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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 exa...

[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>

-----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...

...e <- function(string)gsub(perl=TRUE, "->", "<<-", 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(...

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 As...

...fig 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 trunk configurations they can send to me? Here is my current 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 se...

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 between the two servers where disconnected (the two phones are talkin...

...b1892aebc2b87f295187ebbea@123.45.67.89 CSeq: 1117525281 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:3101@123.45.67.89> Proxy-Authenticate: Digest realm="asterisk", nonce="23e26a38" Content-Length: 0 to 98.76.54.32:5060 ast1*CLI> Sip read: REGISTER sip:123.45.67.89 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457 Content-Length: 0 To: Port 3 <sip:3102@123.45.67.89> From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REG...

...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 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+rc2TEAILET3McRAi79...

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, i'm evaluating performance of centos7 i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 with 500calls (7sec alaw, simple dialplan, pass trough - sipp generators/asterisk receiver with answer/playback) scenario - sipp generators - asterisk - asterisk receiver (i wrote ansible scenario for this if you are interested) then i reinstalled system to centos7 x86_64/distro

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

...and mysql 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 inboun...

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

...issue. The reason I ask is we're getting a lot of call quality 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

...ng '/etc/asterisk/zapata.conf': Found > [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072 process_zap: > Ignoring switchtype > [Mar 30 17:48:42] WARNING[2985]: chan_zap.c:11072 process_zap: > Ignoring signalling However pri show span 1 shows the right values set for both: > ast1*CLI> pri show span 1 > Primary D-channel: 24 > Status: Provisioned, In Alarm, Down, Active > Switchtype: Nortel DMS100 > Type: CPE > Window Length: 0/7 > Sentrej: 0 > SolicitFbit: 0 > Retrans: 0 > Busy: 0 > Overlap Dial: 0 > T200 Timer: 1000 > T203 Timer: 1000...

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

...> app_dbodbc: Query Successful! -- odbcget: Value not found in database. -- Executing Goto("Zap/23-1", "s|5") in new stack -- Goto (macro-sipexten,s,5) -- Executing Dial("Zap/23-1", "SIP/9161111111|20") in new stack Segmentation fault smf-ast1:~# Ouch ... error while writing audio data: : Broken pipe As I said, if I change these back to DBxxx, and noload app_dbodbc.so, no problems at all. 100% repeatable. When I look in the astdb mysql table, all the data that was ODBCput is there. Thoughts?

...# After the dest IP has been put in the message the servers will follow it on their own. if(method == "INVITE") { # If it came from Asterisk send it to a Quintum for final routing if it can't be looked up # As of now its ast0 and ast1 which are 222.222.222.222 and 333.333.333.333 which fall within this subnet if(src_ip == 192.168.0.145) { xlog("L_INFO", "%rm came from Asterisk server %is, URI = %ru"); # native SIP destinati...