similar to: SMS over SIP and Asterisk ??

Hi all: Does somebody know how many meters of cable is supported between Asterisk Digium-FXO cards and the analog telephone ? Thanks, Angel

Hi list, Time to time, my asterisk goes down.Verifying with TOP, I see the swap memory of the computer get used totally but, I don't see what the process is using it. Hereis a copy wath I see doing top. Does somebody have an idea ? My asterisk version is ====>>> Asterisk CVS-HEAD-08/18/04-22:30:24 Thanks Angel. 08:49:19 up 5:23, 1 user, load average: 0.50, 0.70, 0.64 35

Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report "Not Response". I modified the audio level in zapata.conf and after that the fax machine report "Commnunication Error". Do you an idea what could be ? Thanks, Angel. > Message: 3 > Date: Sat, 18 Sep 2004 00:48:23 +1200 > From:

We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5 from Counterpath). As far as we know, Asterisk don't support yet IM (Instante Message) feature,instead Eyebeam have this feature. Is that true? Is there any new version from Asterisk that supports IM? > Eduardo R. Assis > Soluziona Ltda > Consultor S?nior - TELECOM > Al. Tocantins, 125 - 290 andar -

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a

Hi, does anyone know of * being used with icecast in any way. Does * support vorbis at all? can anyone who is working on this give me a shout. ---- Shidan

Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel.

Hi, I just upgraded by 1.0.x home server to 1.2.2. Overall the upgrade went fine, but a strange problem has cropped up with the CALLERID name value of incoming calls from the X101P card. When an incoming call is presented (via Vonage ATA), the calledid value getting double quotes up from: -- Executing NoOp("Zap/1-1", """WIRELESS CALLE" <1404xxxxxxx>") in

Hi List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

I don't understand why so many government sites fail to provide some sort of feed to their daily bulletins. What I am venting about in specific are the Canadian CRTC and FCC sites, every day I have to go to the website and when I reach the content, usually it isn't even HTML but a Word or PDF file. So finally today I decided to do something about it and wrote a little app that scrapes

hi all I spoke to this guy the other day, working with Cisco's VoIP system. He told me they were using a PRI/E1 to transport SMS, and could even do so from their phones. May this be possible with asterisk? I have an E100P in my primary asterisk server connected to a E1/PRI. roy

I have a client requirement that multiple phones can be dialed, however they don't want the pstn phone to pick up automatically because of voicemail etc, nothing can be changed on the phones, how can I handle this requirement, by the way no zap channels are involved, all the pstn phones are behing another sip gateway.

Hello, I'm using Asterisk 1.6.2.0 and I like to use escape characters with SendText, because I can just delete the message from my phone (Thomson Speedtouch ST2030) display by sending a return-char (\n). But \n is not escaped: I tried already: exten => 222, n, SendText(\n) exten => 222, n, SendText("\n") exten => 222, n, SendText('\n') exten => 222, n,

Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf

Hi, I understand SendText() sends text on the current channel. Is there a way to manipulate this feature to SendText toward another SIP device? I use Polycom IP600's. Local sendtext works fine. Would be nice to drop an instant message on another user's phone. thanks! Mark

Hi noticing the cvs updates of late, I'm wondering if there is support for fifo/shell commands in the extended dialplan language? can it fully replace agi scripts? Looks really interesting...........

Someone posted a message suggesting someone try sendtext() and so I thought I'd see if it was useful. Much searching through help at the CLI has failed to find any help for sendtext, but I did find that: "core show function vmcount" fails but: "core show function VMCOUNT" works. Is that a bug and if so, has it been reported? Ira

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Mon, 21 Sep 2015 06:48:52 +0000 > Emil Ohlsson <emo at svep.se> wrote: >> [sip-im] >> exten _X!, 1, NoOp(Got message) >> exten _X!, n, Answer() >> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...) >> exten _X!, n, SendText(Message received) > > I am not