similar to: switch => priority in the dialplan.. (probably an issue for Mark)

Maybe someone has already suggested this, but if your functions accepted strings you could use sub or gsub to replace the -> with a symbol that parsed at the same precedence as <-, say <<-. Then parse it and deal with it. When it is time to display the parsed and perhaps manipulated formulae to the user, deparse it and do the reverse replacement. > encode <-

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing happens

Hi everybody: I configured my Asterisk to register to my VoIP provider, and I can make outgoing calls, but I can't receive any calls with it. I used Ethereal to sniff the activity of it, and I found something that might be causing the problem: When my provider's gateway does the "Request: INVITE mynumber@my-voip-provider.tld ..." my Asterisk asks for "Status: 407 Proxy

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

Hi all. Ive recently setup two Asterisk boxes (running Asterisk@Home to be specific), and Im trying to get a trunk going between them. So far I have tried a combination of IAX and SIP configuring them through AMP and writing the config files manually, but I cant seem to get calls going between the two. I have named each box asterisk1 and asterisk2. Does anyone have some working SIP and/or IAX

Hi All, I've been observing a very odd behaviour of Asterisk, when relating to SIP connections. Here's the scenario: Ast1 is an Asterisk box originating calls via a predictive dialer Ast2 is an Asterisk box connected to 3E1 circuits Ast1 originates calls to Ast2 via SIP, in order to utilize the PSTN lines. (There is a reason I'm using SIP here, so please don't say:

Dear Barry, In general, I believe users are already accustomed with the classical arrows "->" and "<-" which are used as such in quoted expressions. But I agree that "-.>" is a very neat trick, thanks a lot. A small dot, what a difference. All the best, Dmitri On Mon, Mar 4, 2024 at 11:40?AM Barry Rowlingson < b.rowlingson at lancaster.ac.uk> wrote:

Dear All, I was trying to use Realtime Asterisk option for Sip users and peers + Heartbeat + Mysql Replication in order to make a failover system, so that if Ast1 went down for any reason, Ast2 server will have the same data that Ast1 used in the Mysql database and don't need to make the phones re-register. But when I started testing: the calls that where active during the transition

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from

As I understand it (and my understanding is obviously incorrect) the switch => statement sells the Asterisk box to resolve (aka lookup) extensions by querying the remote Asterisk server defined in the switch => statement. The switch => statement is used to centralize dialplans. I've not used the switch => statement yet, I'm just trying to understand the ramifications of using

Hi Pan & Dhaval, In the past 8 weeks, we have delivered a load-balanced asterisks (1.4) based call center with our flexqueue application for icson.com. It has the below features, 1. 2 x asterisk 1.4 boxes, 1 x mysql db box and 1 x flexqueue box(the two are failover configured with heartbeat and custom script, and mysql master-slave replication between two svr), 2 x kamailio boxes(failover

Hi all, I'd like to setup a really lean Asterisk installation that essentially has a full ISDN PRI (AT&T, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan) on a Digium TE207P adapter that all it does is convert the ISDN channels to SIP/IAX channels. Then I would add this Asterisk 'gateway' as a provider on one (or many) Asterisk systems on the back. With such a config I

Here's a good one for the group, I have 2 Ast servers behind a NAT (Sonicwall :-( ) connecting to the same server outside the NAT. Each of the 2 boxes behind register to the outside server. What I am wondering is, would there be a problem if both servers behind the NAT were listening on port 4569, I realized that the NAT'd port gets changed however I wasn't sure if this would be an

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi all, I just got a Mediatrix 1104 evaluation unit -- a 4 port fxs sip gateway. There's no printed documentation shipped with the unit, but I have a piece of software for windows that shipped with a different model (which I haven't tried configuring yet), that uses snmp to set misc variables (ip settings, sip stuff, etc.). Fairly baroque interface & pretty slim on help...

I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward. Example: PSTN1 Calls * dials PSTN2 if PSTN2 presses proper digits bridge the PSTN1 and PSTN2 if no response return to a context Reasons: 2 actually 1. call is forwarded to cell phone but If cell is out of range, turned off,

Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register

This is not really a problem just something I noticed in my testing.. When two or more Asterisk servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2

Is there a way to get asterisk to send you a email when it looses or an extension doesn?t re-register Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 Website: http://www.upperclassman.net Billing Questions: billing at upperclassman.net Rental Questions: rentals at upperclassman.net Maintenance: help at