similar to: Assigning an extension to a roaming phone

Hi list, I have searched through asterisk command lines but haven't found how to do this: - can I list the phones (callerid or IMSIs?) currently registered ? If I do "dialplan show" that lists the configuration I loaded, e.g [ Context 'sip-local' created by 'pbx_config' ] '2102' => 1. Macro(dialSIP|IMSI1) [pbx_config] '2103'

Dear all i'm creating an outgoing call to number xxx with this command: http://host:port/mxml?action=Originate&Channel=Local/xxx at to-external &Exten=testDTMF&Context=cRETEUNICA&Priority=1 wich points correctly to this portion of dialplan: [cRETEUNICA] exten => testDTMF,1,Answer exten => testDTMF,n,Read(digito,,1) exten => testDTMF,n,SayDigits(${digito}) The

All: I am looking to install another asterisk server in an office located in a different part of the country. I think I can configure the sip and extension conf files, so that the internal phones at the two locations can call each other. My question is this, how do I properly configure the sip file for a different provider at the new location? Can I use a different register statement for

I'd be really happy if users could use the voicemail menu to change the number of rings until voicemail picks up. It seems like the current model of separate Dial and Voicemail commands would make that difficult, but is there any plan for such a feature in the future? How about a workaround or 3rd party add on? I store the dial timeout for each user in a database, so I know I could make

Hi all, I would like to add a new partition to my ZFS pool but it looks like it''s more stricky than expected. The layout of my disk is the following: - first partition for Windows. I want to keep it. (no formatting !) - second partition for OpenSolaris.This is where I have all the Solaris slices (c0d0s0 etc). I have a single ZFS pool. OpenSolaris boots on ZFS. - third partition: a FAT

Hello, I have setup the n-way conferencing with Asterisk and it's working when I use with my budgetone 100 phone but it doesn't work for any of the voip software or other ATA that I have. When I turned the debug on, I see that the correct keys (*0) were entered but asterisk doesn't detect the signal to trigger the features event. I have set a test extension to get the input dtmf key

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

Hi to all if someone of you is interested on it, i've changed the code of app_asr.c With these patch you can use the ASR application to play DTMF tones, so you can have your own AGI application that uses the ASR and manages the DTMF tones without change the dialplan. EXAMPLE exten => 003,1,Ringing exten => 003,2,Wait(3) exten => 003,3,Answer exten =>

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence

Hi all, I am new to Asterik. Ia m trying to compile the source with the latest asterisk-1.6.2.0 these are the issues I am getting. initially,I got mkdir: cannot create directory `/var/lib/asterisk' than after reading the archives: I did: ./configure --enable-dev-mode --prefix=/tmp/asterisk --sysconfdir=/tmp/astconf --localstatedir=/tmp/aststate and than make install.: This is the error I

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

Can someone offer a hint on what I'm doing wrong with the basic * config? Just implemented * for the first time using yesterday's cvs. The initial configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956, and using two 7960's for initial testing. When one 7960 calls the other, I get the following and the call is dropped: -- Executing

Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 <sip:IMSI20830061xxxx at 127.0.0.1>' failed for '127.0.0.1' - No matching

I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03

Hi I had installed and configured 2 IAX server , users from 1'st server can dial to the second server and vice versa But when I make calls to users in other server , on my client , I get the caller if as asterisk@192.168.20.99 , the same I get when I try reverse , ie I get on my cleint caller id as astersik@192.168.20.32 Please guide me what

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

Hi Guys, I Have this peers on my sip.conf [provider-302333-3000] type=friend context=provider secret=xpto username=3023333000 host=sip.provider.com fromuser=3023333000 insecure=very canreinvite=no [provider-302222-3001] type=friend context=provider secret=xpto username=3022223001 host=sip.provider.com fromuser=3022223001 insecure=very canreinvite=no I Have in my sip.conf two extension 3000