search for: ast2

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 have wildcard extensions that define which PSTN line to use when dialing out.. For example I have the...

[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>

...t SIP config which doesnt seem to work: sip.conf on asterisk1: register=ast1:****@x.x.x.x [100] username=100 type=friend secret=**** record_out=Never record_in=Never qualify=no port=5060 host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Ext 100" <100> [ast2] type=user secret=**** context=local [astrx2] username=ast1 type=peer secret=**** host=x.x.x.x sip.conf on asterisk2: register=ast2:****@y.y.y.y [101] username=101 type=friend secret=**** record_out=Never record_in=Never qualify=no port=5060 nat=never host=dynamic dtmfmode=rfc2833 context=from...

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: "MOVE TO IAX"). Now, while Ast2 shows in the asterisk cdr's a clear BUSY disposition, for a...

...lt;-", x=string) > decode <- function(string)gsub(perl=TRUE, "<<-", "->", x=string) > rightArrow <- as.name("<<-") > leftArrow <- as.name("<-") > ast1 <- parse(text=encode("x1 + x2 -> a3"))[[1]] > ast2 <- parse(text=encode("y4 <- b5 + (b6 / b7)"))[[1]] > identical(ast1[[1]], rightArrow) [1] TRUE > identical(ast2[[1]], leftArrow) [1] TRUE > ast1[[3]] <- as.name("new_a3") > decode(deparse(ast1)) [1] "x1 + x2 -> new_a3" -Bill On Mon, Mar 4, 202...

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

-----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 and after a few seconds, the line is hung up. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIG...

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

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:

...ysql master-slave replication between two svr), 2 x kamailio boxes(failover configured), 1 x file server boxes, 1 x app server , run freepbx & queuemetrics. all 8 server are dell r310. 2. the gateway is one mx100 with 4 E1 lines plugged, the incoming calls go to kamailio2 , and routed to ast1/ast2 in round robin mode. 3. all agent phones registered to kamailio 1, and the extensions are still maintained with freepbx 4.On asterisks, all trunks with destination to pstn or agent phones, go to kamailio1; and incoming calls trunk from kamailio2. 5.admin also use freepbx to configure inbound rou...

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

...ality problems and even with CBWFQ (qos) enabled, it doesn't make a lick of difference. jitterbuffer (no trunk of course) is also enabled. I'm trying to keep this email short so I won't go into detail. Thanks! ascii layout: (ast1) -- ATM -- sonicwall -- ethernet -- (ast termination) (ast2) -- ATM -- ATM (don't ask) -- sonicw -- ether -- (ast termination) Matt

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 between the two servers where disconnected (the two phones are talking peer to peer thanks to the ca...

...going 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 the SIP client. SIP client --- Ast1 --- IAX2 --- Ast2 --- Zap --- PRI - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAj+rc2TEAILET3McRAi79AJ4tF08y9UkdXZPG9/YqjVoXWMBzhQCfaCE1 JBMkMM43jX3ZEtusH95HJrc= =C/wg -----END PGP SIGNATURE-----

...chan_unicall.c: MFC/R2 UniCall/8 Answer guard expired Mar 31 13:10:51 NOTICE[14902] chan_unicall.c: Unicall/8 event Accepted Mar 31 13:10:51 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 Channel gains Mar 31 13:10:51 VERBOSE[31370] logger.c: -- Executing Dial("UniCall/8-1", "IAX2/ast2:iaxpalancar at 192.168.1.20/1||TtWw") in new stack . . . .....everything continues normally Thanks in advice for your time. Ivan Reyes Tejera -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/200803...

Hi, I'm wondering what the best option to obtain a high availability asterisk server is. I currently use a TE410P (4 x E1) card. I'm thinking of 2 different solutions: - 2 servers configured with Heartbeat + DRBD (drbd mainly for voicemail....) and the E1 span plugged to the 2 servers (with a TE410P in each server). - 2 servers configures with Heartbeat + DRBD with the E1 span hooked