. Just pay the $7 or $10 for the firmware license already, sheesh. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with an hardware phone: Maximum retries exceeded
The phone has to be registered with Asterisk first. What is your setup in sip.conf for this phone? Greg Michele wrote: Hello, this is my first message on thi ML; I'hava a problem: I have a voismart 101 phone and at the moment of registration or when I make a call,in the asterisk's consolle i can read thi warning: Nov 17 23:09:50 WARNING[2806]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1550 (Non-critical Response) and i cannot use the phone. Do you have already see this message? do you know it? I'v serach with google, but i cannot found any information. Thanks a lot. Michele P. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting
I'll stop doing it when Walsh stops posting about it: http://www.faqs.org/rfcs/rfc1855.html (from the RFC) ...Don't wander off-topic, don't ramble and don't send mail or post messages solely to point out other people's errors in typing or spelling. These, more than any other behavior, mark you as an immature beginner. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting - are we there yet?
So, could we just agree to read around our idiosyncrasies and go back to paying attention to the CONTENT of a message, not its FORMAT? Discarding messages because they're in the wrong format is equal to discriminating against another human being based on outward appearance; be it skin-color, religion, nationality, disability, or -- as often found among engineers -- inability to match shoes and belt. In short, it's ridiculous and utterly inappropriate. Amen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Measuring Bandwidth on T1 into *
router. Will he be able to start downloading/uploading on that bandwidth even though its hooked directly to the Asterisk server? If so, how can I prevent the bandwidth usage but still allow VoIP calls? If you do not route IP traffic over the T1 then there is no way anyone can upload or download. Switch it around.. Company B is the same, however we will sell 1 T1 for voice AND data (at a different price). How do I setup Asterisk to handle both? Asterisk doesn't deal with IP data traffic (beyond the voice-related traffic). It's not a router. Cards such as the T100P have the ability to split data DS0 off of a channelized T1, or a PRI, but again, that's something that has nothing to do with Asterisk. In the first case you are plugging a T1 into the back of Asterisk. I am assuming, since you did not specify, that it's a channelized T1 and the channels are provisioned for voice. Where does the router come into play that you mentioned? In the second case you can use Linux itself as a router if you want. If you don't want, there are plenty of hardware options such as a VINA Integrator that work just fine in handling the dual traffic on a channelized T1. You can pick those up off of eBay for about $50 each, and they route IP and do firewalling and have T1-to-analog ports and everything, in a 1U package. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about remote POTS lines
How remote are the remote offices? Miles? States? Countries? Best of my knowledge, the days of exchanges based on proximity to a particular CO are over, and those numbers (assuming they are in the same area code) often can be routed anywhere. You could also look into having a company like VoicePulse take over the PSTN termination and shoot you a VoIP link to the central * server. Greg Jim Dossey wrote: I have a client who asked me about a situation they have. They have a main office and 3 remote offices. We are installing an Asterisk server at the main office with SIP phones in the remotes. Each remote office only has 1 person. The remote offices currently have a POTS line that has a published number. They want to keep that number. The problem is that they would like to somehow link those remote POTS lines back to the main office, so people in the main office can answer their calls when they are away. They could install an asterisk server in those remote offices and link them back to the main office, but that seems like overkill for a single POTS line. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skype API release
Also interesting comment about Skype possibly being interfaced directly into a gaming solution for online game chats, does anyone know if Asterisk has been licensed to offer something similar? Sounds like an area that could be worth investing in. Sort of overkill considering the popularity of programs like TeamSpeak and Ventrilo, which are suited spcifically for that purpose. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting
folder. No supplier gets a purchase if their people are not properly trained in e-mail communication. My employer spends quite a bit as You are kidding, right? Properly trained? By whose standards? What international commerce committee on email standards published the training regimen of which you speak? This must be a joke, ha ha. It must be nice to make enough money, and have enough time to worry about stuff like this. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Point to Point VOIP
How are the two offices connected? In terms of an Asterisk solution, at a high level you are looking at an Asterisk machine on each end, each of which is connected to the existing office phone system or the local PSTN via TDM cards (or T1/E1 with channel banks, etc). Without more details it's hard to be more specific, but you should get an idea there. Greg - Original Message - From: Jacob Arthur [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 07, 2004 10:59 PM Subject: [Asterisk-Users] Point to Point VOIP I am looking for a setup something like the following. I have two offices, one located in the US and one in Australia. I would like to implement a solution whereby I would install a gateway in each of the two offices. When calls are made to a few numbers in the US, the calls would be routed over the gateway to the one in Australia. The gateway in Australia would dial out to a pre-defined number/set of numbers to complete the call. What is the minimum hardware/software configuration I would need to complete this sort of setup? I am relatively new to the concepts behind VOIP, so any help would be greatly appreciated. Is there anyone with a similar setup to this that has any suggestions/tips? Thanks, Jacob Jacob Arthur, MCP ATS [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting
Dude...you seriously need either to relax, or remove yourself from this and all mailing lists if it is that bothersome to you. I consciously changed my Thunderbird formatting to insert replies at the top. I prefer it. So do many others. Get over it, and yourself. Jesus... Greg Kevin Walsh wrote: Paul Fielding [EMAIL PROTECTED] lazily top-posted: Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting. As you seem to find it difficult to move the cursor on your own, perhaps this utility will help: http://home.in.tum.de/~jain/software/oe-quotefix/ You could install it to fix your broken mail reader - if it's not too much effort. When you bottom post, I need to scroll way down the message to see your response The effort involved is clearly too much for you to handle. Are you really that lazy? If I want to see the source message *then* I'll scroll down, but chances are I've already been reading the thread so this isn't necessary. Your laziness will make life difficult for people who find your followups in a future Google search. Just because you've read the entire thread, doesn't mean that someone else will have done the same next year. Then again, the chance of you posting useful information for someone to find in Google does seem to be a bit remote. just my 2 cents That might be all your time is worth. Others get paid a little more than that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting
If I want to address individual points in turn I happily trim and inline. To say that top posting is unprofessional is simply a meaningless blanket statement; in your opinion it may be, but I doubt it's your main criteria for assessing whether you want to do business with someone or not. Greg Tracy R Reed wrote: On Sun, Nov 14, 2004 at 09:50:28AM -0700, Paul Fielding spake thusly: Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting.When you bottom post, I need to scroll way down the message to see your response, while when you Unfortunately, MS has once again shot us all in the foot with their broken standards. I do realize that MS is to blame for most of the top posting that goes on because that is how they set up their email program. Also note that because of prudent quote trimming you only had to look down three lines from the top of the body of the email to see my comment so I am sure that my message showed up in the first page and you did not have to scroll at all. Professional? That's a matter of opinion, I don't think it's any less professional to top post, it's purely a question of what's convenient for different readers. If you are trying to sell me your product and you want to look good you should bottom post and trim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] $200 AMP documentation bounty
How about this. It's obvious that there is a strong demand for a clean, portable, easy-to-use Asterisk manager application. Apparently those that exist currently fall short in one or more areas, or we wouldn't be having this discussion. Why don't we compile a list of features that people want their manager app to do for them, and finally do one that works, is easy to install, and is intuitive? I'd be happy to manage the project, help code it, design it, whatever it takes. It'll give my decade or so of software engineering experience something to do. I'll start off with the obvious: - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit extensions within the Asterisk PBX software. - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit users within the Asterisk PBX software. - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit channels within the Asterisk PBX software. If anyone is interested in helping continue to describe what you want from an Asterisk system manager, I am perfectly willing to do my best and my part to make it happen. Greg dean collins wrote: http://sourceforge.net/projects/amportal/ AMP is this super manager that was set up a few weeks ago, basically it ties together about 3 or 4 other programs and presents it as a nice GUI display.. Lol, in other words for newbies forget it, I've tried like 3 or 4 times to install it and their documentation just plain sucks, so I'm prepared to pay anyone $200 that can write a step by step guide and then this can be posted online to help out others. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, November 12, 2004 9:42 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] $200 AMP documentation bounty On November 12, 2004 09:27 am, dean collins wrote: There is a $200 bounty for helping document a step by step guide to AMP, anyone on this list interested in making easy money feel free to contact me. What is AMP? Asteirisk Manager Panel? (yes I realize this pretty much shows I am not going after the bounty, :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] $200 AMP documentation bounty
Then forgive me for asking, but why not use one of them if they are easy to use? Obviously one, more or all of them are lacking in some way, including the one over which you are obsessing. Greg dean collins wrote: Greg don't mean to be rude but piss off and start a new email :) There are already a number of easy to use gui's about Xorcom and Asterisk Live. I'm paying for documentation for an already existing product that does a great job but I just haven't been able to install because the How To documentation is shocking. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker Sent: Friday, November 12, 2004 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] $200 AMP documentation bounty How about this. It's obvious that there is a strong demand for a clean, portable, easy-to-use Asterisk manager application. Apparently those that exist currently fall short in one or more areas, or we wouldn't be having this discussion. Why don't we compile a list of features that people want their manager app to do for them, and finally do one that works, is easy to install, and is intuitive? I'd be happy to manage the project, help code it, design it, whatever it takes. It'll give my decade or so of software engineering experience something to do. I'll start off with the obvious: - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit extensions within the Asterisk PBX software. - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit users within the Asterisk PBX software. - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit channels within the Asterisk PBX software. If anyone is interested in helping continue to describe what you want from an Asterisk system manager, I am perfectly willing to do my best and my part to make it happen. Greg dean collins wrote: http://sourceforge.net/projects/amportal/ AMP is this super manager that was set up a few weeks ago, basically it ties together about 3 or 4 other programs and presents it as a nice GUI display.. Lol, in other words for newbies forget it, I've tried like 3 or 4 times to install it and their documentation just plain sucks, so I'm prepared to pay anyone $200 that can write a step by step guide and then this can be posted online to help out others. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, November 12, 2004 9:42 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] $200 AMP documentation bounty On November 12, 2004 09:27 am, dean collins wrote: There is a $200 bounty for helping document a step by step guide to AMP, anyone on this list interested in making easy money feel free to contact me. What is AMP? Asteirisk Manager Panel? (yes I realize this pretty much shows I am not going after the bounty, :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] $200 AMP documentation bounty
Check me if I am wrong, but that's the precise project this thread is complaining about... Personally I do not care for web-based administration of real-time systems, regardless of how easy they are to develop. That's just me. At Dean's request I am removing the splinter topic to a new thread, anyway. Greg Geoff Nordli wrote: Those of you that are interested in building a GUI interface should look at the AMP project hosted at sourceforge: http://sourceforge.net/projects/amportal/ I think it would be great to have a unified GUI for Asterisk. Have a great day! Geoff [EMAIL PROTECTED] scribbled on : Then forgive me for asking, but why not use one of them if they are easy to use? Obviously one, more or all of them are lacking in some way, including the one over which you are obsessing. Greg dean collins wrote: Greg don't mean to be rude but piss off and start a new email :) There are already a number of easy to use gui's about Xorcom and Asterisk Live. I'm paying for documentation for an already existing product that does a great job but I just haven't been able to install because the How To documentation is shocking. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker Sent: Friday, November 12, 2004 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] $200 AMP documentation bounty How about this. It's obvious that there is a strong demand for a clean, portable, easy-to-use Asterisk manager application. Apparently those that exist currently fall short in one or more areas, or we wouldn't be having this discussion. Why don't we compile a list of features that people want their manager app to do for them, and finally do one that works, is easy to install, and is intuitive? I'd be happy to manage the project, help code it, design it, whatever it takes. It'll give my decade or so of software engineering experience something to do. I'll start off with the obvious: - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit extensions within the Asterisk PBX software. - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit users within the Asterisk PBX software. - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit channels within the Asterisk PBX software. If anyone is interested in helping continue to describe what you want from an Asterisk system manager, I am perfectly willing to do my best and my part to make it happen. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)
Starting a new thread at Dean's request. Why yet another project proposal? Because the majority of those I have seen so far are web (PHP)-based, which often presumes UNIX admin experience. The Web paradigm may be easy to access and easy to develop, but in terms of administration it is limiting (IMO anyway). Many projects also require to run on the Asterisk server, which is often undesirable. And none of them have the ease of setup, configuration and operation that it seems most users are demanding. So, if anyone is interested, I am suggesting particularly a standalone, cross-platform project that is simple to install, configure, operate and manage. It should operate with or without a database. It can leverage existing projects, but it must not have the existence or installation of those projects as prerequisite. In other words, if this project uses another project's code, it must also include the installation and configuration of that project in this one's installer. (cut and paste from the $200 AMP thread) It's obvious that there is a strong demand for a clean, portable, easy-to-use Asterisk manager application. Apparently those that exist currently fall short in one or more areas, or we wouldn't be having this discussion. Why don't we compile a list of features that people want their manager app to do for them, and finally do one that works, is easy to install, and is intuitive? I'd be happy to manage the project, help code it, design it, whatever it takes. It'll give my decade or so of software engineering experience something to do. I'll start off with the obvious: - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit extensions within the Asterisk PBX software. - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit users within the Asterisk PBX software. - The app must allow the administrator, either graphically or via menu- and dialog-driven GUI, to add, remove and otherwise edit channels within the Asterisk PBX software. If anyone is interested in helping continue to describe what you want from an Asterisk system manager, I am perfectly willing to do my best and my part to make it happen. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
