search for: asteriskpbx

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware pci:0000:03:08.0 wct4xxp+ d161:0220 Wildcard TE220 (4th Gen) astpbx ~ # But dahdi_tool does not show the card and dahdi_cfg gives DAH...

...g ?Asterisk. The Definite guide? book as a reference. I connect and work using SSH Problem I have - I can?t connect to asterisk from remote. Getting error: $ sudo asterisk -rvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Yes, it exist, and service runs: [asteriskpbx at localhost asterisk]$ service --status-all asterisk.service - LSB: Asterisk PBX Loaded: loaded (/etc/rc.d/init.d/asterisk) Active: active (running) since Sun 2015-03-22 21:04:22 CDT; 23min ago Process: 1615 ExecStart=/etc/rc.d/init.d/asterisk start (code=exited, status=0/SUCCESS) Main PI...

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip communicatio...

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

...lls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228 MaxRetries: 0 Dial(Skype/asteriskpbx at somebodyonskype) Priority: 1 Callerid: Somebodyonskype <somebodyonskype> --- SIP/228 is my desk phone, i purposely did not include a context (is it necessary for Skype calls? i guess not because i also did not create a 'dial plan' for it, should be just dump and go i guess) But...

...Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP provider for a PSTN connect. My Ekiga test calls are successful. So it does seem to work :) The router is configured like this: - has a user 'asteriskpbx' so Asterisk doesn't run as root - has a USB stick for logs and CDRs - has a LAN port which I removed from the regular LAN VLAN and put into its own VLAN; only the hardware SIP phone is connected; there's no forwarding to and from this network - firewall allows SIP and RTP packets f...

Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _____________________ Don't PANIC

...sccp is correctly installed and running. I have scoured the internet looking for documentation regarding the 7914 call modules with asterisk and turned up nothing. So this is my question.... Does anyone have any kind of documentation regarding 7960 phones with 7914 expansion modules running on AsteriskPBX??? I also have the proper load file for the expansion module... Any Help would be greatly appreciated, Thanks, Jesse Tyler

Anyone in New Zealand using AsteriskPBX? If so, what hardware are you using to connection to Telecom's lines? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030619/d5c12ca0/attachment.htm

Hi, In digium website. http://store.yahoo.com/asteriskpbx/newitastdmde.html It is said Dev Kit PCI card works with 2.4 kernel. I am wondering if it is also working with 2.6 kernel? Anyone knows? Thanks Michael

...digium twitter 'user' to follow that only tweets important announcements and release information? If there is not, I think there should be. It would be highly utilitarian to get an SMS when there is an update to Asterisk, Dahdi, ADA etc, but I don't want to be bothered real-time with asteriskpbx tweets like: "Anyone trying anything cool with Asterisk over the weekend?". There's nothing wrong with tweets like that, but that kind of 'incidental' communication should really be in a venue of its own. I think that if announcements and release info were made in a dedica...

...meetme application work, that said that it was better to have a real hardware then the zaptelrtc software modules. So, my question is, would any of the following cards work with the meetme application? http://www.goods2world.com/product_info.php?cPath=25&products_id=53 http://store.yahoo.com/asteriskpbx/wildcardx100p.html http://www.digitnetworks.com/store/product_info.php?products_id=28&osCsid=a77255f932eb7a247387647e1da6dbb3 http://www.digium.com/index.php?menu=wildcard_x100p http://69.212.249.116/product_info.php?products_id=30&osCsid=d4225af643f710ece118ad93a64347f9 thanks in advance,...

...6.12-12/Makefile "EXTRAVERSION" it then installs to "/lib/modules/2.6.12-12mdksmp/misc/". When I run "modprobe zaptel" I get "FATAL: Error inserting zaptel (/lib/modules/2.6.12-12mdksmp/misc/zaptel.ko): Invalid module format" uname -a reveals; "[root@asteriskpbx zaptel-1.2.3]# uname -a Linux asteriskpbx 2.6.12-12mdksmp #1 SMP Fri Sep 9 17:20:34 CEST 2005 x86_64 Dual Core AMD Opteron(tm) Processor 270 unknown GNU/Linux" Both the kernel package and the kernel source package are exactly the same version "2.6.12-12mdksmp and 2.6.12-12mdk" res...

...use with nbcluster for my clustering purposes), Postgresql with Slony-II, PGcluster, etc. Or is it already possible? There has been some talk on this before: http://lists.digium.com/pipermail/asterisk-dev/2004-December/007846.html Also, the func_odbc feature seems to be very powerful: http://www.asteriskpbx.org/func_odbc but: 1) would there be potential issues with db handles on a very busy asterisk system after a relatively long run time? 2) would there be a way to "map" the odbc function(s) to the DB functions (Set(DB), read and write, DBdel, etc) so that rewriting the whole dialplan would...

...gt; Or is this normal? Must Twilio respond with a 407 which will cause Asterisk > to authenticate? > > Twilio has a nice technical document to setup a trunk with PJSIP. It includes an example for a pjsip_wizard.conf https://assets.cdn.prod.twilio.com/documents/TwilioElasticSIPTrunking-AsteriskPBX-Configuration-Guide-Version2-1-FINAL-09012018.pdf Maybe that helps. And make sure for your outgoing calls to set the callerid to a valid caller Id which ist authorized with your twilio account. It will not allow outgoing calls if the number is not recognized by twilio -H -- Henning Follmann...

The hardware we are planning to use is: Micronet SP5050 FXO Gateway http://www.micronet.com.tw/Products/VoIP/SP5050.asp Micronet SP5100 IP Phone http://www.micronet.com.tw/Products/VoIP/SP5100.asp We are hoping to use this hardware along with AsteriskPBX to replace our aging PBX system. What I want to acheive is: * Any incoming call from PSTN (via gateway) rings on the receptionists phone for 10 seconds, and then starts ringing on all phones. * Any user can just dial 1 to get the first avaliable outgoing line onto the PSTN. * All users can have vo...

...do not. I'm running CVS-HEAD-07/13/04-21:34:00. Here's the definition of my SER proxy in sip.conf: [ser-to-tollfree] type=peer insecure=yes host=128.151.224.35 context=from-proxy1 secret=cracksmokingpassword In the first two examples I show below, things work great. The first query is (AsteriskPBX -> SER -> AsteriskPRI) and seems to work fine; my calls go through, no problem. The second query is (MTA -> SER -> AsteriskPRI) and also works fine - no problems. Note the address of record - it's "JohnTodd" in the second working example. The third example is from the e...

...t; Or is this normal? Must Twilio respond with a 407 which will cause > Asterisk to authenticate? > > Twilio has a nice technical document to setup a trunk with PJSIP. It includes an example for a pjsip_wizard.conf https://assets.cdn.prod.twilio.com/documents/TwilioElasticSIPTrunking-AsteriskPBX-Configuration-Guide-Version2-1-FINAL-09012018.pdf Maybe that helps. And make sure for your outgoing calls to set the callerid to a valid caller Id which ist authorized with your twilio account. It will not allow outgoing calls if the number is not recognized by twilio -H -- Henning Follmann...

...e shure that srv records in _msdcs.mysambadomain.local are resolving when virtual dc is down) and smbd fileserver with "winbind offline logon = true" in smb.conf. it works on HP Microserver G7 with AMD Turion II CPU and 2GB RAM and serves 15 clients in small office + it's also hosts asteriskPBX guest... All issues that I mentioned are true when there is only one dc in network and it runs as kvm guest. If it's not your case and you have other dc online when fileserver starts. Consider your virtual DC as phisical machine and everything will be fine... On 26.12.2016 21:06, Miguel Medal...