similar to: Stable-Stable Asterisk

Hi again, folks. I've been getting feedback from this list and elsewhere that softphones are generally not considered good enough for hardcore business use. Can someone point me to where I can find more detail on this debate? Is the problem that the technology isn't mature, that the load on the computer is too high, or simply that it doesn't work well in a poorly designed

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

Hi, Is there a list of all the extension states as sent by the manager interface? (I know I could look them up in the source but that involves some "backtracing".) The ones I know are: -1: no hint for the extension 0: registered && idle 1: busy 4: unreachable, not registered 8: ringing I've recently seen 16 (== hold?) but can't find that value documented anywhere.

If it is posible for a imcoming call to ring both the Polycom desk phone and my cell phone at the same time, if I dont answer fall back to my voice mail box. I would like to hire someone to cofigure that for me. Bob -- We've Got Your Name at http://www.mail.com ! Get a FREE E-mail Account Today - Choose From 100+ Domains -------------- next part -------------- An HTML attachment

Hi, folks. I am planning on implementing Asterisk in 2006, and need to budget for it now, so I need to know what I'll need to get. My company has about 50 users, and is currently languishing on a very old Comdial PBX. All of our client computers are Macs; our servers are mostly OS X, with a couple Debians and a Red Hat. I am thoroughly experienced at systems administration, and can

Awhile back I had heard some talk, in this list I believe that Agent callback login was going to be deprecated in 1.4, I see it is still there. Does anyone know what is happening with this? -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 voip at rockynet.com

Hi all! I have an issue with TDM400P FXO card. When a call enter into my IVR and select the proper option, the person that ansswer the call say your "thanks for contact us ..." but the caller cant hear this words because a delay between asterisk and caller part or between asterisk and the ATA device. What is the item on zapata.conf that can affect this delays. Thanks for any help

Would like wish to ALL a Marry Christmas and a happy new year, full of peace, love, happinesses and much success. That let us have one excellent year of 2008. Best Regards Josue Conti.

Hello, A user who uses my Asterisk made me part of a worry about listening to his voicemails. He has received 4 voicemails on January 3, respectively at 3H00 pm, 3H36 pm, 3H41 pm and 4H40 pm. He has received notifications by e-mail at these times. On first listen to his messages, at 8.00 pm, Asterisk has announced two new voicemails(15H00 and 15H36). He has erased thos voicemails. At 8.30pm ,

Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal "*" followed by

Hi, I have other information. I analyzed the core dump: (gdb) core 22534 [New LWP 22534] Reading symbols from /usr/libexec/dovecot/doveadm-server...Reading symbols from /usr/lib/debug/usr/libexec/dovecot/doveadm-server.debug...done. done. [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by

Has anyone seen this before? These phones are behind an edgewater. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bKaf87f1c9f;received=xxx.xxx.xxx.xxx. From: 7408 <sip:5D03C49B-7408@dpl.voip.rockynet.com>;tag=23943befc9dc103 To: 7408 <sip:5D03C49B-7408@dpl.voip.rockynet.com>;tag=as2c0b7dcd Call-ID: 723559d029d27c820c8dae4b01e45c77@192.168.50.31 This phone is

Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general]

I am experiencing intermittent network failures on rsync 3.0.7 built using cygwin for Windows-XP (SP2). I am using GCC v4.4.2 and the latext version of cygwin. The rsync error long indicates things like: rsync: writefd_unbuffered failed to write 4092 bytes to socket [generator]: Connection reset by peer (104)rsync: read error: Connection reset by peer (104) rsync error: error in rsync

Hello everyone. I know it's a little bit off-topic, but I was just wondering... Has anyone ever had any experience with asterisk, a wi-fi meshed lan (with more than one access point) and wi-fi sip phones? I made some tests but I'm not really satisfied.... Wi-fi phones are a curse (as far as I know even Nokia eSeries -I personally own an e70 model- have their flaws): - random sip

Hi Stephan, thanks for your replay. This is bt full output: (gdb) bt full #0 clear (this=0x0) at /usr/src/debug/clucene-core-2.3.3.4/src/core/CLucene/util/VoidMap.h:123 No locals. #1 lucene::search::FieldSortedHitQueue::_shutdown () at /usr/src/debug/clucene-core-2.3.3.4/src/core/CLucene/search/FieldSortedHitQueue.cpp:60 No locals. #2 0x00007f7dc5eabe96 in _lucene_shutdown () at

>But currently, I only have one ethernet jack per office. Routing >another 60 or so ports would add a very substantial expense in both >cabling and backbone expansion (what category ethernet is required, >BTW?). Most decent phones have an ethernet passthrough (2 port) so you can plug in your PC. As long as your LAN is decent (Cat5 100baseT switched) the overhead using VoIP is

Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded

Hi, i have a problem with libclucene-core.so.2.3.3.4. I See segfault error to messages log, when dovecot store mail to INBOX. set 12 13:41:21 box02.par-tec.it kernel: doveadm-server[20285]: segfault at 38 ip 00007f575b68fc42 sp 00007ffc0272e750 error 4 in libclucene-core.so.2.3.3.4[7f575b569000+188000] set 12 13:41:22 box02.par-tec.it kernel: doveadm-server[18871]: segfault at 38 ip