[asterisk-users] Aastra IP phone configuration generator
For anyone who is interested I've recently created an Aastra IP Phone config generator. I don't know if one existed but thought I'd create it anyways. It can be found at http://www.lraweb.pwp.blueyonder.co.uk/. If you have any problems or stuff you want adding then please contact at the address listed on the web page. Regards Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with SuSe 10
Title: Asterisk with SuSe 10 Has anyone had any experience with the Asterisk on a SuSe 10 platform? I'm currently using FC3 but because we use SuSe within other parts of the business I'm being pushed to changed the OS. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with SuSe 10
Thanks, I've got it running on my test box but didn't know if there was any global objection to using it. I've had a few funnies with it but that might be down to Supermicro and P4's with the EM64T thing. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Klang Sent: 24 January 2006 15:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with SuSe 10 On Tuesday 24 January 2006 09:26, Lee Archer wrote: > Has anyone had any experience with the Asterisk on a SuSe 10 platform? > I'm currently using FC3 but because we use SuSe within other parts of > the business I'm being pushed to changed the OS. Just about all of my production Asterisk servers are on SuSE 9.3. My development and demo boxes are SuSE 10. Both run great. I do however usually tweak the RPM that came with it to add in a few patches. If you are comfortable with running Asterisk 1.0.9 then the RPM works very well. SuSE always seems to really think things through when they package applications. For running something newer than Asterisk 1.0.9 SuSE 10 is also works fine. For your own sanity you'll want to not install/uninstall the SuSE Asterisk RPMs. One possible gotcha: be careful of possibly conflicting kernel modules in /lib/modules/`uname -r`/extra as the Zaptel drivers are not part of any Asterisk package but rather the kernel. The zaptel compile from source installs modules to /lib/modules/`uname -r`/misc so you'll want to delete the files in extra. You'll also have to remember that each time you update the kernel RPM. Hope that helps. The bottom line from me is Thumbs Up. /BAK/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budgetone mass deployment?
I had a problem with the scripts you can bulk generate, they are linked to the MAC address you initially put in, so if the phone packs in you can't just rename the file. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Blundell Sent: 30 January 2006 13:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream Budgetone mass deployment? On Mon, 2006-01-30 at 15:14 +0200, Dmitry Ivanov wrote: > On Monday 30 January 2006 13:03, Phil Blundell wrote: > > Personally I'd be a bit wary of mass Budgetone deployment for other > > reasons, but the remote configuration stuff shouldn't be a problem. > > What reasons do you mean? Just that, from my limited experience of Budgetones, they seem to be generally a bit buggy. But if they work OK in your environment, there's probably no reason not to use them. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Streaming MOH
Title: Streaming MOH Hi, I'm having some problems getting this to work with Asterisk 1.2.4. Does it work for anyone? Does anyone have a site I can test this with? Regards Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outbound Caller ID number on E1
I have this problem in the UK on BT too. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: 02 February 2006 11:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Outbound Caller ID number on E1 Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van Sittert BSc (Physics & Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming MOH
I've got it working now but the playback through the handset is sloow. I can tell it's music but you couldn't sing along to it... Still maybe it's about the right speed for a hangover. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Gladden Sent: 02 February 2006 15:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Streaming MOH Not tried 1.2.4 yet I'm using 1.2.3 and an old version of mpg123 You should be able to use any streaming mp3 that you can find on shoutcast for test. http://www.shoutcast.com Click one of the 'tune in buttons' to download a playlist (pls) file and open in your favorite text editor. Or let it open in your MP3 player and view the properties of the stream. I have several streaming servers here, if you need a test link or want to listen to live air traffic in Detroit Michigan, send me a personal email and I can give you a link for testing. I'd rather not post it here only to end up indexed by google in a few days ;-) Steve > Hi, I'm having some problems getting this to work with Asterisk 1.2.4. > Does it work for anyone? Does anyone have a site I can test this with? > > Regards > > Lee > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Musiconhold in zapata.conf
Title: Musiconhold in zapata.conf I've been trying to change the musiconhold= in the zapata.conf to use something other than default. However it doesn't seem to do it. I know the other musiconhold source works but whatever I set it to in the zapata.conf file it always plays whatever is [default] in musiconhold.conf. Also random=yes doesn't work. [default] mode=files directory=/var/lib/asterisk/mohmp3 random=yes [livestream1] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@ /etc/asterisk/playlist Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Double ring
Title: Double ring Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Double ring
Oddly enough I'm on Aastra phones too. Doesn't happen with Grandstream phones. I've tried callprogram=yes and no to no effect. What firmware did you have, I'm on 1.3. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 12:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I was getting something very similar with my Aastra test phones until I change 'callprogress=' to 'no'. Thanks, Bob On Fri, 10 Feb 2006 12:13:47 - "Lee Archer" <[EMAIL PROTECTED]> wrote: > Can anyone shed any light on to why I get a double ring when calling >external numbers? When calling out I hear the actually ring-ring of >the called phone and the asterisk ring tone. I'm using the same >config I used with 1.0.10 but have now upgraded to 1.2.4. > > Regards > > Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Double ring
Sorry I meant callprogress. I've tried it set to yes and no with no difference. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 13:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring 'callprogress', in zapata.conf: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf Thanks, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Friday, February 10, 2006 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring Oddly enough I'm on Aastra phones too. Doesn't happen with Grandstream phones. I've tried callprogram=yes and no to no effect. What firmware did you have, I'm on 1.3. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 12:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I was getting something very similar with my Aastra test phones until I change 'callprogress=' to 'no'. Thanks, Bob On Fri, 10 Feb 2006 12:13:47 - "Lee Archer" <[EMAIL PROTECTED]> wrote: > Can anyone shed any light on to why I get a double ring when calling >external numbers? When calling out I hear the actually ring-ring of >the called phone and the asterisk ring tone. I'm using the same >config I used with 1.0.10 but have now upgraded to 1.2.4. > > Regards > > Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Double ring
Am I the only one with this problem? I've got Aastra phones running the 1.3 firmware. It doesn't happen on the Grandstream phones but I'd like to know if anyone else has Aastra 9133i phones with the 1.3 firmware and Asterisk 1.2.4. I'm running a TE110P Pri card. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: 10 February 2006 13:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring Sorry I meant callprogress. I've tried it set to yes and no with no difference. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 13:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring 'callprogress', in zapata.conf: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf Thanks, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Friday, February 10, 2006 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring Oddly enough I'm on Aastra phones too. Doesn't happen with Grandstream phones. I've tried callprogram=yes and no to no effect. What firmware did you have, I'm on 1.3. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 12:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I was getting something very similar with my Aastra test phones until I change 'callprogress=' to 'no'. Thanks, Bob On Fri, 10 Feb 2006 12:13:47 - "Lee Archer" <[EMAIL PROTECTED]> wrote: > Can anyone shed any light on to why I get a double ring when calling >external numbers? When calling out I hear the actually ring-ring of >the called phone and the asterisk ring tone. I'm using the same >config I used with 1.0.10 but have now upgraded to 1.2.4. > > Regards > > Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Double ring
Hi, the setting progressinband=no seems to fix the problem with my Aastra phones. The Grandstreams were unaffected and still are. Thanks Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Domjan Attila Sent: 10 February 2006 14:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I have the similar problem with thomson sip voip cable modems: http://bugs.digium.com/view.php?id=6083 On Fri, 2006-02-10 at 12:13 +, Lee Archer wrote: > Can anyone shed any light on to why I get a double ring when calling > external numbers? When calling out I hear the actually ring-ring of > the called phone and the asterisk ring tone. I'm using the same > config I used with 1.0.10 but have now upgraded to 1.2.4. > > Regards > > Lee > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Double ring
Progressinband=no fixed the issue for me. I've been onto Aastra support already about it. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward de Zeeuw Sent: 14 February 2006 14:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I noticed the issue today and came looking for confirmation when I came upon this thread. My Grandstream does not have this problem. SPA-941, Snom 320 and Aastra 480i all demonstrate this issue. I'm going to Lee Archer wrote: > Am I the only one with this problem? I've got Aastra phones running > the > 1.3 firmware. It doesn't happen on the Grandstream phones but I'd > like to know if anyone else has Aastra 9133i phones with the 1.3 > firmware and Asterisk 1.2.4. I'm running a TE110P Pri card. > > Regards > > Lee > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Lee > Archer > Sent: 10 February 2006 13:50 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Double ring > > Sorry I meant callprogress. I've tried it set to yes and no with no > difference. > > Lee > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Bob > McDowell > Sent: 10 February 2006 13:14 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Double ring > > > 'callprogress', in zapata.conf: > > http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.co > nf > > > > Thanks, > > Bob McDowell > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Lee > Archer > Sent: Friday, February 10, 2006 7:04 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Double ring > > Oddly enough I'm on Aastra phones too. Doesn't happen with Grandstream > phones. I've tried callprogram=yes and no to no effect. What firmware > did you have, I'm on 1.3. > > Regards > > Lee > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Bob > McDowell > Sent: 10 February 2006 12:27 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Double ring > > > I was getting something very similar with my Aastra test phones until > I change 'callprogress=' to 'no'. > > Thanks, > > Bob > > On Fri, 10 Feb 2006 12:13:47 - > "Lee Archer" <[EMAIL PROTECTED]> wrote: > >> Can anyone shed any light on to why I get a double ring when calling >> external numbers? When calling out I hear the actually ring-ring of >> the called phone and the asterisk ring tone. I'm using the same >> config I used with 1.0.10 but have now upgraded to 1.2.4. >> >> Regards >> >> Lee >> > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ### > > This message has been scanned by F-Secure Anti-Virus for Microsoft > Exchange. > For more information, connect to http://www.f-secure.com/ > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ### > > This message has been scanned by F-Secure Anti-Virus for Microsoft > Exchange. > For more information, connect to http://www.f-secure.com/ > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ### > > This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. > For more information, connect to http://www.f-secure.com/ > __
RE: [Asterisk-Users] Firmware version 1.3.1 released for Aastra IPphones
Title: Firmware version 1.3.1 released for Aastra IP phones There is no release note, just a text file that says AASTRA TELECOM INC. February 2006 FC-46-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for customer release. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth OwenSent: 15 February 2006 02:00To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Firmware version 1.3.1 released for Aastra IPphones Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range ofIP phones (480i, 480iCT, 9112i and 9133i).The firmware and release notes (no updated admin and user guides yet)are available for download at:http://www.aastra.com/support/enterpriseipContrary to what the version numbering would suggest, this is a significantupdate with many new features and bug fixes. See the release notes forfull details, but here are some hightlights for Asterisk users: - Context-sensitive softkeys. Softkeys can now be configured for eachof the following call states: idle, incoming, outgoing and connected - Speed dial using the BLF key - Per-line outbound proxy - Use the Icom key to make intercom calls - Further XML enhancements - Voice quality (transmit level) issues resolved - Keypad now continues to work when a second incoming call appearsAnd much more. ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firmware version 1.3.1 released for AastraIPphones
Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 16 February 2006 14:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Firmware version 1.3.1 released for AastraIPphones On Thu, 2006-02-16 at 13:28 +, Lee Archer wrote: > There is no release note, just a text file that says > > AASTRA TELECOM INC. > > February 2006 > > FC-46-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for > customer release. > http://www.aastra.com/support/show_manuals.asp?p=241 Worked for me:) -- Dave Cotton <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firmware version 1.3.1 released for Aastra IPphones
Title: Firmware version 1.3.1 released for Aastra IP phones Any chance of getting a config option in that allows you set headset/speaker in the audio menu? Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth OwenSent: 15 February 2006 02:00To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Firmware version 1.3.1 released for Aastra IPphones Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range ofIP phones (480i, 480iCT, 9112i and 9133i).The firmware and release notes (no updated admin and user guides yet)are available for download at:http://www.aastra.com/support/enterpriseipContrary to what the version numbering would suggest, this is a significantupdate with many new features and bug fixes. See the release notes forfull details, but here are some hightlights for Asterisk users: - Context-sensitive softkeys. Softkeys can now be configured for eachof the following call states: idle, incoming, outgoing and connected - Speed dial using the BLF key - Per-line outbound proxy - Use the Icom key to make intercom calls - Further XML enhancements - Voice quality (transmit level) issues resolved - Keypad now continues to work when a second incoming call appearsAnd much more. ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for Aastra IPphones
Nice one it works. Is there a complete list of all the options you can use in the config files? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Owen Sent: 17 February 2006 13:39 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Firmware version 1.3.1 released for Aastra IPphones The follow should work from the configuration files (aasta.cfg/.cfg), although I haven't tried it... audio mode: Where is a number between 0 and 3 0 = speaker 1 = headset 2 = speaker/headset 3 = headset/speaker Gareth Lee Archer wrote: > > Any chance of getting a config option in that allows you set > headset/speaker in the audio menu? > > Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
Worth saying that the Aastra 9133i with the 1.3.1 firmware is a pretty good phone. I used to run GXP-2000's, still have 10 new in a box and another 20 in demo/test circulation, but I also run a few dozen 9133i, 480i and 9112i phones and I think Aastra are getting their now. Biggest problem I had with GXP are the usual power flakyness, which you can't really do much about but apart from that no real problems. Now the GXP firmware is getting there might offer them as a cheaper phone to the 9133i. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 19 February 2006 13:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grandstream GXP-2000 On Sat, 18 Feb 2006, Michael J. Liberatore wrote: > Well the gxp-2000 has BLF, the polycom 501 does not correct? I had an > astra 480i and it was prety bad, but I was going to test the 9133i for > an inexpensive phone to compete with the gxp2000. The gxp2000 is not > bad though, the new firmware helps a lot, but once they work out the > echo bugs fully and the various minor stuff it will be a good sub $100 > phone. I am yet to find a phone under $300 that's perfect... The snom > 360 is nice, but I have lots of problems with those too. I havent > tried any polycom's though and starting to think they might be some of > th ebest... The GXP2000 is good value for the money. It is not a great phone but for your $80 you get a lot more than one would expect. 7 programmable buttons with BLF, Backlight, dual 100bt. Stuff you dont find on some phones over twice the price... All phones have their warts, even cisco. For $80 I can live with the GXP2000's warts, grandstream do seem to be actively improving the firmware and fixing what they can. Asterisk features (mwi, blf) "just work" out of the box without the gyrations one has to go through for other vendors phones. I have some $200+ phones which have some serious warts and the vendors do not seem terribly interested in fixing them. Big money does not always mean good value. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones
OK, well the audio option was the last one I required for now. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Owen Sent: 19 February 2006 16:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones The short answer is all the officially supported configuration parameters are in the admin guide and release notes. Options that aren't documented aren't guaranteed to work between releases. So, sorry but the current documentation contains "all" the config options. Gareth -Original Message- From: [EMAIL PROTECTED] on behalf of Lee Archer Sent: Fri 2/17/2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: Firmware version 1.3.1 released for AastraIPphones Nice one it works. Is there a complete list of all the options you can use in the config files? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Owen Sent: 17 February 2006 13:39 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Firmware version 1.3.1 released for Aastra IPphones The follow should work from the configuration files (aasta.cfg/.cfg), although I haven't tried it... audio mode: Where is a number between 0 and 3 0 = speaker 1 = headset 2 = speaker/headset 3 = headset/speaker Gareth Lee Archer wrote: > > Any chance of getting a config option in that allows you set > headset/speaker in the audio menu? > > Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000
Yes this is quite an issue. The POE converter is 'optional'. I bought a 480i a while back and after waiting a few days had to order the POE cos the dealer hadn't told me it was actually required! Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 20 February 2006 19:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream GXP-2000 On Mon, 20 Feb 2006, Richard Amerman wrote: > One thing to keep in mind with PoE is that you can simply use an > injector at the phone location. At least with the 480i you can easily > order the phone with the power injector. Aastra does not really make it clear that the 480i is poe _only_. A lot of people are very suprised when I explain to them that the 480i is poe only. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming Music On Hold
I spent a days or two on this and in the end did
Musiconhold.conf
[livestream1]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@
/etc/asterisk/stream.playlist
Then in stream.playlist I just put the links from Shoutcast I wanted to
use
http://64.236.34.67:80/stream/1040
http://64.236.34.196:80/stream/1040
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: 22 February 2006 21:18
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Streaming Music On Hold
Thanks. I got it working. Yay.
Now, it seems that Asterisk is very fussy with the streams. A lot don't
work, especially when the URL ends in something.pls. Anyone know if
that's true? Is Asterisk's support of this still pretty limited?
Doug.
-Original Message-
From: Jonathan Augenstine [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 22, 2006 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Streaming Music On Hold
Try this:
musiconhold.conf:
[stream2]
mode=mp3
directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr
extensions.conf:
exten => 1234,1,Answer
exten => 1234,2,MusicOnHold(stream2)
exten => 1234,3,Hangup
On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote:
> Ok, I'm tearing my hair out trying to get Asterisk moh streaming to
work. After several hours jerking around with icecast and muse, I tried
to point my asterisk system directly at two streams I know work.
>
> This is what extensions.conf has:
>
> [default]
> mode=quietmp3
> directory=/var/lib/asterisk/mohmp3
>
> [stream2]
> mode=custom
> directory=/var/lib/asterisk/mohmp3-empty
> application=http://pubint.ic.llnwd.net/stream/pubint_wnpr
>
> and this is how I am testing it:
> exten => 1234,1,Answer
> exten => 1234,2,SetMusiconHold(stream2) exten =>
> 1234,3,WaitmusiconHold(60) exten => 1234,4,Hangup
>
> and this is the console output I get when I dial 1234:
>
> Asterisk Ready.
> *CLI> -- Executing Answer("SIP/3250072-ed28", "") in new stack
> -- Executing SetMusicOnHold("SIP/3250072-ed28", "stream2") in new
stack
> -- Executing WaitMusicOnHold("SIP/3250072-ed28", "60") in new
stack
> -- Started music on hold, class 'stream2', on channel
'SIP/3250072-ed28'
> -- Stopped music on hold on SIP/3250072-ed28
>
> If I replace SetMusiconHold(stream2) with SetMusiconHold(default), I
get the default music on hold. Running ngrep on port 80 shows me that
the Asterisk system is not sending or receiving ANY data on port 80.
What am I doing wrong? Yes, it has network and DNS connectivity.
>
> Can't believe it's this hard!
>
> Doug.
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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RE: [Asterisk-Users] mpg123 alternative?
I used madplay with * 1.0 and moved to native for playing mp3's with 1.2 with no problems. Depends what you want to play, doesn't native stop when there is no one to play to then restart when there is someone to play to? Might be a problem if you want to plays ads and don't have many callers, like in my environment, to play the MOH to. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolás Gudiño Sent: 23 February 2006 13:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 alternative? Hi Rich, > Been using mpg123 for moh for the last two years or so. However, when > I have * config errors, often times get a endless stream of console > messages and need to kill the two mpg123 processes. > > Is there an alternative to mpg123 that eliminates that issue? > > I see references in musiconhold.conf relative to madplay, native file > format, asterisk-addons, etc. Not sure why the asterisk-addon approach > hasn't been moved into trunk, or if madplay is a better choice on this > fc3 trunk box. > > Any suggestions? I've switched to native moh and never had to worry again about dead or unresponsive mpg123 processes. -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 alternative?
Check out the musiconhold.conf.sample in the asterisksource/configs folder. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: 23 February 2006 18:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 alternative? Ah! Now this is actually something I've not been able to get my head around: > Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which > has its own MP3 player. Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I use it ? I still seem to have the usual two mpg123 processes running with 1.2.4, with whatever music on hold is set in musiconhold.conf I'm sure it is very obvious, but I can't for the life of me figure out what I'm supposed to do to use the built-in MP3 player facilities. I just have the following in my musiconhold.conf: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 random=yes Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HDLC error
Title: HDLC error Can anyone help and point me in a useful direction. I'm using * 1.2.4 with Zaptel 1.2.4. I have a TE110P card and it’s a Supermicro P8SCT mobo. If I run the PRI card in the PCI-X slot it shares an interupt with eth0 but I don't get problems. I've been trying to move it onto it's own IRQ, by moving the card to a regular PCI slot but I now get the errors Mar 2 08:43:09 NOTICE[6272]: chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 2 08:43:20 NOTICE[6272]: chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 I've done a few searches and tried a few things but still get it. My zaptel.conf looks like span=1,1,0,ccs,hdb3 bchan=1-8 dchan=16 The system works but there is popping and the above messages. I'd rather run the cards on different IRQ's but I'm not sure if it's the mobo I'm using or the something in the config I need to change. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to change Budgetone dialtone?
Hi try http://www.grandstream.com/y-downloads.htm Download the IP Phone Custom Ringtones Generation Tool Unzip and read the readme Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: 07 March 2006 13:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to change Budgetone dialtone? Good day! Is is possible to change dialtone (and other tones as well) in BT-102? ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to change Budgetone dialtone?
Sorry... Just ignore me. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: 07 March 2006 14:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to change Budgetone dialtone? On Tuesday 07 March 2006 15:49, Lee Archer wrote: > Download the IP Phone Custom Ringtones Generation Tool Unzip and read > the readme Ringtone != dialtone. ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ChanSpy on Asterisk v1.0.7
What's the best way to get 1.0.8? I've downloaded the latest from CVS but when I compile it it says 1.0.6!! Is that right? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: 23 June 2005 16:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ChanSpy on Asterisk v1.0.7 Just use CVS-HEAD.. stable is a pile of crap. /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jun 23, 2005, at 10:09 AM, Tim Karl wrote: > I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried > looking on VOIP-info.org's ChanSpy page (http://www.voip- > info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to > the link regarding bug 3836 (http://bugs.digium.com/ > bug_view_page.php?bug_id=0003836). I downloaded the attachments and > tried to use the patch and compile the source. However, it seems that > these files are for a different version of Asterisk. Searching Google > provides no relevant material. > > If anyone has any information as to where I can find ChanSpy for > Asterisk v1.0.7 please reply. Thank you for your help. > > --Timothy Karl > ___ > 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.11/26 - Release Date: 22/06/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.11/26 - Release Date: 22/06/2005 ___ 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] Spinlock with ZAPTEL
Title: Spinlock with ZAPTEL Hi, I'm using Fedora Core 3 and the 1.0.8 version of ZAPTEL. Why do I get spinlocks when I modprobe -r it but the HEAD version unloads fine? Regards Lee -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.5/32 - Release Date: 27/06/2005 ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Running commands from dialplans
Title: Running commands from dialplans Hi, is it possible to run a command like system but from outside of a dial plan? E.g. ; include extension contexts generated from AMP #include extensions_additional.conf In extensions.conf but I need to run a command. Regards Lee -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.13/47 - Release Date: 12/07/2005 ___ 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] isdn30 / pri lines in the UK
Also NTL don't drop the leading 0 on incoming numbers like BT do. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton Sent: 18 July 2005 11:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] isdn30 / pri lines in the UK NTL install isdn 30. No idea how good they are though. Chris - Original Message - From: "1 2" <[EMAIL PROTECTED]> To: Sent: Thursday, July 07, 2005 2:50 PM Subject: [Asterisk-Users] isdn30 / pri lines in the UK > anybody recommend a supplier in the UK for a pri/isdn30 line (other than > BT) > > thanx very much > > __ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.0/50 - Release Date: 16/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.0/50 - Release Date: 16/07/2005 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New voiceovers for Allison Smith: submit today
Could anything with 'press pound' in it be recorded with 'press hash' please for us UK users? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: 20 July 2005 23:55 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] New voiceovers for Allison Smith: submit today I'm sending in a set of voiceover requests to Allison Smith this afternoon. I haven't kept up with the -users list to know if there is someone keeping track of this stuff any more... We only have a few phrases for her to record, and if anyone has applications which require Allison's voice for the "asterisk-sounds" repository, let me know. I'll be sending this in around 22:00 PDT today, so act fast. Please format the requests in the style: %filename%text-to-speak example: %auth-incorrect.gsm%Login incorrect. Please enter your password followed by the pound key. Any pronunciation keys should be in-line, inside of [brackets]. Please email directly to me. JT ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/53 - Release Date: 20/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/53 - Release Date: 20/07/2005 ___ 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] Grandstream GXP2000 resetting all the time
I have all my GXP-2000's set to dynamic with no problems. You need to make sure they have the latest firmware, as this fixed a few issues and improves the overall usage of the phone. Hopefully they will make the useless LED's work so we can line monitor etc... Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan C. Smith Sent: 20 July 2005 18:46 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Grandstream GXP2000 resetting all the time Do you have the address set to dynamic or static in sip.conf? -Original Message- From: Michiel van Baak [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 20, 2005 1:59 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Grandstream GXP2000 resetting all the time On 16:06, Wed 20 Jul 05, [EMAIL PROTECTED] wrote: > All, > > I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 > phones. > All seems well other than the phones have to be reset up to 5 times per day. > It is like they lose thier ip connection or maybe thier SIP connection. Has > anyone else experienced this issue? I have the phones set for static IP > addresses and that doesnt seem to help either. Any help would be greatly > appreciated. > > Marc Hi, Are you using the latest firmware on the phones ? We use 1.0.1.9 and have no problems at all. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?" ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/53 - Release Date: 20/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/53 - Release Date: 20/07/2005 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping call
Title: Dropping call Hi, after upgrading from 1.0.7 to 1.0.9 I now seem to have a call drop problem. It mostly happens after about 1min 30 secs but also happens are random intervals. Everything was fine with 1.0.9 and I'm using the same config files. Could it be a zaptel problem? Does anyone have any ideas? Regards Lee -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/54 - Release Date: 21/07/2005 ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: zaptel make problems
On a different note using Fedora Core 3 I get CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_chan_write': /usr/src/zaptel/zaptel.c:1745: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: In function `ioctl_load_zone': /usr/src/zaptel/zaptel.c:2392: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: In function `zt_common_ioctl': /usr/src/zaptel/zaptel.c:2744: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:2804: warning: ignoring return value of `copy_to_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:2807: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:2889: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:2919: warning: ignoring return value of `copy_to_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: In function `zt_chanandpseudo_ioctl': /usr/src/zaptel/zaptel.c:3641: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:3651: warning: ignoring return value of `copy_to_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:3654: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:3713: warning: ignoring return value of `copy_to_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c:3717: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: At top level: /usr/src/zaptel/zaptel.c:176: warning: 'fcstab' defined but not used When building the stable or head zaptel with kernel linux-2.6.11-1.35_FC3. The module compiles but it never used to give this message on FC2. Anyone got any ideas? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 22 July 2005 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: zaptel make problems On Fri, 2005-07-22 at 07:59 +0200, Aldo Bergamini wrote: > [EMAIL PROTECTED] is believed to have said: > > > > >and watch linus himself rant about how this is incorrect to do (yet > >all the distros do it) :P > > > > Well, this is reassuring for a newbie like me. > > Even the pros (as anybody building a distro ought to be, and most of > the times, really is) can do obvious errors... Who said it's an error, Linus just does not like it and thinks says it's incorrect, it causes no errors, and when you have multiple kernel sources on the same machine it makes life much easier. I would agree that going through multiple symlinks is bad practice, this could also be Linus' argument, or maybe it's multiple times through the same symlink in the case of a kernel compile. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/55 - Release Date: 21/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/55 - Release Date: 21/07/2005 ___ 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] SATA
Title: SATA Has anyone had any problems with SATA, either on board or 3rd party setup? I've currently got a problem where an AMD non SATA FC2 system is working fine but an Intel system with a 3Ware SATA card and FC3 is radomly not syncing with the ISDN30. It allows and receives calls but at random intervals drops them. Regards Lee -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/55 - Release Date: 21/07/2005 ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 - two processes
I noticed this, but then I moved to madplay which only uses 1 process. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: 27 July 2005 03:38 To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 - two processes Yes, I always have two. MARK. Billy Dunn wrote: > Does everyone have two processes running for mpg123? I always have > them when I'm running an idle Asterisk box. No calls going in or out > and nothing off hook. Is this normal? Thanks! > > 5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 > fpm-calm-ri > 5015 ?S 0:00 /usr/sbin/asterisk > 5061 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 > fpm-calm-ri > > ___ > 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I found problem with TE110P and the new kernel offedora, "kernel panic"
I had a problem with this card and 2.6.11 kernel. I am using FC3 but sticking with the 2.6.9 kernel. I had a lot of make warnings on the zaptel build and the card played up. It also wouldn't do a modprobe -r without crashing the system. With 2.6.9 zaptel compiles fine and I can unload the mod as and when. Also stay well away from the 2.6.12 FC3 kernel as it didn't work at all and didn't come with any sources. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yousef HerzallahSent: 27 July 2005 09:34To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] I found problem with TE110P and the new kernel offedora, "kernel panic" I installed a new fedora 3 and i did the yum update, In this way I upgrade the kernel 2.6.11-35, before I used the 2.6.9-77 for fedora and it was work perfectly no problem. When I made the upgrade I got the “kernel panic” every time that I remove the drivers or restart the computer. --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ___ 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] Zaptel error: Unable to create channel op type'Zap'
I tend to make it pause for 10 secs when loading the module as I have had a few occurances of loading before /dev/zap has been populated. Wouldn't trust the 2.6.12 kernel as far as I could virutally throw it. Has anyone had any problems with PCI-X systems? In particular call dropping? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman Sent: 27 July 2005 12:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap' I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: > Hi, > > In struggeling with this problem for a two weeks now. > I have a X100P clone card in my * box but I'm not able to get it to run. > I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA > EPIAML500EA > > The compiling of both zaptel and asterisk went without any errors. > I can run zaptel and asterisk without any errors. > When I run ztcfg I don't get any errors too. > > But when I try to place a call trough my x100p I get this error > message in asterisk: > NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op > type 'Zap' > > Outside calls are not comming in either. > > Here are my zapata.conf and zaptel.conf: > > > -zapata.conf- > [channels] > signalling=fxs_ks > context=incoming > channel=>1 > > -zaptel.conf- > loadzone = nl > defaultzone=nl > > fxsks=1 > > --- > > The funny part comes here: > I'm installing a *box for a friend with a ISDN card and the same > problem occures. > So I probarbly doing something wrong in fedora... > > Any ideas??? > > Thanks, > Peter > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ___ 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] Setup faxing with latest CVS/STABLE
I have been trying to get faxing working with stable but I have had no luck since cvs 1.0.4. I've tried 3 versions of SpanDSP and the system answers the fax but looks like it isn't training properly. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Goddard Sent: 06 August 2005 23:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Setup faxing with latest CVS On Saturday 06 Aug 2005 22:41, Erick Johnson wrote: > I have been trying to setup faxing with a recent CVS-HEAD. I have > downloaded and compiled spandsp-0.0.2pre18 and gotten > apps_makefile.patch, app_txfax.c and app_rxfax.c > > I'm not suprised that the patch failed. Does anyone know what changes > need to be made for this to work? > > I have very little Fedora experience and no experience in changing > make files so this is all new to me. At least take a look, you'll find the changes are very simple. B ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.2/65 - Release Date: 07/08/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.2/65 - Release Date: 07/08/2005 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems
But where do can you get this later firmware from? I'm still on 1.0.0.78 on my
480i.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Passchier
Sent: 05 August 2005 00:04
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems
The ALERT_INFO variable works for 480i firmwares 1.2.1.207 and up (like the
1.2.5-series).
Set it like in the example below:
exten=_*55XXX,1,SetVar(ALERT_INFO=info=alert-autoanswer)
exten=_*55XXX,2,Dial(SIP/${EXTEN:3},12,Ttr)
(*55 will be the prefix for the normal phone number; if a single digit is used
--or anything of a different length-- adjust the slicing of the ${EXTEN}, like
${EXTEN:1} for a single digit)
The 'info=alert-autoanswer' is the only value that seems to do something.
Peter Passchier
Sayson Technologies Ltd.
210 - 1910 Quebec St
Vancouver, BC V5T4K1
Canada
Phone: 604.730.1842
Fax: 604.732.8726
*
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RE: [Asterisk-Users] Hitachi wip5000
This phone works fine, however the initial firmware it came with was awkward. Once updated no problems, even NTP works! Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: 11 August 2005 01:23 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hitachi wip5000 WIP 5000 works fine. Only issue I had was send text messages from the phone. That did not work for me. ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.9/72 - Release Date: 14/08/2005 ___ 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] CRM software
Title: CRM software Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together. Regards Lee ___ 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] Faxing help
Title: Faxing help Hi, I have still had no luck with faxing and am getting a couple of pages of the following debug message Changed from phase 1 to 4 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: Unlimited Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) >>> DIS: 80 00 ce f4 80 80 81 80 80 80 18 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down Slow carrier up Slow carrier down This goes on for a few seconds before hanging up. Anyone got any ideas what this means or how it can be fixed? Regards Lee ___ 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] fedora core 3 kernel source - could someone throwthe dog a bone!
I found that only the kernel is installed. I'd avoid 2.6.12 for now as I had problem with the zaptel driver and stay with 2.6.9. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: 24 August 2005 22:33To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] fedora core 3 kernel source - could someone throwthe dog a bone! This could be a duplicate post, sent it originally 4 hours ago, it never showed up! I know this is a question with an obvious answer to some, but I am not one of them. Installed FC3, but this time I decide to update since my ISOs are a bit old, so typical yum update Downloaded the FC3 SRPM for my kernel 2.6.12… Installed the SRPM package Ran rpmbuild –bp –target=i686 kernel-2.6.spec Tried to build zaptel – error; You do not appear to have the sources for the 2.6.12-1.1372_FC3smp kernel installed. So I assume that either a) I did not build the correct source for the smp kernel, or b) I am missing a symbolic link to the kernel source. No help from the FC3 release notes, no help from a Google. So, if you don’t mind, throw me the bone…###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2
Hi, do you have an on-site NTP server? I found that after the firmware update NTP from the * server stopped working. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus MogollonSent: 24 August 2005 22:11To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2 Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12 to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls start stuttering. The only way this can be fixed is by rebooting the phone. We were able to replicate this problem in all phones while some Polycoms we have do not suffer from this problem. Again, this problem happened AFTER we upgraded to the new firmware. Has anyone seen this?Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com/###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2
Well it's only worked once and I've left the phones several hours. I've done various debugs and the phone is asking for NTP and the server is answering but its not getting set. Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus MogollonSent: 25 August 2005 12:54To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2 Hi Lee: NTP is working as expected, but it does take a couple of minutes (!) to get the date from the serverJesus Mogollon 2005/8/25, Lee Archer <[EMAIL PROTECTED]>: Hi, do you have an on-site NTP server? I found that after the firmware update NTP from the * server stopped working. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jesus MogollonSent: 24 August 2005 22:11To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2 Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12 to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls start stuttering. The only way this can be fixed is by rebooting the phone. We were able to replicate this problem in all phones while some Polycoms we have do not suffer from this problem. Again, this problem happened AFTER we upgraded to the new firmware. Has anyone seen this?Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com/### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2
I can time sync with time.nist.gov but not with any internal servers. I read in the changelog about them fixing something related to NTP on the same subnet but it doesn't say whether it should work or shouldn't. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus MogollonSent: 25 August 2005 12:54To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2 Hi Lee: NTP is working as expected, but it does take a couple of minutes (!) to get the date from the serverJesus Mogollon 2005/8/25, Lee Archer <[EMAIL PROTECTED]>: Hi, do you have an on-site NTP server? I found that after the firmware update NTP from the * server stopped working. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jesus MogollonSent: 24 August 2005 22:11To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] GXP 2000 Firmware 1.0.1.2 Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12 to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls start stuttering. The only way this can be fixed is by rebooting the phone. We were able to replicate this problem in all phones while some Polycoms we have do not suffer from this problem. Again, this problem happened AFTER we upgraded to the new firmware. Has anyone seen this?Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com/### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fedora core 3 kernel source - couldsomeonethrowthe dog a bone!
The issue I have had with all other FC3 kernels apart from the 2.6.9 one was that the zaptel build would throw lots of warnings up. This would have the knock on of hanging the system, spinlock I think the problem was, on a modprobe -r. Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: 26 August 2005 17:46To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] fedora core 3 kernel source - couldsomeonethrowthe dog a bone! What was the issue with zaptel and 2.6.12? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: Thursday, August 25, 2005 1:22 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] fedora core 3 kernel source - could someonethrowthe dog a bone! I found that only the kernel is installed. I'd avoid 2.6.12 for now as I had problem with the zaptel driver and stay with 2.6.9. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: 24 August 2005 22:33To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] fedora core 3 kernel source - could someone throwthe dog a bone! This could be a duplicate post, sent it originally 4 hours ago, it never showed up! I know this is a question with an obvious answer to some, but I am not one of them. Installed FC3, but this time I decide to update since my ISOs are a bit old, so typical yum update Downloaded the FC3 SRPM for my kernel 2.6.12… Installed the SRPM package Ran rpmbuild –bp –target=i686 kernel-2.6.spec Tried to build zaptel – error; You do not appear to have the sources for the 2.6.12-1.1372_FC3smp kernel installed. So I assume that either a) I did not build the correct source for the smp kernel, or b) I am missing a symbolic link to the kernel source. No help from the FC3 release notes, no help from a Google. So, if you don’t mind, throw me the bone… ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting wcte11xp card to use IRQ
Title: Setting wcte11xp card to use IRQ Hi, is it possible to set a wcte11xp card to use a certain IRQ? I've tried a few things but it always shares the IRQ with eth0 even though the system has 4 spare ones. I can't set it via the BIOS. Regards Lee ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFAX
Title: RxFAX
Does anyone have any ideas on why I can fax out using TxFax fine but I can't receive? The system detects a fax and…
-- Executing RxFAX("Zap/1-1", "/var/spool/asterisk/fax/1124786077.0.tif|debug") in new stack
Slow carrier up
Slow carrier down
Changed from phase 1 to 4
DIS:
Prefer 256 octet blocks
Can receive fax
Supported data signalling rates: V.27ter and V.29
R8x7.7lines/mm and/or 200x200pels/25.4mm
2D coding
Scan line length: 215mm
Recording length: Unlimited
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm
Minimum scan line time for higher resolutions: T15.4 = T7.7
North American Letter (215.9mm x 279.4mm)
North American Legal (215.9mm x 355.6mm)
>>> DIS: 80 00 ce f4 80 80 81 80 80 80 18
HDLC underflow in state 9
Changed from phase 4 to 3
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Is what I get. I am using 1.0.9 and SpanDSP .18 with a 100XP card.
Regards
Lee
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RE: [Asterisk-Users] RXFax
I have the exact same problem. TXFAX is fine. It's someone in rxfax that's the problem as my system going into receive mode then hangs up. Odd thing is I had this working before but now it doesn't. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Sent: 15 September 2005 05:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] RXFax I got txfax to work, but the rxfax fails on the training.I've read where spandsp has debugging features, but I don't see the log even though I specificed the debug on the rxfax call.I'm using a Digium FXO card and the ztmonitor was reporting 100%. Does anyone have anything I can check? Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Will Digium Wildard work with PCI-Xor PCI Express
I had trouble with a TE110P card in a Supermicro mobo - P8SCT. The PRI line kept dropping calls when the card was in a standard PCI slot. In the end the only way to fix it was to install the card in the PCI-X slot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth Sent: 22 September 2005 21:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Will Digium Wildard work with PCI-Xor PCI Express Just correcting myself. The 3 PCI-X slots are one 64-bit 133 MHz and two 64-bit 100 MHz. Matt Matt Roth wrote: > Don't bank on it. We were going to use a Wildcard as a timing source > on our Dell PowerEdge 6850 and the BIOS didn't see it. Depending on > the PCI-X slot I installed it in, sometimes the box wouldn't even > boot. For perspective the 6850 has 4 PCI-e slots, and 3 PCI-X slots > (one 64-bit 133 MHz, two 32-bit 100 MHz). > > I believe the timing is only needed for music on hold, IAX trunking, > and MeetMe conferencing. We're not doing trunking or conferencing > (for now) so we're going with ztdummy. If the timing isn't perfect > only our music on hold will suffer, which is no big deal. If we run > into other problems, we might try popping our quad-span card in there > just to see if it works. > > Keep in mind that Digium no longer produces Wildcards. I'm not sure > why they don't work with our 6850 and the techs at Dell didn't know > either. Maybe they are not 100% PCI compliant. > > Matthew Roth > InterMedia Marketing Solutions > Software Engineer and Systems Developer > > Kevin Bockman wrote: > >> Chuck Bunn wrote: >> >>> Does anyone know if the Digium Wildcard will work on a PCI Express >>> or PCI-X motherboard. Specifically I am looking at the Dell 850 1U >>> rack server for use with Asterisk. >> >> >> >> They will work in PCI-X of course but not PCI Express. They are >> totally different. >> >> You will need the 3.3v cards. >> >> >> Kevin >> ___ >> --Bandwidth and Colocation sponsored by Easynews.com -- >> >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with date & time on Aastra 480i sincerelease 1.3
Does anyone know whether there is some sort of time zone option? I've emailed Aastra who didn't come back to me. I would like to set the time zone - e.g. Britain-London, in the cfg files so I don't have to set it on 40 phones... Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: 26 December 2005 16:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with date & time on Aastra 480i sincerelease 1.3 Jacques Leisy wrote: > Thanks Robert. I tried of course with time server disabled: 0 too. > Is it working for you? Which time server are you using, an external one? > Works for me and I'm using an internal one which is then synced to an external one. Try ONLY these entries. Remove the time format and date format and backup ntp servers: time server disabled: 0 time server1: 192.168.0.10 If this doesn't work, you should check your firewall rules (if any) and the versions of ntpd (4.2?) that you are running. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with date & time on Aastra 480isincerelease 1.3
Thanks, so would I be correct in assuming time zone name: UK-London time zone code: GMT time zone minutes: 0 And will this have any affect on the daylight savings in march? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 03 January 2006 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem with date & time on Aastra 480isincerelease 1.3 On Tue, 2006-01-03 at 14:13 +0000, Lee Archer wrote: > Does anyone know whether there is some sort of time zone option? I've > emailed Aastra who didn't come back to me. I would like to set the > time zone - e.g. Britain-London, in the cfg files so I don't have to > set it on 40 phones... > in aastra.cfg time server disabled: 0 time server1: 192.168.1.253 time format: 1 date format: 0 time zone name: FR-Paris time zone code: CET time zone minutes: 60 works for me. -- Dave Cotton <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with date & time on Aastra 480isincerelease 1.3
Still no joy, if I set my phone to a different time zone then reboot it isn't being updated to use London. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pete Barnwell Sent: 03 January 2006 14:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem with date & time on Aastra 480isincerelease 1.3 On Tue, 2006-01-03 at 14:13 +0000, Lee Archer wrote: > Does anyone know whether there is some sort of time zone option? I've > emailed Aastra who didn't come back to me. I would like to set the > time zone - e.g. Britain-London, in the cfg files so I don't have to > set it on 40 phones... time zone name: GB-London time zone code: GMT time zone minutes: 60 Rgds Pete ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
I had a problem which I spoke to Grandstream about. It seemed that around 7 seconds in it goes for time sync and if it fails it doesn't retry. This problem was highlighted by the .12 firmware and a Windows DHCP server we were using. Upon moving to a Linux DHCP server the process was much quicker and NTP worked. However there isn't an auto DST mode This upset a lot of people here where I work as all the clocks were wrong. Shame is these are reasonably cheap and fairly descent phones but we are now moving towards the Aastra range. I've tried out .13 and NTP worked fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: 31 December 2005 10:35 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP Hi all Slightly OT but I know a lot of GS experts hang out here - I just upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF functionality with Asterisk (which so far works as expected), but as a side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on the display. Anyone heard of this? Any workarounds (other than go back to 1.0.1.12) ? (Hmmm.. just regressed to 1.0.1.12 and it's still not working - curiouser and curiouser said Alice...) Thanks Peter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with date & time on Aastra480isincerelease 1.3
Actually it worked, but only after I defaulted all the settings on the phone and let it pick the config up fresh. Anyone know if there is any headset config options to default to headset/speaker? Thanks Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: 03 January 2006 14:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem with date & time on Aastra480isincerelease 1.3 Still no joy, if I set my phone to a different time zone then reboot it isn't being updated to use London. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pete Barnwell Sent: 03 January 2006 14:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem with date & time on Aastra 480isincerelease 1.3 On Tue, 2006-01-03 at 14:13 +0000, Lee Archer wrote: > Does anyone know whether there is some sort of time zone option? I've > emailed Aastra who didn't come back to me. I would like to set the > time zone - e.g. Britain-London, in the cfg files so I don't have to > set it on 40 phones... time zone name: GB-London time zone code: GMT time zone minutes: 60 Rgds Pete ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recommendations on a WiFi phone for *?
Don't waste your money. It works with Asterisk but it's a pain to setup and use. It's too expensive but at least the firmware is starting to get there. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk-User Sent: 10 January 2006 11:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *? Has anyone tried out Hitachi IPC-5000 ? It looks nice and it's a bit exensive, but I would still like to hear how does it behave around Asterisk. Ivan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Odd you should have this problem as I had exactly the same. In my case it was a slow DHCP server. Around 7 seconds in the phones tries to time sync. If the phone hasn't got an IP address then this time sync fails but it doesn't retry. I emailed Grandstream about it but got nowhere. I changed my DHCP server from Windows to Linux and now DHCP is much faster and time sync is working. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Edelbrock Sent: 21 January 2006 06:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP On Dec 31, 2005, at 7:28 AM, Ross C wrote: > Peter, > > After upgrading to 1.0.1.13 I had some miscellaneous problems on one > of my GXP-2000's--it would grab an IP address, but it wouldn't get the > time/date, it wouldn't register, blah blah blah. I could access the > web interface OK, so it wasn't a network issue (I don't think). > Anyway...I ended up resetting to factory defaults and all is well now. > Maybe try that? That has solved some other problems I've had as well. I just got a 2000 which does exactly this (our first for evaluation.. which is somewhat disappointing thus far). I could see in a packet sniffer a weird cycle of DHCP requests like it got an IP but kept retrying? A power cycle doesn't solve the problem (it's had many, and dozens of software resets). A reset with the MAC input doesn't work either for me. The phone was at an older FW when I got it (ending in .9, I think) and then updated to to the latest stable (.12 I think off the top of my head). Btw- the firmware update was a pain. HTTP updates were hitting the server (Apache) with 'bad request' results. I needed to set up my own tfpt server to make it work. Off lan updates weren't working, either, in any case. The phone will register and work when it has a static address assigned, but not when set for DHCP. In all cases, the clock is always wrong. I can see with a packet sniffer that the NTP request is sent and received, but with no effect on the phone display. Was there a resolution to this issue? The GXP-2000 seems to be a very popular phone, so I can't imagine others on the list not experiencing this? Or is this part of a batch with unresolvable problems that I need to send back to the seller? Thanks! TGIF! :') Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Installing beta2
Title: Installing beta2 Once built no matter whether I do make install or make clean I get the same output [EMAIL PROTECTED] asterisk]# make clean build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c make: *** [.depend] Interrupt I am using FC3 and any help would be appreciated. Regards Lee ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing beta2
Hi, I had removed all old versions before starting and downloaded from CVS. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ WeschkeSent: 02 November 2005 12:20To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Installing beta2 Are you installing over a previous source tree? If so, please rm -rf the previous source tree and install the new source tree from scratch. On 11/2/05, Lee Archer <[EMAIL PROTECTED]> wrote: Once built no matter whether I do make install or make clean I get the same output [EMAIL PROTECTED] asterisk]# make clean build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c make: *** [.depend] Interrupt I am using FC3 and any help would be appreciated. Regards Lee ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing beta2
Hi it says [EMAIL PROTECTED] ~]# rpm -q kernel-source zlib zlib-devel openssl openssl-devel package kernel-source is not installed zlib-1.2.1.2-3.fc3 zlib-devel-1.2.1.2-3.fc3 openssl-0.9.7a-42.1 openssl-devel-0.9.7a-42.1 Which is odd cos the sources are installed. I'm using the 2.6.9-1.667 kernel and have all the links to the build directory. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: 02 November 2005 13:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Installing beta2 Can you issue the following command on FC3 and let us know the results? rpm -q kernel-source zlib zlib-devel openssl openssl-devel On 11/2/05, Lee Archer <[EMAIL PROTECTED]> wrote: > > Hi, I had removed all old versions before starting and downloaded from CVS. > > Regards > > Lee > > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke > Sent: 02 November 2005 12:20 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Installing beta2 > > > > Are you installing over a previous source tree? If so, please rm -rf the previous source tree and install the new source tree from scratch. > > > On 11/2/05, Lee Archer <[EMAIL PROTECTED]> wrote: > > > > > > Once built no matter whether I do make install or make clean I get > > the same output > > > > [EMAIL PROTECTED] asterisk]# make clean build_tools/make_version_h > > > include/asterisk/version.h.tmp if cmp -s > > include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ > > mv include/asterisk/version.h.tmp include/asterisk/version.h > > ; \ fi rm -f include/asterisk/version.h.tmp > > build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c > > > > build_tools/make_version_h > include/asterisk/version.h.tmp if cmp > > -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ > > mv include/asterisk/version.h.tmp include/asterisk/version.h > > ; \ fi rm -f include/asterisk/version.h.tmp > > build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c > > > > make: *** [.depend] Interrupt > > > > I am using FC3 and any help would be appreciated. > > > > Regards > > > > Lee > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ### > > This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. > For more information, connect to http://www.f-secure.com/ > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRI
RE: [Asterisk-Users] spandsp-0.0.2pre21c broken?
I get an error when patching the makefile, seems the order is different. Had the same problem with rc1 and 2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: 13 November 2005 17:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] spandsp-0.0.2pre21c broken? Guys. Has anybody been able to compile spandsp-0.0.2pre21c against 1.2rc2? Seems spandsp-0.0.2pre21c is broken. :( Compiles great against 1.2rc1 but no luck so far with rc2. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN card required
Title: ISDN card required Can anyone point me in the direction of a quality, works with Asterisk, BRI card. I need minimum 2 port/4 channel. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN card required
Thanks to all. I'll probably go with the quadBri card they do. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristof Hardy Sent: 14 November 2005 14:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ISDN card required Lee Archer wrote: > Can anyone point me in the direction of a quality, works with > Asterisk, BRI card. I need minimum 2 port/4 channel. Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine. Cheers. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN card required
For us it boils down to the card with the less hassle. Anyone used this sirrix quad card? Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob LithSent: 14 November 2005 18:24To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] ISDN card required Junghanns is a very good card, Sirrix (www.sirrix.de) also do a good card with its own channel driver - saves hassels with BRIstuff needed with Jungahnns. In the end its down to personal preference. Sirrix comes in quad version, Junghans in quad and octo. RegardsRob On 11/14/05, Klaus Darilion <[EMAIL PROTECTED]> wrote: Kristof Hardy wrote:> Lee Archer wrote:>>> Can anyone point me in the direction of a quality, works with >> Asterisk, BRI card. I need minimum 2 port/4 channel.>>> Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine.Hi Kristof!(sorry for the empty email)Do you use it with asterisk 1.2 (CVS)? AFAIK the bristuff package for1.2 is quiet out-of-date.btw: have you ever used chan_misdn from beronet with quadBRI cards? Anyexperiences? regardsklaus___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN card required
I could do without playing with multiple versions of drivers trying to find one that works. And I could do with not spending days trying to make the card work with the ISDN lines. Bascially I could do with a card which has linux 2.6 drivers, works with Asterix and is documented. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 15 November 2005 09:26 To: Lee Archer Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ISDN card required Less hassle? Then I would recommend the Eicon Diva Server Cards. Active cards with full support for any ISDN line protocol, Modem, Fax and support with Asterisk via a generic channel driver (chan_capi, tested with other cards too). How less hassle do you need? Armin On Tue, 15 Nov 2005, Lee Archer wrote: > For us it boils down to the card with the less hassle. Anyone used > this sirrix quad card? > > Regards > > Lee > > > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith > Sent: 14 November 2005 18:24 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] ISDN card required > > > Junghanns is a very good card, Sirrix (www.sirrix.de) also do a good > card with its own channel driver - saves hassels with BRIstuff needed > with Jungahnns. In the end its down to personal preference. Sirrix > comes in quad version, Junghans in quad and octo. > > Regards > Rob > > > On 11/14/05, Klaus Darilion <[EMAIL PROTECTED]> wrote: > > Kristof Hardy wrote: > > Lee Archer wrote: > > > >> Can anyone point me in the direction of a quality, works with > > >> Asterisk, BRI card. I need minimum 2 port/4 channel. > > > > > > Ack. Like Mark pointed out, I also used Junghanns.net cards, works > fine. > > Hi Kristof! > > (sorry for the empty email) > > Do you use it with asterisk 1.2 (CVS)? AFAIK the bristuff package for > 1.2 is quiet out-of-date. > > btw: have you ever used chan_misdn from beronet with quadBRI cards? > Any > experiences? > > regards > klaus > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] receive fax with asterisk
I also get an error following the README when I run sh build. [EMAIL PROTECTED] iaxmodem-0.0.5]# sh buildiaxmodem.c: In function `printtime':iaxmodem.c:139: warning: passing arg 4 of `strftime' makes pointer from integer without a castiaxmodem.c: In function `main':iaxmodem.c:671: warning: passing arg 4 of `strftime' makes pointer from integer without a castiaxmodem.c:674: warning: passing arg 4 of `strftime' makes pointer from integer without a cast Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rusty DekemaSent: 17 November 2005 02:15To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] receive fax with asterisk I can't build the libiax2 that comes with it on my FreeBSD system :(. Unfortunately FreeBSD is not a supported platform.-Rusty On 11/16/05, Lee Howard <[EMAIL PROTECTED]> wrote: Jonathan k. Creasy wrote:>I am using the libiax2 that I just got out of CVS with "cvs checkout>libiax2">As the README states (clearly) you must use the libiax2 and the spandspthat come with IAXmodem.Lee.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] receive fax with asterisk
Ignore me, the build does complete. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 17 November 2005 08:19To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] receive fax with asterisk I also get an error following the README when I run sh build. [EMAIL PROTECTED] iaxmodem-0.0.5]# sh buildiaxmodem.c: In function `printtime':iaxmodem.c:139: warning: passing arg 4 of `strftime' makes pointer from integer without a castiaxmodem.c: In function `main':iaxmodem.c:671: warning: passing arg 4 of `strftime' makes pointer from integer without a castiaxmodem.c:674: warning: passing arg 4 of `strftime' makes pointer from integer without a cast Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rusty DekemaSent: 17 November 2005 02:15To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] receive fax with asterisk I can't build the libiax2 that comes with it on my FreeBSD system :(. Unfortunately FreeBSD is not a supported platform.-Rusty On 11/16/05, Lee Howard <[EMAIL PROTECTED]> wrote: Jonathan k. Creasy wrote:>I am using the libiax2 that I just got out of CVS with "cvs checkout>libiax2">As the README states (clearly) you must use the libiax2 and the spandspthat come with IAXmodem.Lee.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2 chan_modem not installing?
Title: 1.2 chan_modem not installing? After compiling the released version I get the follow error when I run asterisk. I didn’t get the fault with the beta's or rc's using the same config. I have tried the FTP version and the CVS downloaded version and get the same error Nov 17 11:53:50 VERBOSE[21915] logger.c: == Parsing '/etc/asterisk/modules.conf': Nov 17 11:53:50 VERBOSE[21915] logger.c: == Parsing '/etc/asterisk/modules.conf': Found Nov 17 11:53:50 VERBOSE[21915] logger.c: [chan_modem.so]Nov 17 11:53:50 WARNING[21915] loader.c: /usr/lib/asterisk/modules/chan_modem.so: cannot open shared object file: No such file or directory Nov 17 11:53:50 WARNING[21915] loader.c: Loading module chan_modem.so failed! Chan_modem isn't compiled…... Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1.2 chan_modem not installing?
I did download again but the problem was that chan_modem was # out of the Makefile so wasn't made. I removed the # and rebuild and its running fine now. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 17 November 2005 12:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1.2 chan_modem not installing? Lee Archer wrote: > After compiling the released version I get the follow error when I run > asterisk. I didn't get the fault with the beta's or rc's using the > same config. I have tried the FTP version and the CVS downloaded > version and get the same error > > Nov 17 11:53:50 VERBOSE[21915] logger.c: == Parsing > '/etc/asterisk/modules.conf': Nov 17 11:53:50 VERBOSE[21915] logger.c: > == Parsing '/etc/asterisk/modules.conf': Found > > Nov 17 11:53:50 VERBOSE[21915] logger.c: [chan_modem.so]Nov 17 > 11:53:50 WARNING[21915] loader.c: > /usr/lib/asterisk/modules/chan_modem.so: cannot open shared object > file: No such file or directory > > Nov 17 11:53:50 WARNING[21915] loader.c: Loading module chan_modem.so > failed! > > Chan_modem isn't compiled.. > > Regards > > Lee > > ### Straight from the 'Asterisk 1.2 Released!' (Note: for a short time, a tarball of Asterisk 1.2.0 was present on the FTP servers with a build problem related to the chan_modem drivers; this has been corrected, and if you downloaded the new version before receiving this announcement, please re-download to ensure you have the proper version.) Re-grab the tarball or source from the main FTP and re-try that :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXmodem
Title: IAXmodem Hi, I wonder if you can give me some pointers please. I have hylafax running, I've tested it with a modem off the serial port so I know the install does work, and I've installed IAXmodem to be able to fax out via asterisk. I've set everything up as in the README that comes with IAXmodem but im not getting the faxes sent. I can see hylafax sending to the IAXmodem but at this point something isn't working and I'm getting Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 7ms SCall: 4 DCall: 29764 [172.16.5.137:4569] CAUSE : No authority found Unknown IE 042 : Present in the iax log and no dialtone in the hylafax log. The IAXmodem is setup in my asterisk as an IAX2 extension. Any ideas? Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: 1.2 chan_modem not installing?
Ah that's why then. Is there something else I should set in my modules.conf? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: 17 November 2005 14:00 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: 1.2 chan_modem not installing? In article <[EMAIL PROTECTED]>, Lee Archer <[EMAIL PROTECTED]> wrote: > I did download again but the problem was that chan_modem was # out of > the Makefile so wasn't made. I removed the # and rebuild and its > running fine now. chan_modem has been obsoleted in 1.2. Unless you are specifically using it, you should edit /etc/asterisk/modules.conf to prevent it loading. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXmodem
I have in my iax cfg files... [EMAIL PROTECTED] asterisk]# more iax.conf [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes #include iax_additional.conf #include iax_custom.conf [EMAIL PROTECTED] asterisk]# [EMAIL PROTECTED] asterisk]# more iax_additional.conf [601] username=601 type=peer secret=password record_out=Never record_in=Never qualify=500 port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device <601> Am I incorrect to create it as an extension within asterisk? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: 17 November 2005 14:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXmodem Lee Archer wrote: > Hi, I wonder if you can give me some pointers please. I have hylafax > running, I've tested it with a modem off the serial port so I know the > install does work, and I've installed IAXmodem to be able to fax out > via asterisk. I've set everything up as in the README that comes with > IAXmodem but im not getting the faxes sent. I can see hylafax sending > to the IAXmodem but at this point something isn't working and I'm > getting > > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > REJECT >Timestamp: 7ms SCall: 4 DCall: 29764 [172.16.5.137:4569] >CAUSE : No authority found >Unknown IE 042 : Present > > in the iax log and no dialtone in the hylafax log. > > The IAXmodem is setup in my asterisk as an IAX2 extension. > The "no dialtone" response from IAXmodem indicates that iaxmodem is not able to register with Asterisk. Double-check that you have entries in iax.conf for IAXmodem and that you've reloaded Asterisk. The IAX-REJECT "No authority found" also seems to indicate that the entries in iax.conf do not exist. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXmodem
I still get the same messages. However registration with asterisk is happening. Asterisk -- Registered IAX2 '601' (AUTHENTICATED) at 172.16.5.137:32771 IAXmodem Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00010ms SCall: 2 DCall: 01413 [172.16.5.137:4569] USERNAME: 601 DATE TIME : 192036682 REFRESH : 60 APPARENT ADDRES : IPV4 172.16.5.137:32771 MESSAGE COUNT : 0 CALLING NUMBER : 601 CALLING NAME: device Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00010ms SCall: 01413 DCall: 2 [172.16.5.137:4569] Registration completed successfully. My system is setup with 9 for an external line, am I correct in entering 9+dest fax number in the hylafax print box? Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: 17 November 2005 16:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXmodem Lee Archer wrote: >disallow=all >allow=ulaw >allow=alaw >allow=gsm > IAXmodem uses slinear. allow=slinear Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ip phone
We had 1 way speech on them for a while but the latest firmware seems to have fixed it. The 10mb LAN ports in the back are old too. Also I wouldn't recommend the GXP-2000 either. We have a few here. As a basic phone it's fine but don't try anything fancy like PoE as ours keeps failing and we have to run them off the PSU. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: 18 November 2005 09:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ip phone On Fri, 2005-11-18 at 15:04 +0700, stevanus wrote: > Hi, > > Maybe grandstream budgetone 100 series will fulfill your requirement. > It's very good for such a cheap sub-50 phone. > Once, I've tested and I've found myself that it's a good performer > (even it has compatibility problem with old switch in my office :P) > You can search the supplier through googling it. Don't ask me as I > don't know any information about it. I have heard bad things about that phone. Specifically audio quality is questionable, the power connector that ships is the wrong size so it tends to fall out, there are firmware issues that locks the phone up, etc. Does anyone have any experience with that phone specifically? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXmodem
Title: IAXmodem Thanks for the help. I got it working, type=friend Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 17 November 2005 13:53To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] IAXmodem Hi, I wonder if you can give me some pointers please. I have hylafax running, I've tested it with a modem off the serial port so I know the install does work, and I've installed IAXmodem to be able to fax out via asterisk. I've set everything up as in the README that comes with IAXmodem but im not getting the faxes sent. I can see hylafax sending to the IAXmodem but at this point something isn't working and I'm getting Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 7ms SCall: 4 DCall: 29764 [172.16.5.137:4569] CAUSE : No authority found Unknown IE 042 : Present in the iax log and no dialtone in the hylafax log. The IAXmodem is setup in my asterisk as an IAX2 extension. Any ideas? Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ver1.2 installation problem
I had this problem with Fedora. I updated the kernel to the latest one available for core 3 and changes the links to point to the new source code. It worked fine then. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Wegrzyn - asterisk Sent: 23 November 2005 03:42 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ver1.2 installation problem Hi, After I compile asterisk v.1.2 is tells me that last thing to do is to make install. Unfortunately it goes it to loop after I type make install this is the loop: else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c en
[Asterisk-Users] Aastra 1.3 firmware
Title: Aastra 1.3 firmware Has anyone had any luck with the BLF option yet? I have set up as per the manual/front end, configured the hints in Asterisk and nothing shows. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra 1.3 firmware
Title: Aastra 1.3 firmware As always right after asking it works Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 23 November 2005 11:09To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Aastra 1.3 firmware Has anyone had any luck with the BLF option yet? I have set up as per the manual/front end, configured the hints in Asterisk and nothing shows. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra 1.3 firmware
Title: Aastra 1.3 firmware Yes I had also noticed this. Also setting dns2 to 0.0.0.0 in the config file is ignored and I couldn't set the timezone via the config I had to configure it on the phone. Anyone have any other issues? Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos ChavezSent: 23 November 2005 19:44To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Aastra 1.3 firmware On Wed, 2005-11-23 at 11:08 +0000, Lee Archer wrote: Has anyone had any luck with the BLF option yet? I have set up as per the manual/front end, configured the hints in Asterisk and nothing shows. RegardsLee It works for me but if you use a BLF then you cannot use the same button for speed dial which makes this option worthless. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax sending problems
Title: Fax sending problems Hi, I've got iaxmodem setup but I'm getting failed fax sending. When I send a fax it is spooled through the system and I hear the destination fax machine pick up, it's sat near me, and the transfer starts. However after about 30 seconds the line drops and the fax machine reports an error. The faxsend process still thinks it is running and has to be killed. The logs report Nov 24 10:50:12.89: [ 8222]: SEND send frame number 132 Nov 24 10:50:12.89: [ 8222]: SEND send frame number 133 Nov 24 10:50:12.89: [ 8222]: MODEM set XON/XOFF/FLUSH: input interpreted, output disabled Nov 24 10:50:12.89: [ 8222]: DELAY 200 ms Nov 24 10:50:13.09: [ 8222]: <-- [11:AT+FTM=146\r] Nov 24 10:50:13.10: [ 8222]: --> [7:CONNECT] Nov 24 10:50:13.10: [ 8222]: DELAY 400 ms Nov 24 10:50:13.50: [ 8222]: <-- data [1025] Nov 24 10:50:13.50: [ 8222]: <-- data [1027] Nov 24 10:50:13.50: [ 8222]: <-- data [1029] Nov 24 10:50:13.50: [ 8222]: <-- data [1029] Nov 24 10:50:15.02: [ 8222]: <-- data [1029] Nov 24 10:50:15.02: [ 8222]: <-- data [1033] Nov 24 10:50:15.02: [ 8222]: <-- data [1034] Nov 24 10:50:15.02: [ 8222]: <-- data [1031] Nov 24 10:51:15.01: [ 8222]: MODEM TIMEOUT: writing to modem Nov 24 10:51:15.01: [ 8222]: MODEM WRITE SHORT: sent 1031, wrote 984 Nov 24 10:51:15.01: [ 8222]: SEND end page Nov 24 10:51:15.01: [ 8222]: Unspecified Transmit Phase C error Nov 24 10:51:15.01: [ 8222]: <-- [9:AT+FTH=3\r] Nov 24 10:51:22.57: [ 8222]: MODEM TIMEOUT: sending HDLC frame Nov 24 10:51:22.57: [ 8222]: MODEM input buffering enabled Nov 24 10:51:22.57: [ 8222]: <-- [5:ATH0\r] Is this an error with my modem or the receiving fax machine? Regards Lee ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] v1-2 install mkdep loop
I found running a later kernel and source code fixed it. I had it on Fedora Core 3 using kernel 2.6.9 but after updating to 2.6.12 the problem went away. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: 24 November 2005 14:16 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] v1-2 install mkdep loop On Mon, 21 Nov 2005, Bob Knight wrote: > Just pulled a v1-2 onto a system that was running a v1-0. > > Zaptel and libpri, build and install just fine. > Building asterisk is fine. > But when I try to do a make install on asterisk, it goes into an > infinite loop doing on .depend doing: build_tools/mkdep > > I did the same thing on another box the other day with a different > pull and did not have any problems. Do you think this is something > related to this box? Hi Bob! Long live the PM3! This is an issue that many many people have been running into, and has been discussed on the dev list. Check the following: http://lists.digium.com/pipermail/asterisk-dev/2005-November/016509.html I'm not sure there is a specific fix, although there are many suggestions in that thread. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 codec problems
I have a system that has had 5 G729 licenses for over a year and I've come to install the v31 G729 codec from the Digium ftp server but it won't see the license. Does anyone know how to get around this problem? It is registered and I do have newer systems running this v31 version of the codec but in the license file the product line is different. On the older system it says Product: Digium-G729 but on the newer systems it says Product: G.729 Codec. I've tried Digium support but had no reply and thought I'd try the list. Regards Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queue forks asterisk and then leaves the extraprocesses lying around
Are you using freePBX by any chance?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel
Roberts
Sent: 08 November 2006 08:55
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queue forks asterisk and then leaves the
extraprocesses lying around
Hi,
I have a problem with Queue where by a call comes in to the queue and if
all the phones are busy and the queue reaches the timeout, it will fork
a process and leave it sitting there before going off to the next step
in the dial plan and continuing normally. This doesn't cause any
problems except for I assume that it will eventually use up all the
memory on the machine and it messes with my process monitoring.
It doesn't seem to matter what I have as the next step after the Queue
command and it happens only sometimes. It seems like it might even be a
timing issue given that it's less likely to happen if any one of the
phones ring.
The new asterisk processes that get started up look like they think
they're new asterisk instances or though they don't actually do anything
or interfere with the first asterisk instance.
Has anyone had any problems like this? Am I doing something wrong?
The appropriate part of my dial plan looks like this:
exten => 101,1,Answer
exten => 101,n,GotoIf($["${CONTEXT}"="from-internal"]?USERCID:SETCID)
exten => 101,n(USERCID),Macro(user-callerid,)
exten => 101,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME})
exten =>
101,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${TIMES
TAMP}-${UNIQUEID})
exten => 101,n,Queue(101|tr|||30)
exten => 101,n,Goto(ext-local,83,1)
exten => 101*,1,Macro(agent-add,101,)
exten => 101**,1,Macro(agent-del,101,101)
and from queues.conf
[101]
wrapuptime=0
timeout=15
strategy=ringall
retry=5
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=
member=Local/[EMAIL PROTECTED],0
member=Local/[EMAIL PROTECTED],0
maxlen=2
leavewhenempty=no
joinempty=Yes
context=
announce-holdtime=no
announce-frequency=0
and some logs to show what I mean by the new asterisk process thinking
that it is actually a new asterisk.
-- snip --
Nov 8 21:44:38 DEBUG[25896] channel.c: Hanging up channel
'Local/[EMAIL PROTECTED],2'
Nov 8 21:44:38 DEBUG[25627] devicestate.c: Changing state for
Local/[EMAIL PROTECTED] - state 0 (Unknown) Nov 8 21:44:38 DEBUG[25904]
app_queue.c: Device 'Local/[EMAIL PROTECTED]' changed to state '0'
(Unknown) Nov 8 21:44:38 DEBUG[25897] cdr_addon_mysql.c: cdr_mysql:
inserting a CDR record.
Nov 8 21:44:38 DEBUG[25897] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode,uniqueid) VALUES
('2006-11-08 21:44:38','49761450','49761450','83','from-internal',
'Local/[EMAIL PROTECTED],2','','AGI','recordingcheck|20061108-214438
|1162975478.49',0,0,'NO ANSWER',3,'','1162975478.49') Nov 8 21:44:38
DEBUG[25897] channel.c: Hanging up channel
'Local/[EMAIL PROTECTED],2'
Nov 8 21:44:38 DEBUG[25627] devicestate.c: Changing state for
Local/[EMAIL PROTECTED] - state 0 (Unknown) Nov 8 21:44:38 DEBUG[25905]
app_queue.c: Device 'Local/[EMAIL PROTECTED]' changed to state '0'
(Unknown)
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/extconfig.conf': Nov 8 21:44:38 DEBUG[25902]
config.c:Parsing /etc/asterisk/extconfig.conf
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/extconfig.conf': Found
Nov 8 21:44:38 VERBOSE[25902] logger.c: == Parsing
'/etc/asterisk/manager.conf': Nov 8 21:44:38 DEBUG[25902] config.c:
Parsing /etc/asterisk/manager.conf
... lots of asterisk start up logs ...
-- snip --
Regards,
Nigel
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RE: [asterisk-users] Queue forks asterisk and then leaves theextraprocesses lying around
Hi, have a look at http://www.freepbx.org/trac/ticket/1174 it's currently in the bug list. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel Roberts Sent: 08 November 2006 09:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue forks asterisk and then leaves theextraprocesses lying around Hi Lee, On Wed, 08 Nov 2006 at 09:00:27 -0000, Lee Archer wrote: > Are you using freePBX by any chance? Yes, version 2.1.1. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17
I installed zaptel 1.2.17 and shortly afterwards got a problem of calls not
clearing properly. I ran dmesg which showed
Unable to handle kernel NULL pointer dereference at virtual address
009c
printing eip:
f8a79fa8
*pde =
Oops: [#1]
Modules linked in: zttranscode button battery ac ipv6 edd wcte11xp
zaptel crc_ccitt i2c_i801 i2c_core tg3 generic shpchp pci_hotplug parport_pc lp
parport dm_mod ext3 jbd sg fan thermal processor 3w_ piix sd_mod scsi_mod
ide_disk ide_core
CPU:0
EIP:0060:[]Tainted: G U VLI
EFLAGS: 00010082 (2.6.13-15.15-default)
EIP is at zt_chanandpseudo_ioctl+0xd28/0xf70 [zaptel]
eax: ebx: f74403ac ecx: edx:
esi: b723f2b0 edi: f749ca78 ebp: 0046 esp: f50b3e28
ds: 007b es: 007b ss: 0068
Process asterisk (pid: 5430, threadinfo=f50b2000 task=f7bbf060)
Stack: 462f0587 41a0d314 01ff 0001 0246
0001
f50b3f38 005b 0001 dfcf089c
f50b3ebc
f50b3efc f61ae400 f6a0d3b4 005b 005b f6a0d314
0001
Call Trace:
[] generic_file_aio_write+0x58/0xc0
[] ext3_file_write+0x1b/0x93 [ext3]
[] do_sync_write+0xb6/0x110
[] zt_ioctl+0x93/0x100 [zaptel]
[] zt_ioctl+0x0/0x100 [zaptel]
[] do_ioctl+0x4e/0x60
[] vfs_ioctl+0x4f/0x1c0
[] sys_ioctl+0x37/0x70
[] sysenter_past_esp+0x54/0x79
Code: ff 89 f8 89 f1 e8 75 88 77 c7 31 ff c7 85 94 06 00 00 00 00 00 00
e9 77 f4 ff ff 8b 4c 24 24 e9 e4 f8 ff ff 8b 04 95 20 0d aa f8 <8b> 80 9c 00 00
00 e8 5d a9 6c c7 8b 44 24 20 8b 04 85 20 0d aa
I've since installed zaptel 1.2.16 again and it's fine. Is anyone else getting
this problem?
Regards
Lee
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RE: [asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17
It was fixed in 1.2.17.1. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 26 April 2007 21:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17 On Wed, Apr 25, 2007 at 08:57:37AM +0100, Lee Archer wrote: > I installed zaptel 1.2.17 and shortly afterwards got a problem of > calls not clearing properly. I ran dmesg which showed > > Unable to handle kernel NULL pointer dereference at virtual address 009c > printing eip: > f8a79fa8 > *pde = > Oops: [#1] > Modules linked in: zttranscode button battery ac ipv6 edd wcte11xp zaptel crc_ccitt i2c_i801 i2c_core tg3 generic shpchp pci_hotplug parport_pc lp parport dm_mod ext3 jbd sg fan thermal processor 3w_ piix sd_mod scsi_mod ide_disk ide_core > CPU:0 > EIP:0060:[]Tainted: G U VLI > EFLAGS: 00010082 (2.6.13-15.15-default) > EIP is at zt_chanandpseudo_ioctl+0xd28/0xf70 [zaptel] > eax: ebx: f74403ac ecx: edx: > esi: b723f2b0 edi: f749ca78 ebp: 0046 esp: f50b3e28 > ds: 007b es: 007b ss: 0068 > Process asterisk (pid: 5430, threadinfo=f50b2000 task=f7bbf060) > Stack: 462f0587 41a0d314 01ff 0001 0246 0001 > f50b3f38 005b 0001 dfcf089c f50b3ebc > f50b3efc f61ae400 f6a0d3b4 005b 005b f6a0d314 0001 > Call Trace: >[] generic_file_aio_write+0x58/0xc0 >[] ext3_file_write+0x1b/0x93 [ext3] >[] do_sync_write+0xb6/0x110 >[] zt_ioctl+0x93/0x100 [zaptel] >[] zt_ioctl+0x0/0x100 [zaptel] >[] do_ioctl+0x4e/0x60 >[] vfs_ioctl+0x4f/0x1c0 >[] sys_ioctl+0x37/0x70 >[] sysenter_past_esp+0x54/0x79 > Code: ff 89 f8 89 f1 e8 75 88 77 c7 31 ff c7 85 94 06 00 00 00 00 00 > 00 e9 77 f4 ff ff 8b 4c 24 24 e9 e4 f8 ff ff 8b 04 95 20 0d aa f8 <8b> > 80 9c 00 00 00 e8 5d a9 6c c7 8b 44 24 20 8b 04 85 20 0d aa > > I've since installed zaptel 1.2.16 again and it's fine. Is anyone else getting this problem? Not me, but others do. Try 1.2.17.1 . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX trunk problem
I wonder if anyone can help me with this. I have 4 sites running asterisk and calls coming into any of these sites are received locally and forwarded to a central operator. E.g. Call comes in on site A and is forwarded to the operator on site B. 99/100 the operator will send the call back to the site from where it came but site B's Asterisk server seems to be staying in the loop. E.g. A > B > A. I've had a look and can't see anything obvious as I had assumed that asterisk would pass the call off. Thanks Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX trunk problem
> I wonder if anyone can help me with this. I have 4 sites running > Asterisk and these are linked via IAX trunks and ADSL lines. Calls > coming into any of these sites are received locally and forwarded to a > central operator. E.g. Call comes in on site A and is forwarded to > the operator on site B. 99 out of 100 times the operator will send > the call back to someone at the site from where it came but site B's > Asterisk server seems to be staying in the loop. E.g. A > B > A. > I've had a look and can't see anything obvious as I had assumed that > Asterisk would pass the call off. I've tried notransfer on the trunks > but site B's Asterisk server doesn't seem to be joining the endpoints > and staying in the loop and therefore the call is going over the > trunks twice. > > Thanks > > Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_zap.c: Failed to read gains: Invalid argument
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P card in E1 mode. I've recently noticed in my logs the following Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI) Jan 5 01:27:11 VERBOSE[22490] logger.c: == Parsing '/etc/asterisk/zapata.conf': Jan 5 01:27:11 VERBOSE[22490] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 1, with 0 conference users Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 1, PRI Signalling signalling Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 2, with 0 conference users Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 2, PRI Signalling signalling Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 3, with 0 conference users Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 3, PRI Signalling signalling Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 4, with 0 conference users Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 4, PRI Signalling signalling Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 5, with 0 conference users Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 5, PRI Signalling signalling Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 6, with 0 conference users Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 6, PRI Signalling signalling Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 7, with 0 conference users Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 7, PRI Signalling signalling Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 8, with 0 conference users Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 8, PRI Signalling signalling Jan 5 01:27:11 VERBOSE[22490] logger.c: -- Automatically generated pseudo channel Jan 5 01:27:11 VERBOSE[22490] logger.c: == Starting D-Channel on span 1 Which seems to suggest that I've done something wrong with the rx and txgain option in /etc/asterisk/zapata.conf. But these haven't been changed in 18 months and still say ; You may also set the default receive and transmit gains (in dB) ; rxgain=4.0 txgain=0.0 Have I done something wrong or has something changed? Thanks Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
Sorry I should have stated that I've tried +x, -x, x.y and x and I still get the same. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling Sent: 05 January 2007 08:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument Lee Archer wrote: > I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium > TE110P card in E1 mode. I've recently noticed in my logs the > following > > Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 01:27:11 > VERBOSE[22490] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI) > > Jan 5 01:27:11 VERBOSE[22490] logger.c: == Parsing > '/etc/asterisk/zapata.conf': Jan 5 01:27:11 VERBOSE[22490] logger.c: > == Parsing '/etc/asterisk/zapata.conf': Found > > Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid > argument Jan 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read > gains: Invalid > > Which seems to suggest that I've done something wrong with the rx and > txgain option in /etc/asterisk/zapata.conf. But these haven't been > changed in 18 months and still say > > ; You may also set the default receive and transmit gains (in dB) ; > rxgain=4.0 txgain=0.0 > > Have I done something wrong or has something changed? Don't use fractional gains. i.e. use rxgain=4 and txgain=0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
Yes I get the same message after reload chan_zap.so Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 05 January 2007 08:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote: > I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium > TE110P card in E1 mode. I've recently noticed in my logs the > following > > Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 01:27:11 > VERBOSE[22490] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI) > Jan 5 01:27:11 VERBOSE[22490] logger.c: == Parsing > '/etc/asterisk/zapata.conf': Jan 5 01:27:11 VERBOSE[22490] logger.c: > == Parsing '/etc/asterisk/zapata.conf': Found Jan 5 01:27:11 > DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan 5 > 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid > argument This is a debug message and not even a warning message. I'm not sure that this is something to worry about. The code there tries to first read the gains and set the gains based on them. The return value from the ioctl that sets the gains does not seem to be checked in several code pathes, though. So it may actually fail silently. Do you get the same debug messages on 'reload chan_zap.so' ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
So anyone else any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 05 January 2007 09:30 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument On Fri, Jan 05, 2007 at 10:53:17AM +0200, Tzafrir Cohen wrote: > On Fri, Jan 05, 2007 at 07:47:15AM -0000, Lee Archer wrote: > > I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium > > TE110P card in E1 mode. I've recently noticed in my logs the > > following > > > > Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 > > 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI) > > Jan 5 01:27:11 VERBOSE[22490] logger.c: == Parsing > > '/etc/asterisk/zapata.conf': Jan 5 01:27:11 VERBOSE[22490] logger.c: > > == Parsing '/etc/asterisk/zapata.conf': Found Jan 5 01:27:11 > > DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan > > 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid > > argument > > This is a debug message and not even a warning message. I'm not sure > that this is something to worry about. Sorry, my stupid misreading of the code. If this message was given, ZT_SETGAINS will not be called. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra480i. Which one to choose ?
Aren't Aastra due to release new phones and some form of operator/reception addon? The Aastra user/admin guides are a lot more up2date that they used to be. I'd like Aastra to add a GSM codec to their phone and have a more regular firmware release schedule. I agree with the list below though that Polycom does have a better line up currently, and especially point 7 - when rebooting the phone please don't drop the network ports. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: 23 January 2007 02:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra480i. Which one to choose ? Here are another $0.02 We too have put in a lot of polycoms and aastras. I agree with a lot of what you noted below...but there are two big strikes against aastra: 1. Firmware bugs. Even some basic functions of the 480i are unusable/unstable due to firmware bugs. The word from support is always "wait for the next firmware" 2. Poor documentation. Their documentation is out of date and lacking a LOT of critical functions. (eg: Try to setup a hold button on the wireless handset using a config file) We're steering more customers towards polycom now. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, January 22, 2007 9:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra 480i. Which one to choose ? With over 300 Polycoms, and around 80 Aastra 480i under my belt here is my $0.02. 1. Sound quality, Polycom wins but the Aastra has excellent sound quality as well. 2. Complete product line, Polycom wins. 3. Cordless Aastra, although it's not the best cordless. 4. Backlit, Aastra 5. PoE, Aastra 6. Speakerphone, they both have good speaker phones. Although in general the answer to 1 goes here as well. 7. They both have 2 network ports, but I havnt' done any tests on the speed, I did however notice that when restarting the phone, the Polycom will not shut the network ports down, while the Aastra will. On another note, in general the Polycoms give me less problems. The Aastras are not yet that stable. See my next post to the list. On 1/22/07, Bruce Reeves <[EMAIL PROTECTED]> wrote: > Your list seems to lean heavily to the Aastra, while I choose the > Polycom > 501/601 over the Aastra, I did like the unit I tested and the > cordless. In the end the fact that most of the people using the phones > would use the speaker phone, Polycom and their history of conference > phones made the choice. We rolled 75 phones at one site and another 30 > now at remote locations. As far a a receptionist phone, we choose to > use a software operator panel instead of a phone that took up most of > the desk, there were initial concerns but the results have been > excellent. If you have not already done so grab a few people from > different parts of the office and have them give their 2 cents, it > will help to have their perspectives on the quality and feel of the phones. > > > On 1/22/07, Vikas <[EMAIL PROTECTED]> wrote: > > I need to provide a 80 people office with VOIP. > > > > I want to commit to one vendor Polycom or Aastra. Price of the > > phones is not a factor in the decision. The quality of the phones is > > the factor. > > > > Some of the features that I am evaluating on are: (arranged in order > > of priority) 1. Sound quality 2. complete product line with > > conference phone and receptionist phone (not on Aastra) 3. cordless > > (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not > > on 501) 6. speaker phone 7. 2 network ports. > > > > Which one will you choose ? > > > > Vikas > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > Bruce > Nortex Networks > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more info
RE: [asterisk-users] Music on Hold on IP Phones with FreePBX 2.2.0
Have you tried the #freepbx IRC channel or the freepbx mailing list? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Arnilo S. Baluyos (Mailing Lists) Sent: 23 January 2007 01:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Music on Hold on IP Phones with FreePBX 2.2.0 Hello everyone, We just installed a new Trixbox 2.0 server and updated FreePBX to 2.2.0 from 2.2.0rc3. We are having some problems with regards to Music on Hold on IP phones. When we press the "Hold" button, the caller doesn't hear the MOH sound. This functionality used to work with the older [EMAIL PROTECTED] installation on the same hardware and configuration. However, we don't have any problems with softphones only on IP phones. Is there anyone also having the same problem? Best regards, Matt -- Stand before it and there is no beginning. Follow it and there is no end. Stay with the ancient Tao, Move with the present. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TE110P and HDLC problems
I had this problem and in the end it appeared to be slot timing on the mobo. I had to put the TE110P in the 1st slot - which happened to be a PCI-X slot. That was using a Supermicro motherboard too. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 25 January 2007 20:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE110P and HDLC problems There was a recent driver fix that *might* help you. It's not in an official 1.x.x release yet, but if you check out 1.2 from svn, you should get the latest version of the driver with the fix. Matthew Fredrickson On Jan 25, 2007, at 9:15 AM, Marc Patino Gómez wrote: > Hi!, > > this issue makes me crazy. I read a lot of docs, also * mailling list > and I try a lot of things without success. > > Any help will be appreciated. Here is the info: > > Hardware: > > Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon > 5050 Digium TE110P > > Software > - > Asterisk version 1.2.12.1 > Zaptel version 1.2.8 > > /etc/zaptel.conf > > loadzone=es > defaultzone=es > span=1,1,0,ccs,hdb3,crc4 > bchan=1-15 > dchan=16 > bchan=17-31 > > The dammed errors: > > Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got > event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jan 25 14:11:52 > NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got > event: HDLC Abort (6) on Primary D-channel of span 1 Jan 25 14:11:52 > NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got > event: HDLC Abort (6) on Primary D-channel of span 1 ... > > I tried the following without success: > > - Disable Hyper Threading. > - Disable NIC's, RAID controller, usb, serial... to avoid shared IRQ, > so TE110P has his own IRQ as shows lspci -vb. > - Also I tried with APIC and without APIC. > .. > > > These HDLC errors appear when I physically loop the E1 interface in > the Card and also appear, and more frequently, when I connect the E1 > circuit (from the Telco) to the interface of the Card. > > > Thanks a lot > > -- > --- > - > > Marc Patino Gómez > Dpto. Sistemas > > Claranet España. Servicios Internet > C/General Almirante 2-28, Torres Cerdá > 08014 Barcelona > Tel. Información General: 902 884 633 Tel. Soporte Técnico: 902 884 622 > Fax: +34 93 445 19 20 > www.claranet.es > > Claranet Group: United Kingdom - Spain - France - Germany - Portugal - > Netherlands - USA > > --- > - > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] freepbx with ASTERISK 1.4
Yes check the freepbx website, and in particular trac bug #1610. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of younss azzayani Sent: 16 February 2007 11:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] freepbx with ASTERISK 1.4 Hi everybody, it's possible to configure freepbx 2.2 with asterisk 1.4? Have a nice day Younss AZ KASTERISK.COM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] freepbx with ASTERISK 1.4
I said what to do before. http://freepbx.org/trac/ticket/1610 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M. Sent: 16 February 2007 14:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] freepbx with ASTERISK 1.4 On Fri, 2007-02-16 at 09:33 -0500, McGhee, Stefano wrote: > > it's possible to configure freepbx 2.2 with asterisk 1.4? > > Look here for the archives: > > http://lists.digium.com/pipermail/asterisk-users/ > > Search for the subject "FreePBX 2.2.0 and Asterisk 1.4.0". > > You'll find EXACTLY what you're looking for. :-) > Look at: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.amportal.user /5377 Regards, > Stefano > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
