similar to: PickUpChan and Intercept

Paul Madley wrote > >Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 >release, and therefore I don't think any config changes will fix it. >We've been told to roll back to our previous 1.4.13 installation. It >also seems to manifest itself in "ghost ringing" as I've called it; >place a call to a SIP extension, then put down the

Hi All, I've installed Enhanced PickupChan on asterisk 1.2.14 using howto from http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp . from extensions.conf: exten => 0,1,Dial(SIP/eosoiris|20|tTrR) exten => 200,1,Dial(SIP/dzalewski|20|tTrR) exten => _7.,1,Pickup2(${EXTEN:1}) When I try to pickup ringing SIP channel from other IP headset I go disconnected. here is debug from

Hi, I have configured my SNOM 360 to monitor another extension by setting the following: [default] exten => user1,hint,SIP/user1 The next step was to define a function key on the phone as an extension with the value <sip:user1 at 192.168.0.101> and later with <sip:user1 at 192.168.0.101|*8> When someone now calls extension 97 (which is the number of the corresponding phone),

All out of the blue I get these errors? Any Ideas why Please help May 26 09:54:28 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:30 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:33 WARNING[3964]: channel.c:507

Hi folks, I've already searched the mailing list but no one else seems to have my same problem. I'm using Asterisk with the following configuration: Fedora Core 4 (but I also tried Fedora 3) 1 Digium TE110P 1 TDM40B 1 HFC-S 'Cologne' bristuff 0.2.0-RC8o (zaptel 1.0.9.2) I compiled right, I can load kernel modules but when I try to unload wcte11xp module (the one

Hi, > Does anyone of you has a working configuration with SNOM phones that are > able to pickup a call from a flasing LED? Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and therefore I don't think any config changes will fix it. We've been told to roll back to our previous 1.4.13 installation. It also seems to manifest itself in "ghost

Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello.. How is this possible?? I have 65 active calls .. but making new calls and registering isn't possible anymore the CLI command restart now didn't even work .. had to kill the process before it worked again... myasterisk*CLI> show channels Channel (Context Extension Pri ) State Appl. Data 0 active channel(s) 65 active call(s) Jun 27 16:22:06

Customer is having problems with his internet connection, I have in my context: [jimballboutiques] . exten => 1235690251,1,SetGroup(customer) exten => 1235690251,2,CheckGroup(3) exten => 1235690251,3,Dial(SIP/jimball,20,r) exten => 1235690251,4,VoiceMail(u1235690251@jimballboutiques) exten => 1235690251,103,VoiceMail(u1235690251@jimballboutiques) . Now I've had it

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

Ok, but is there a way to check if echo cancellation is active on a call in progress ? Thanks Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Steve Davies Inviato: marted? 28 marzo 2006 16.43 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Echo cancellation

Hello to all of you! I'm very new to this list and to asterisk and stuff at all. To build my asterisk server I installed a new machine running the new SUSE Linux 10.0 (retail version on DVD). I need asterisk (tried 1.0.9), bristuff (off junghanns.net, -0.2.0-RC8o) and the florz-patch because I have two HFC-S-ISDN cards in that machine. Now when it comes to compiling I get a huge bunch of

Hi everybody, I'm running an asterisk box -cvs version since few monthes, updated it middle of may and a last one on thursday (24 june) Since this one, my H323 calls loose they audio, both sides. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab

Hi, I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap :) I was wondering if someone could point me at 3 things that I appear to have "lost"? 1) ZapEC(off) - Is there an equivalent dialplan command to request no EC on a channel before dialling in DAHDI? 2) rxfax(file.tiff) - I have found ReceiveFax(), but I am aware that much has happened in the faxing stakes

Hi All, I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT 600 via MGCP. Got it working really nice but now have a pretty bad problem: 1. When I perform a flash on the telephone, I usually get a second dialtone, but when I dial, dialtone doesn't break. If I flash back and forth a few times, it will eventually give me no dialtone.. here if I dial, it successfully

Trying to compile BRIstuff 0.2.0-RC8o. Ran the download.sh and compile.sh scripts to automate the process. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"1.0.9-BRIstuffed-0.2.0-RC8n\" -DINSTALL_PREFIX=\"\"

>> Same problem with debian sarge on a dell and asterisk 1.0.7 from >> packages, unloading the module freezes the system, (rebooting the >> machine worked right), I installed zaptel 1.2beta and it seems to work, >> but I haven't really tested it, only loaded/unloaded/loaded and placed a >> couple of calls. >Interesting. The zaptel part of the bristuff

I did it Steve, but on some calls i still have the EC on OFF. What can i check? Could it depend of my zapata.conf ? Thanks Giordano Grandis -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Steve Davies Inviato: marted? 28 marzo 2006 17.08 A: Asterisk Users Mailing List - Non-Commercial Discussion

Hi I am getting this error after installing and configuration of asterisk. Aug 24 17:53:50 WARNING[9924]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'Zap/3-1', 10 retries! I have upgraded asterisk to latest version but still receiving the same error. Can someone helpme to resolve this issue. Regards, Shafqat Hamid -------------- next part