similar to: Fwd: blf problems after dialplan reload

Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [root at pbx01 asterisk-1.6.1.1]# grep -i speex

Hi there, I am running Asterisk 1.2. I have a Grandstream GXP2000 and Aastra 9133i with BLF/Speedial configured for other extensions. The hint's are all configured in extensions.conf and it seems to work as it is supposed to until I reload the configuration in Asterisk or reboot the server. Then neither phone displays the BLF of other extensions until I reboot the phone. It continues

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show

Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel! > Using chan_sip you need to create another ?user aand then dial both > > Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk... :( I'll create another user. Thanks Luca Bertoncello (lucabert at

Israel Gottlieb <isrlgb at gmail.com> schrieb: > At the end of the Command you could use options one of them is the c (not > apital) which sends a cancel event to the phone > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote: exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,Rc)

Hi Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expected Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/7e14a4e0/attachment.html>

I tried 1.0.1.12, 1.0.1.13 and now 1.0.2.13 on the GXP2000. I doubt it's the firmware if it is happening on two completely different phones. I read of a bug in Asterisk that exhibited this exact behaviour that was apparently fixed in Asterisk 1.2.0 as far as I can tell. http://bugs.digium.com/view.php?id=3644 > -----Original Message----- > From: stoffell [mailto:stoffell@gmail.com]

Hi all Is there anyway i could get in the dialplan the amount of time a caller waited in the queue before exiting? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160518/b3b082aa/attachment.html>

Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 running Linux on 2006-06-17 When I used monitor, I seem to get most calls cut off if they run very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any ideas what might kill the recording process? I'm beginning to wonder if soxmix is truncating the file when it blends the in/outbound streams together

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

I recently began using the curl cmd to do an external callerid lookup on my own customer database. I've noticed certain lookups will cause a crash and not show anything in the messages file or the console. The curl command is connecting to an external webserver which has a oracle db connection. The file its hitting is PHP and does a very simply lookup showing the text like "C1234 Bobs

Hello, I have 2 asterisk servers that are not working well together. One is acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX devices. And the other is acting like my sip gateway (PBX02) to various providers. They are both on a private network and should be trusting each others IP 100%. But the PBX02 challenges PBX01's requests all the time even though

Hi guys i upgraded to asterisk 16.10 and in most recordings you here only leg A in the recording sometimes you might hear a word of leg B Did any body hit this problem? Thanks, israel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200512/90ee8dc2/attachment.html>

Is there anyway to "spoof" a BLF signal to a phone from within a dialplan? That being so I can turn a BLF light on or off on a phone when a certain thing happens in a dialplan?

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

Hi, I used Grandstream (gxp2000, gxp2020) and Snom (370) SIP Phones, but with 2 extensions BLF status does not work correctly. have someone ever tested a Sip Phone with more then 60 BLF without problems? Can someone suggest me model and brand? Thanks, bye. -------------- next part -------------- An HTML attachment was scrubbed... URL:

has anyone done it? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080528/ff622d28/attachment-0001.htm

I'm trying to get the BLF to work correctly on my Polycom phones. I have the buddy watch working correctly, but can't get the BLF to change based on the state... Example: When an extension is ringing, I get the same 'red light' that I get when the extension is actually in use... I was wondering if anyone had any experience with getting the Polycom phone to