Re: [Asterisk-Users] Grandstream Firmware 1.0.5.16 Attended Transfer
Where can you get version 18? I only see v. 16 on www.grandstream.com/Firmware. On Wed, 2004-11-24 at 15:15 +0200, Altus Snyman wrote: There is a version 18! ___ 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] Problems with udev on FC3
I've been testing * on FC3. I have everything compiled and installed. However, when I do 'modprobe wcfxo' (I have an X100P clone), I get the following in /var/log/messages: Nov 24 10:23:40 jfd wait_for_sysfs[3366]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zaptimer' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <[EMAIL PROTECTED]> Nov 24 10:23:40 jfd wait_for_sysfs[3368]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zapchannel' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <[EMAIL PROTECTED]> Nov 24 10:23:40 jfd wait_for_sysfs[3370]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zappseudo' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <[EMAIL PROTECTED]> Nov 24 10:23:40 jfd wait_for_sysfs[3372]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zapctl' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <[EMAIL PROTECTED]> Nov 24 10:23:44 jfd wait_for_sysfs[3377]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zap1' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <[EMAIL PROTECTED]> I added the udev rules as described in README.udev. Any ideas? TIA ___ 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] Coverting Cisco 7960 to SIP
On Wed, 2004-11-17 at 22:34 +0100, Håkan Persson wrote: Hi! I just bought a Cisco 7960G and I want to convert it into a SIP phone. All information to do this seems to be at http://cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml but the firmware download link on the page takes me to a page that informs me that "There are currently no files for this type." I have registred at the site but this doesn't seems to help. Where is the needed software located? Best, Håkan Try this link. You have to register with cisco.com first, but registration is free. http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 ___ 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 to display which extensions are in use on the phone's display?
On Wed, 2004-11-17 at 16:03 -0500, Andrew Kohlsmith wrote: Perhaps a better idea is to have extension appearances (similar to a CAP module) and a number of parking slot appearances. This, IMO, is far more worthy of development work, as it scales much better. Call pickup still works from any phone in the pickupgroup, so picking up a ringing line should not be any different. I vote for parking slot appearances, so users don't have to remember an extension number where a call is parked. They can just press the flashing button. ___ 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] AstLinux 0.1.3 released
On Wed, 2004-11-17 at 14:51 -0600, Kristian Kielhofner wrote: > Jim Dossey wrote: > > > On Wed, 2004-11-17 at 13:18 -0600, Kristian Kielhofner wrote: > > > >>Hello everyone, > >> > >>Since releasing my very beta, test version of AstLinux almost two weeks > >>ago, there have been over 300 downloads from all over the world, with > >>over 100 in the first 24 hours. I was very surprised by the response, > >>and I have come up with a new version with many, many more features and > >>software. I encourage everyone that downloaded the original to download > >>this new release, as it fixes many problems that some were experiencing. > >> > >>http://www.krisk.org/astlinux/ > >> > >>Thanks. > > Can you legally redistribute the Digium G.729 code? > > I have included the binary, it cannot be used until you license it with > a valid G.729 license code from digium. Basically, I downloaded the > applicable G.729 Asterisk module and register program from their FTP > site. I have seen other people put it up places, so I didn't see why I > could not. (That doesn't make it right, of course). I would love for > someone to tell me why I can't redistribute it, and why I should remove > it. It would be just as easy to write a script to go and fetch them > both from ftp.digium.com... > > -- > Kristian Kielhofner Ok. I was just wondering. I didn't know that you could just download the code from the Digium ftp site. I've never used or licensed it before, but I will be soon. ___ 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] AstLinux 0.1.3 released
On Wed, 2004-11-17 at 13:18 -0600, Kristian Kielhofner wrote: Hello everyone, Since releasing my very beta, test version of AstLinux almost two weeks ago, there have been over 300 downloads from all over the world, with over 100 in the first 24 hours. I was very surprised by the response, and I have come up with a new version with many, many more features and software. I encourage everyone that downloaded the original to download this new release, as it fixes many problems that some were experiencing. http://www.krisk.org/astlinux/ Thanks. Can you legally redistribute the Digium G.729 code? ___ 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 emulate a multiline phone in Asterisk
On Tue, 2004-11-16 at 09:19 +0100, Peter Svensson wrote: On Mon, 15 Nov 2004, Jim Dossey wrote: > I have a client who currently has a Toshiba PBX. We are trying to > replace it with an Asterisk system. One of the features that they have > on their current PBX is the ability to select a POTS line by pressing a > button on their phones. They have 10 POTS lines and they can select any > line by just pressing the corresponding button on their phone. They can > also join an existing call by just picking up the handset and pressing > the button for the call they want to join. This isn't like a conference > call because you have one outside line coming in and 2 or more people > inside the office on that call. This is commonly referred to as a "key system". Asterisk is not terribly well suited as a key system since it is a pbx. The difference is (leaving out all the subtleties) that a key system is based on the concept of pstn lines while the pbx is based on the concept of extensions. Even the whole pstn cloud can be thought of as extensions. You could possibly make something similar if you really _really_ want to using agi scripts and redirects through the manager port to a meetme conference. Before going down that road I suggest you first consider if your client is best served by this solution or if it is just an artifact of their old system. Peter Thanks for that answer. That clears up a lot of questions that I had. This isn't really a functionality problem. It's more a problem of breaking the users of their old habits and switching them to a new platform. ___ 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] How to emulate a multiline phone in Asterisk
I have a client who currently has a Toshiba PBX. We are trying to replace it with an Asterisk system. One of the features that they have on their current PBX is the ability to select a POTS line by pressing a button on their phones. They have 10 POTS lines and they can select any line by just pressing the corresponding button on their phone. They can also join an existing call by just picking up the handset and pressing the button for the call they want to join. This isn't like a conference call because you have one outside line coming in and 2 or more people inside the office on that call. I've looked through documentation and searched the Wiki, but I can't find a way to do that in Asterisk? I could let them select a POTS line by setting up an extension on each of their 10 lines, and letting them dial that extension. But can you do that when someone else is already on a call on that line? Also, related to that question: Are there any SIP phones out there that have 10 programmable buttons that could be programmed to select one of their 10 POTS lines - so they don't have to dial the extension number or some key sequence like "*99"? TIA Sorry if this has been discussed before. I'm fairly new to Asterisk. ___ 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] Question about remote POTS lines
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
Re: [Asterisk-Users] Auto dialout
On Mon, 2004-11-15 at 15:23 -0700, Kyle Hagan wrote: Im trying to use the auto-dial out in asterisk and having problems I get the following errors: Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:182 apply_outgoing: Unknown keyword ' Channel' at line 5 of /var/spool/asterisk/outgoing/1.call Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:182 apply_outgoing: Unknown keyword ' MaxRetries' at line 6 of /var/spool/asterisk/outgoing/1.call Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:182 apply_outgoing: Unknown keyword ' RetryTime' at line 7 of /var/spool/asterisk/outgoing/1.call Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:182 apply_outgoing: Unknown keyword ' WaitTime' at line 8 of /var/spool/asterisk/outgoing/1.call Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:182 apply_outgoing: Unknown keyword ' Context' at line 13 of /var/spool/asterisk/outgoing/1.call Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:182 apply_outgoing: Unknown keyword ' Extension' at line 14 of /var/spool/asterisk/outgoing/1.call Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:182 apply_outgoing: Unknown keyword ' Priority' at line 15 of /var/spool/asterisk/outgoing/1.call Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:191 apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/1.call Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:301 scan_service: Invalid file contents in /var/spool/asterisk/outgoing/1.call, deleting Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:349 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/1.call' Kyle >From the error messages it appears that you have a space at the beginning of each line. ___ 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] OT: Grandstream problems
On Fri, 2004-11-12 at 16:16 -0600, Rich Adamson wrote: > > I just got a Grandstream 100 yesterday. I plugged it into my network and after a few > seconds it showed the correct > > time and date. After that I tried to access the built-in web server. After several > tries I finally got the login screen. That > > took about 10 tries before it displayed anything. Then I entered the default password > 'admin' and clicked the Login > > button. After that I got nothing. The phone still seemed to be working - I could press > the Menu button and the menu > > would come up and I could scroll through it. > > > > I also tried changing out the ethernet cable, and plugging it into different ports on my > Linksys switch (BEFSX41), but > > nothing works. > > > > Do I just have a bad unit? > > Be carefull with assuptions regarding the Linksys. There were several > different models (versions) of that hardware, and the early models > did not support 100 meg ethernet worth a darn. Some packets make > it through and a large number did not. Change the port speed on > your phone to 10 meg (if you can) and try again. (I seen your post > on the ping packet loss and that sort of reminded me of problems > seen before.) > > Rich > > Thanks for the feedback (and the reply from Dave Cotton as well). I'm going to try the phone on a completely > different network and see what happens. > > For the record, the problem was an incompatibility between the Budgetone and the Linksys ethernet ports. I plugged a > KTI 10Mb hub between the Linksys and the Budgetone and it started working. Don't plug a Budgetone 101 into a > Linksys BEFSX41. OR, downgrade the Budgetone to 10 meg, or, replace the linksys with a newer one. How do you downgrade the Budgetone to 10Mb? I don't see anything on the configuration page to do that. Also the specs on the Budgetone say it is a 10Base-T port. ___ 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] OT: Grandstream problems
On Fri, 2004-11-12 at 12:45 -0500, Jim Dossey wrote: On Fri, 2004-11-12 at 10:55 -0600, Rich Adamson wrote: > I just got a Grandstream 100 yesterday. I plugged it into my network and after a few seconds it showed the correct > time and date. After that I tried to access the built-in web server. After several tries I finally got the login screen. That > took about 10 tries before it displayed anything. Then I entered the default password 'admin' and clicked the Login > button. After that I got nothing. The phone still seemed to be working - I could press the Menu button and the menu > would come up and I could scroll through it. > > I also tried changing out the ethernet cable, and plugging it into different ports on my Linksys switch (BEFSX41), but > nothing works. > > Do I just have a bad unit? Be carefull with assuptions regarding the Linksys. There were several different models (versions) of that hardware, and the early models did not support 100 meg ethernet worth a darn. Some packets make it through and a large number did not. Change the port speed on your phone to 10 meg (if you can) and try again. (I seen your post on the ping packet loss and that sort of reminded me of problems seen before.) Rich Thanks for the feedback (and the reply from Dave Cotton as well). I'm going to try the phone on a completely different network and see what happens. For the record, the problem was an incompatibility between the Budgetone and the Linksys ethernet ports. I plugged a KTI 10Mb hub between the Linksys and the Budgetone and it started working. Don't plug a Budgetone 101 into a Linksys BEFSX41. ___ 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] OT: Grandstream problems
On Fri, 2004-11-12 at 10:55 -0600, Rich Adamson wrote: > I just got a Grandstream 100 yesterday. I plugged it into my network and after a few seconds it showed the correct > time and date. After that I tried to access the built-in web server. After several tries I finally got the login screen. That > took about 10 tries before it displayed anything. Then I entered the default password 'admin' and clicked the Login > button. After that I got nothing. The phone still seemed to be working - I could press the Menu button and the menu > would come up and I could scroll through it. > > After trying that for awhile, I decided to try a firmware upgrade. So I entered the grandstream tftp server address into > the phone (168.75.215.189) and it did seem to do the upgrade. At least I think it did. The "P" entry in the revision > screen shows 1.0.5.16. There are lots of other revision numbers in there that I'm not sure about. > > I also tried changing out the ethernet cable, and plugging it into different ports on my Linksys switch (BEFSX41), but > nothing works. > > Do I just have a bad unit? Be carefull with assuptions regarding the Linksys. There were several different models (versions) of that hardware, and the early models did not support 100 meg ethernet worth a darn. Some packets make it through and a large number did not. Change the port speed on your phone to 10 meg (if you can) and try again. (I seen your post on the ping packet loss and that sort of reminded me of problems seen before.) Rich Thanks for the feedback (and the reply from Dave Cotton as well). I'm going to try the phone on a completely different network and see what happens. ___ 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] OT: Grandstream problems
On Fri, 2004-11-12 at 10:19 -0500, Jim Dossey wrote: I know this is off topic a little, but I thought I would ask the people that use this stuff everyday instead of trying to talk to some tech support guy. I just got a Grandstream 100 yesterday. I plugged it into my network and after a few seconds it showed the correct time and date. After that I tried to access the built-in web server. After several tries I finally got the login screen. That took about 10 tries before it displayed anything. Then I entered the default password 'admin' and clicked the Login button. After that I got nothing. The phone still seemed to be working - I could press the Menu button and the menu would come up and I could scroll through it. After trying that for awhile, I decided to try a firmware upgrade. So I entered the grandstream tftp server address into the phone (168.75.215.189) and it did seem to do the upgrade. At least I think it did. The "P" entry in the revision screen shows 1.0.5.16. There are lots of other revision numbers in there that I'm not sure about. I also tried changing out the ethernet cable, and plugging it into different ports on my Linksys switch (BEFSX41), but nothing works. Do I just have a bad unit? TIA I should add that it was a Grandstream Budgetone 101. Also, I can ping the phone, but I get about 10% ping failures. ___ 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] OT: Grandstream problems
I know this is off topic a little, but I thought I would ask the people that use this stuff everyday instead of trying to talk to some tech support guy. I just got a Grandstream 100 yesterday. I plugged it into my network and after a few seconds it showed the correct time and date. After that I tried to access the built-in web server. After several tries I finally got the login screen. That took about 10 tries before it displayed anything. Then I entered the default password 'admin' and clicked the Login button. After that I got nothing. The phone still seemed to be working - I could press the Menu button and the menu would come up and I could scroll through it. After trying that for awhile, I decided to try a firmware upgrade. So I entered the grandstream tftp server address into the phone (168.75.215.189) and it did seem to do the upgrade. At least I think it did. The "P" entry in the revision screen shows 1.0.5.16. There are lots of other revision numbers in there that I'm not sure about. I also tried changing out the ethernet cable, and plugging it into different ports on my Linksys switch (BEFSX41), but nothing works. Do I just have a bad unit? TIA ___ 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] External call initiation
On Tue, 2004-11-09 at 18:46 -0700, Damon Estep wrote: > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED]] On Behalf Of Jim Dossey > Sent: Tuesday, November 09, 2004 3:21 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] External call initiation > > I have a client who needs to be able to initiate an outgoing call from a > legacy Unix application. They use a legacy accounting system on a Unix > system using CRT as a telnet client. They have the ability now to have > the Unix system auto-dial their phone to place a call. For example, > they can pull up a customer or vendor record in CRT. The application > uses the CRT "pass-thru" option to pass a modem dial string through > their PC to a modem plugged into the PC's serial port. The modem is > attached to their telephone so that they lift the receiver, pick an > outgoing line, and the modem dials the number. > > I need to be able to do that using Asterisk and a SIP phone (or an ADSI > phone). Any ideas? Is this possible? > > I was thinking that maybe there would be a way for the Unix box to > command the Asterisk box to connect to the SIP phone, and place the > outgoing call, and then join those 2 connections together. > > TIA > Have your unix app drop a call file in the * queue via ftp, here is more info http://www.voip-info.org/wiki-Asterisk+auto-dial+out Thanks! That was exactly what I was looking for. ___ 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] External call initiation
I have a client who needs to be able to initiate an outgoing call from a legacy Unix application. They use a legacy accounting system on a Unix system using CRT as a telnet client. They have the ability now to have the Unix system auto-dial their phone to place a call. For example, they can pull up a customer or vendor record in CRT. The application uses the CRT "pass-thru" option to pass a modem dial string through their PC to a modem plugged into the PC's serial port. The modem is attached to their telephone so that they lift the receiver, pick an outgoing line, and the modem dials the number. I need to be able to do that using Asterisk and a SIP phone (or an ADSI phone). Any ideas? Is this possible? I was thinking that maybe there would be a way for the Unix box to command the Asterisk box to connect to the SIP phone, and place the outgoing call, and then join those 2 connections together. TIA ___ 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
