similar to: softphone/debug panel with BLF

Hi all, I have a question: how can I see hints of a remote Asterisk in IAX2 trunk?? I want to set BLF on my phones to look state of other phones also in other Asterisk server. Someone have any idea or solution? I use Asterisk 1.4.24. Thanks all Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

I'm looking for some ideas of people who have setup a sidecar (eg: Aastra 560M). Obviously it's handy for BLF (to see who's on a call)...but what else? Anyone want to share interesting things they've done with a sidecar? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am having some troubles with Snom phones and maybe someone can help me. Let me say this: BLF and pickup works great with Polycomes and Grandstream etc... So I think my problem might not be Asterisk related but I am not 100% sure. The snom phones subscribe to my extensions (hint priority) as expected. The light blinks (ringing) or is turned on (in the call) as expected. My problem is to

Hello, I'm just wondering if anyone has fixed the 'InUse&Ringing' problem. * v1.4.23.1 Ta

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List (Again) Thanks to some answers on this list and another I now have a MultiTenant system/setup working the way that I want it to, So now my next job is to find a SIP SoftPhone that I can brand to my own company images and so on. Again an OSS would be preferred, Even though X-Lite is bar far one of the best free SIP clients I have used and it

Hello, all. I am attempting to use IMAP voice mail storage in Asterisk 1.6.1.1 on CentOS 5.3 using Zimbra 5.1.6. I will not be using it as it has proved terribly unstable - Asterisk segfaults on every voice mail message although the message is successfully deliver to my email inbox - but I thought I should report it. Here are the errors from the Asterisk console: -- Executing [210 at

I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the "Outside NIC" Some of the phones are being disconnected with Asterisk

I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? Leif

I want to use JabberSend in my dialplan, but I saw that my Asterisk does not support Jabber. Also I have nowhere a module res_jabber.so... So I thought I'd rebuild my Asterisk. In menuselect I saw that res_jabber was dependent of 'iksemel' and 'gnutls'. In my yum repositories I can find a gnutls.i386, but what is this iksemel-beast ??? There is info to find via google on

Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. Best Regards, -- -- Marc LEURENT lftsy at leurent.eu -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have some trouble with the BLF indicator. I have two phones that use the same hint: 13 => hint,1,SIP/phone13&SIP/phone13-wlan This works great from the asterisk side, but it seems the status change is too quick for the attached Grandstream-phones. When I ring the extension the hint changes to "Ringing". The Grandstream blinks. Great. Now, when someone picks up one of

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

I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial.c.diff It works, but... I can't figure out setting/reading an option. It looks fairly easy, but the flag is always set. *** app_dial.c.org 2009-11-04 22:15:50.000000000 +0100 --- app_dial.c 2009-12-01 09:29:19.000000000 +0100

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than "NOT_INUSE". I have two extensions: 6666 and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use 6666 to call 6668 and in the dialplan have a noop to see what

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

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

Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service....)