similar to: Cisco SPA504G, transfer asterisk page()

Hello, I am struggling to have a SPA504G to auto answer (for intercom/paging). I have tried the following SIP headers (not all together), but without luck: SIPAddHeader(Call-Info:\;answer-after=0); SIPAddHeader(Call-Info: answer-after=0); SIPAddHeader(Alert-Info: info=intercom); SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(P-Auto-Answer: normal); Any other ideas? Leandro PS I have set

We have a queue monitoring application running so we can see the caller ID of callers in a queue. If we see a VIP in the queue, is there any method to force that call to be first in line? If there's a softphone, or queue managing application already written that does this, I'd love to know.

When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more

If you have experience with these phones... We are trying to figure out how to transfer an established call on the SPA504G while a second call is incoming. At present, the receptionist has to answer every single incoming call before the XFER softkey is seen again. This is completely unpractical for a receptionist that may have 4 or more calls coming in at the same time. When the

Hi!I've a problem using ssh without password: I want use rsync for automatic scripts,I'm using this 2 names for my asterisk@home2.5 linux (based on red hat), rsync11 and rsync12. This is the way I use to change the configuration and then using without password , but the password is always asked: [rsync11@asterisk11]$ ssh-keygen -t rsa Generating public/private rsa key pair. Enter file

Hi! I' ve this problem: I've 2 asterisks box, asterisk11 and asterisk12, and one wi_fi phone. I call from wi_fi to a X-lite phone on a windows xp.I've setuped the X-lite to my vrrp IP (vrid IP) and the call is ok, I call from the wi_fi to X-lite and from the X-lite to wi_fi. In asterisk panell is all ok, and I listen the voice to the xp and in the wi_fi phone. asterisk12 is my master.

I was asked today if we could somehow have a trainee on the phone with a supervisor conferenced in, but somehow have it so anything the supervisor says is only heard by the trainee and not the customer. Is there a feature like that?

I'm asking about this scenario: Asterisk(public IP) <--> Internet <--> Router (public IP) <--> SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia

someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

can i test my asterisk11 on a single machine on which asterisk is installed that sounds are working from both end properly. i have installed asterisk 11 on ubuntu12.04 with twinkle soft phone. regards abhi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130216/40fa976d/attachment.htm>

Hi, all I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc.

Hi, all Sorry that forgot add mail subject last one. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database

Hi, all Sorry for null subject last mail. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database

Am 03.03.2015 um 18:16 schrieb James B. Byrne: > CentOS-6.5 (FreePBX-2.6) > Asterisk-11.14.2 (FreePBX) > snom870-SIP 8.7.3.25.5 > > I am having a very difficult time attempting to get TLS and SRTP > working with Asterisk and anything else. At the moment I am trying to > get TLS functioning with our Snom870 desk-sets. And I am not having > much luck. > > Since this

Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip

WORLDWIDE EMERGENCY The code below needs to be executed before any SIP or PJSIP call destined to the US network, or soon no call will terminate. This is called Stir-Shaken, a new law from the FCC. If this is not working the whole Asterisk industry will crash, vanish, be gone. I am assuming that the caller ID and the Destination Number are in the variables "${CALLERID(num):-10}"

I have a customer that has a campground. Wants to see who's at the gate, remotely, via camera, and talk to that person through a "traditional squawk box" and be able to open the gate remotely from that phone. He doesn't want to have a separate camera feed, etc, he wants to do it all on one phone. Does such a way to do this exist by using Asterisk and some kind of relay