similar to: Are busy and congestion behaving differently than documented?

What we would like to see happen or emulated is that if someone makes a call via our SIP provider to a PSTN number that is actually busy that we get an actual BUSY tone at the telephone. In our test case this is a PAP2-NA SIP device It would appear that when we call the far end (PSTN phone number) that is busy we do not get any busy indication at the user end (originating telephone on our

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

I read the wiki and looked at the examples, but I'm still having problems. I have a Digium 4 port card with POTS lines plugged into all four ports. How do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten

I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-9999) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message "all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any

Hello all. I'm still investigating the cause of freezes on my asterisk server. It's a minimal installation: the only things I remember running are httpd, sshd, sendmail and asterisk itself. I have a DID from Voicepulse. No telephony cards or SIP phones ... I'm just trying to figure out the voicemail issues at this point. So a call comes in, and the caller can type a voicemail number

i tried using music on hold option in the dial command exten => 7777,1,Dial(zap/1/7777,20,m) when someone calls me and i picked up the phone, the call will be suddenly dropped. however, if i use a sip client instead of zap (also changing the dial statement to sip), i can answer the incoming call without a problem. is this a known bug? (asterisk cvs 05-03-04 using RedHat v9 on Via mini-ITX)

-----Original Message----- From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Walter H. Sent: Monday, November 13, 2017 4:32 AM To: centos at centos.org Subject: [CentOS] Strrange behavior of VirtualHosts in Apache (CentOS6) > Hello, > > there is a short explanation about virtual hosts in Apache ... > https://wiki.centos.org/TipsAndTricks/ApacheVhostDefault That page

Hello, I'm using Asterisk with an ISDN30e PRI line (only 16 channels active). Every now and then I get a CONGESTION error even-though there are only 2 channels in use out of the 16. When this happens, the user just needs to re-dial and the call goes through OK. [2008-10-14 15:41:40] -- Executing [s at macro-to-isdn:1] Dial("SIP/216-bc0aab90", "Zap/g1/0123456789") in

Hello, I'm using Asterisk with an ISDN30e PRI line (only 16 channels active). Every now and then I get a CONGESTION error even-though there are only 1 or 2 channels in use out of the 16 at that time. When this happens, the user just needs to re-dial and the call goes through OK. On a SNOM phone when the problem occurs, a "Service Unavailable 907" error is shown. [2008-10-14

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call

Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the

Here is some info that may allow some "locked" PAP2 and PAP2-NA units to be used with Asterisk: I have a PAP2-NA (from a provider other than Vonage) for which I did not know the admin password, though the "user" pages were accessible to me. The provider had set it up to fetch at startup, its configuration file by HTTP from a numeric IP. It was running 2.0.10(LSc). A search

I have a handful of Linksys PAP2-NA's all talking nicely to Asterisk using standard telephones. I've been running them for the better part of this year. No complaints whatsoever. We chose the PAP2-NA's mainly due to cost and especially the ease of provisioning. In an effort to inexpensively bridge our office PBX (InterTel Axxess) to our VoIP network, we've opted to connect

Hi, I have 3 PAP2 connected to *, they work fine but there are some things which I would like to improve, some of them are: - double ring tone when placing a call (I hear two tones it seems like the PAP2 is generating it's own tone) - some kind of noise (like glitches or something) when I pick up the phone (seems like some polarity thing) - I'd like to keep the tone after

1) I'm working on a quick replacement for DISA, and I ran into the following snag: When I specify "Playtones(dial)" I can only get around 7 seconds of wait time before the dialtone stops, and the context goes to the "h" extension. Is there a way around this fixed timeout? The DigitTimeout setting doesn't seem to have any effect at all on this hangup problem. I

Hi, I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem when trying to play a Busy tone over a IAX trunk from the PSTN. It seems as though Busy(20) returns non-zero immediately (it does not wait 20s), so the caller never hears the busy tone, but the call just appears to hang up. I don't believe this happens when trying to play a Busy on a SIP trunk. The busy part of

Hello, We use Asterisk with PAP2 and today we connected the FXS ports of PAP2 to CO ports of our Panasonic KX-TD1232. Problem is that Panasonic doesn't ring - that is doesn't ring every time the PAP2 is ringing. When we reset either Asterisk or the PAP2 it usually rings, but after couple of minutes it stops and only the automatic operator is answering - after 2 rings. We tried changing

Hi everyone, I'm in the very early stages of rolling out an asterisk box at work, and one of the things I'm setting up is a trap for telemarketers >;) What I want to do is have a sipgate number in the UK here which rings for 10 seconds without calling a hard or softphone, then goes to a voicemailbox. The problem I'm having is that Playtones doesn't seem to be sending any

The best I can do so far (which appears to be a bit of a hack) is (<:0>S0), which says to add a '0' to the start of the string and dial immediately. This gives asterisk an extension dialled of '0', which isn't the 's' that i'd hoped for, but is a good start! (S0) by itself doesn't work, nor does (<:>S0). Any other suggestions? Thanks James >

hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked