[Asterisk-Users] Asterisk At Home Snom Hints
Hi, everybody. I don´t know if it is an * or an AAH issue - I can´t get the Snom-Phone-hints working under AAH 1.5 running * 1.0.9. I tried with the Snom 360 softphone and it just doesn´t work. Is there any known issue? Is there a AAH mailing list available? Thank you in advance. Best regards, Armin Lediger ___ --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] [EMAIL PROTECTED] Music on Hold
Hello, What could be the problem if [EMAIL PROTECTED] is not starting mpg123 even though I did not touch the MOH-config files? There is no error message in asterisk at debug/verbose level 9. It seems asterisk doesn´t even launch mpg123, but it´s hard to say - maybe it launches it for 1 second and I just can´t see it... I have googled a lot lately but could not find any hints how to solve this problem... I am running [EMAIL PROTECTED] at version release 1.0.9. mpg123r ist the mpg123 release. What could I do? Could it be a class problem? Any help I would really appreciate. Best regards, Armin Lediger ___ --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] Chan_sccp and dynamic DNS
Hi, everybody. I was just running into a DynDNS Problem. One of our phones has a fixed IP, but the * box only has a dynDNS IP and a fixed name. So all phones running SIP are able to survive the change of the IP address, but not our Cisco 7920 - it changes state to connecting to CM0 and hangs there until reboot. After reboot it looks up the current IP address of the * box and works fine for the next 24 hours. My question now is: Is there an option in sccp.conf or somewhere else to tell the 7920 to do DNS lookups once in a while so it updates the IP address dynamically? Any help there would be great! Thanks in advance. Armin Lediger -- HotZone GmbH Würzburg - schnurlos glücklich! WLAN und VoIP Dienstleistungen Arndtstr. 5 97072 Würzburg [EMAIL PROTECTED] Voice 0931-2056064 Fax 0931-2056063 ___ 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] WG: Cisco 7920 WLAN Phone
Hi, I´ve been following some information about the latest chan_sccp drivers for a few weeks. I installed and tried about any variant of driver that exists in the chan_sccp system. Despite a few little changes in the CLI options I am still encountering a lot of problems with the phone. When not using the phone for a while, it disconnects from asterisk (1.0.7) after a few hours and tells on the display connecting to callmanager1 so it seems like it is ignoring the settings for callmanager0 (=asterisk box). Then I have to restart the phone to get a connection to asterisk. Is there anybody out there who has successfully worked with the 7920 phone for longer than a few days and who has some hints on adapting the functionality of the chan_sccp driver to this phone in order to get a stable connection between phone and asterisk. Best regards, Armin Lediger -- HotZone GmbH Würzburg - schnurlos glücklich! WLAN und VoIP Dienstleistungen Arndtstr. 5 97072 Würzburg [EMAIL PROTECTED] Voice 0931-2056064 Fax 0931-2056063 ___ 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
AW: [Asterisk-Users] WG: Cisco 7920 WLAN Phone
Armin Lediger ha scritto: When not using the phone for a while, it disconnects from asterisk (1.0.7) after a few hours and tells on the display what chan_sccp are you running? Hi. I am currently using the chan_sccp-cm in the updated version. Best regards, Armin ___ 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
AW: [Asterisk-Users] call pickup with snom function keys now working with cvs-head + patch sipsubscribe-20050715.rev779.txt
hi listmembers, please test my new patch to chan_sip.c which is to make call pickup on the snom phones (and maybe other phones that support 'INVITE/Replaces') work and make comments in the bugtracker http://bugs.digium.com/view.php?id=3644 so it can make its way into the cvs. This really sounds very exciting. Excuse my beginner´s question: Where and how do I get this patch and how do I install a patch like this in an environment that should not be stopped for more than a few minutes? Is it any easy step to install this patch? Best regards, Armin ___ 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
AW: [Asterisk-Users] WG: Cisco 7920 WLAN Phone
It looks like you are running a old firmware or the locale skinny file (SCCP-dictionary.xml) is not present in the TFTP server. Oh, good ideas. I hadn't justified the cisco contract yet for firmwares, but maybe I soon have :) I'll check out the dictionary first. There is a new firmware version for this phone. I upgraded about 1 week ago to cmterm_7920.4.0-02-00.bin expecting some more stable functionality. Get it and try out - you should then not have any Euro characters any longer. Best regards, Armin ___ 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] Cisco 7920 WLAN Phone
Hi, I just wrote these lines to asterisk-users@lists.digium.com not knowing that there is a separate mailing list for chan_sccp stuff. So here it comes: I´ve been following some information about the latest chan_sccp drivers for a few weeks. I installed and tried about any variant of driver that exists in the chan_sccp system. Despite a few little changes in the CLI options I am still encountering a lot of problems with the phone. When not using the phone for a while, it disconnects from asterisk (1.0.7) after a few hours and tells on the display connecting to callmanager1 so it seems like it is ignoring the settings for callmanager0 (=asterisk box). Then I have to restart the phone to get a connection to asterisk. Is there anybody out there who has successfully worked with the 7920 phone (firmware cmterm_7920.4.0-02-00.bin) for longer than a few days and who has some hints on adapting the functionality of the chan_sccp driver to this phone in order to get a stable connection between the phone and asterisk. Best regards, Armin Lediger -- HotZone GmbH Würzburg - schnurlos glücklich! WLAN und VoIP Dienstleistungen Arndtstr. 5 97072 Würzburg [EMAIL PROTECTED] Voice 0931-2056064 Fax 0931-2056063 ___ 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
AW: AW: [Asterisk-Users] WG: Cisco 7920 WLAN Phone
You´re welcome! Keep us informed on your 7920 experiences. :-) Armin Great, I should have of course updated the firmware as a first. Thank you! Armin Lediger wrote: It looks like you are running a old firmware or the locale skinny file (SCCP-dictionary.xml) is not present in the TFTP server. Oh, good ideas. I hadn't justified the cisco contract yet for firmwares, but maybe I soon have :) I'll check out the dictionary first. There is a new firmware version for this phone. I upgraded about 1 week ago to cmterm_7920.4.0-02-00.bin expecting some more stable functionality. Get it and try out - you should then not have any Euro characters any longer. Best regards, Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_sccp / wiki
Latest cvs update as recent as 5-31-2005 Did i miss anything? I could only find the mayday release from 5-1-2005? Some words in general; I am currently using the chan_sccp on a 7920 Wireless Phone. Usually after hanging up a call, asterisk freezes - reboot necessary. Hardware is a VIA EPIA 5000 board... I don´t know if it´s a hardware problem, i just did not get the impression that the chan_sccp works very fine with the 7920. Comments very welcome. Best regards, Armin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Tuesday, May 31, 2005 3:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Chan_sccp / wiki The chan_sccp page at http://www.voip-info.org/tiki-index.php?page=chan_sccp2 has been updated. See the bottom of the page. Thanks. Comments welcome. -- respectfully, Joseph === -= ** = ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.7 on VIA EPIA 5000
Hi everybody. A few weeks ago I wrote to this list asking about anyone being able to compile asterisk on a via EPIA 5000 board. I received some reply saying like no problem, just change to PROC=i586 in the Makefile. So I did. :-) Now I try to compile and I get to a certain point where I get the following error message: Chan_sip.c:9221: internal compiler error: output_operand: invalid expression as operand Please submit a full bug report. Well, I am running a 9.1 SuSE Professional. Anyone encountered an error message like this before? I´d appreciate any kind of help. Thanks in advance. Armin Lediger ___ 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] Problems with VIA Chipset
Hi, I am trying to install asterisk 1.0.7 on a VIA EPIA 5000 board - anyone of you already managed to do so? I got V1.0.6 running, but 1.0.7 seems not to compile. I don´t want to bother you all with output code of the errors I get when I try to compile asterisk; I am just curious if anyone of you made it! Thanks for a quick reply! Sincerely, Armin Lediger ___ 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: HINT
Yes it does.
Armin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julian J. M.
Sent: Saturday, May 07, 2005 1:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: HINT
But that only works when SIP/201 receives a call, right?
What if SIP/201 is making a dialout call, does it show as
busy in the phone's keypad?
Julian J. M.
On 5/7/05, Thorben Jensen [EMAIL PROTECTED] wrote:
Could you please give us some more detail as to what you did, in
terms of configuring the hint, and specifically what
changes in the
behavior of the running server-phone interaction as a result?
You need to set the hint for the phone when the phone is
being dialed
like
this:
exten = 201,hint,SIP/201
exten = 201,1,macro(dial-sip,201)
It's important that you write the full name of the phone
SIP/201 as
you can't use substitutions like this SIP/${EXTEN} - it
took me a long
time to figure that out.
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[Asterisk-Users] Asterisk 1.0.7 and VIA EPIA
Hi, guys. Anyone of you succeeded in installing 1.0.7 on a Via Epia 5000 board? If yes, what else do I have to modify in the Makefile except setting PROC=i586 to successfully compile *? I am running Suse Linux 9.2 and would not like to switch to another distribution. Any help would be great! Thanks! Armin ___ 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
