similar to: DTMF

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 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

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

Hi, I've brought this up in the past and there was a good discussion - am wondering if there have been any new developments. Our dialtone service, like I am sure is true for most ITSPs, touts the ability to drop your POTs lines for significant savings. For businesses we have a low-cost Atom based PBX and a "fax relay" setup locally with hylafax/iaxmodem to solve that issue,

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

OK I have just gone live with asterisk in a new office with approx 40 Polycom 501 handsets. I have a few questions: 1) Call Parking: I am able to park calls using the standard Asterisk call parking system (transfer to ext *70 etc...) I would like to make this a little easier for my users The Polycom 501s w/ SIP 1.5.2 seem to support some type of standard call parking, however I don't

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

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

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:

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural "monopoly". From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them,

Has anyone here used T1s with RBS with asterisk? Cary Fitch

Double , Triple and sometime 5 calls<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD90PTc1MzQmc3RhcnQ9MCZwb3N0ZGF5cz0wJnBvc3RvcmRlcj1hc2MmaGlnaGxpZ2h0PQ%3D%3D&b=2> Many time we face an issue where even if an agent is on Call, another call comes in. Sometimes, even if agent hang up the call, call stays back and another come sin and

Hi All, When I want use Asterisk as a PBX to cooperate SIP ITSP, I can not set the caller ID, so SIP ITSP do not accept the call. In Asterisk, I set a account in sip.conf to register on ITSP SIP Server: register => 6292@218.1.121.237/6292 And I added a user 6292 in Asterisk just like the account on ITSP SIP Server: [6291] type=friend username=6291 callerid=6291 host=dynamic

Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look

On 03/24/2015 04:28 PM, Richard Mudgett wrote: > > > On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <jeff at jeff.net > <mailto:jeff at jeff.net>> wrote: > > > Hello, > > I am wondering if asterisk does anything at all to RTP packets > passed from channel to channel if no transcoding is involved? Can > I assume that the packet that