similar to: Hoteling

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

I have a GXP2010 phone, the one with 18 blinky lights ;) I currently provision the phone by writing out the conf file, encoding it and sending it to the tftp server. I was wondering if anyone had managed to automate the web side of provisioning ? TIA Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security

Since 1.4.22 realtime status for sip peers seems to be broken. If I do a "sip show peers" from the CLI I get this: 2001/2001 192.168.2.234 D 5060 UNKNOWN Cached RT It is arbitrary which peers will say OK and which will say UNKNOWN and it changes over time. This is a problem with an application like the Asternic Flash panel because it uses the peer

Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra,

Hi, Just wanted to let you know that we have just made our GPL toolkit "Gemeinschaft" available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see

I have a Sipura 962 endpoint on Asterisk 1.4 where the MWI works perfectly, however my theory is that it should be broken. Obviously I'm wrong but "Sip show subscriptions" does not show the endpoint subscribing to the MWI status on Asterisk, even though all of the other endpoints on the system DO subscribe for their respective mailboxes, including SNOM & Polycom endpoints.

I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? TIA!!! Thanks,

1.6 UPGRADE.txt: > * If you use any interface for modifying voicemail aside from the built in > dialplan applications, then the option "pollmailboxes" *must* be set in > voicemail.conf for message waiting indication (MWI) to work properly. This > is because Voicemail notification is now event based instead of polling > based. The channel drivers are no longer

Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090127/0963b5a4/attachment.htm

What is the best way to have 1 phone check multiple voicemail accounts. I am using polycom 650 phones, and am wondering if mwi can work when checking multiple accounts. -Chris Sent from my BlackBerry? wireless handheld

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Just a quick question ---cut--- [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 -

Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!

Hello Asterisk People, I am having trouble connecting asterisk to a huawei SoftX3000 Switch, i can succesfully connect other softphones like Zoiper, but when it comes to Asterisk SIP Client, the system doesn't authenticate, i have the following configuration: peer: 10.220.0.2 username: 4857768 password: 4857768 the configuration is as follows: in the general section: register =>

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

On starting Asterisk (1.4) I get a whole bunch of WARNING[5858]: pbx_ael.c:4040 ast_compile_ael2: Warning: file /etc/asterisk/extensions.ael, line 932-932: Empty Extension! I find it a bit disturbing that this message has a level of WARNING (instead of NOTICE maybe) because the extensions in question are empty on purpose. The only reason they exist are the hints. hint(SIP/3000) 3000 => {}

I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Right now, I am setting up asterisk to use voicemail with my Cisco Call Manager (Which I detest BTW...) and I have everything working, EXCEPT: I cannot get my externnotify script to run when any changes have been made to the VoiceMail... Scenario: Bob gets a call -> Bob