similar to: Weird SIP behaviour

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, It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem.. My setup.. UA1--[AST1]--{IAX}--[AST2]--UA2 | | PSTN1 PSTN2 I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent.. I

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

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

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

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

[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>

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:

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

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...

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 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

Does anyone have any experience using asterisk with an external predicitve dialer, like MediaTel? Specifically: The predictive dialer dials out over T1 circuits. It connects to asterisk via amphenol cable from an fxs card in the dialer to asterisk with a tdm2406 fxo card. In the analog world, the dialer dials out through the t1 circuit, and the fxs card is plugged into a 66 block so the

Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows "bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

I have been impressed with the quality and meagre bandwidth of the G729 codec from Digium. I am in a testing phase of our roll out, we are using 5 Asterisk PBXs in various countries to provide connectivity for our employees, owners and family. As we are testing, and our setup is somewhat complex due to the peculiarities of our connectivity, there has had to be a lot of changes to servers, cards to

Hello, I am replacing an exisiting call center with a new asterisk based solution. This will initially consist of to phone servers. The first being the main PBX, and the second being a predictive dialer. The dialer will have sip extensions for all the agents, while the main pbx will hand pretty much everything else. The two boxes will be right next two each other, and are currently

Hi Guys I'm configuring a TE212P card and have the following two entries in my /etc/asterisk/zapata.conf switchtype=dms100 signalling=pri_cpe When I reload asterisk I get the following messages: > -- Reloading module 'chan_zap.so' (Zapata Telephony) > == Parsing '/etc/asterisk/zapata.conf': Found > [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072