similar to: Keeps Ringing After Answer

Asterisk Users: We have our * dial plan set up to ring 5 phones in the office and it delivers the call to the first that picks up their receiver. The problem is that after the pickup, everyone else's SIP phone keeps ringing at least once and sometimes twice. This interferes with the conversation, while others pick up the phone and get nothing. Does anyone else have similar problems or

Hello Everyone, I have a strange issue where the caller's phone keeps ringing even after the 200 OK. I am using the latest version of Asterisk 1.8, and wanted to know if anyone could give me any pointers before posting the SIP debug messages. Kind Regards, Nick from Toronto.

Hi, I am new to this group.I searched for my problem in the forum but could not find any solution. So here it goes: In my work place we have an asterisk box. Everything works fine except the fact that when I first call the work phone number from my cell the auto-attendend works fine but If I hang-up and call the same number again, the call keeps ringing instead of auto-attendent answering. I also

Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing even after they've been picked up... Here's one users summary: When I pick up the phone, I hear a dial tone and I am able to dial out. But for some odd reason, the receiving line picks up while the outgoing line is still ringing. And the receiving line can hear everything while the phone is still ringing. I tested

Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and

Me again... I have service with a company that does not allow for a BYOD plan. They will not give out credential or server info either. Is it possible to run the FXS port of the ATA to an FXO port in *? The service I have is throug Broadvox Direct using the Mediatrix 2102. I have tried this using loop start and kewl start. The * box sees the incoming ring, picks up and starts my dial plan. But

Hello, As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco 7960 with *8 but the extension then keeps ringing indefinitely, even though I picked up the call. Is this a known issue? Thanks, -- There are no Debian developers in any part of Hell, because the good karma incurred by being one takes you straight to the pearly gates. Of course, the frequent flame wars you put

I have 3 phones, SIP/3, SIP/6 and SIP/9 SIP/3 subscribes on hint on SIP/9 Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it is steady red. That's correct. But when 9 hangs up the hint goes to "InUse&Ringing", the light on 3 is still flashing. It keeps flashing until somebody calls 9 and hangs up again. -- Executing [9 at local:1]

Hi, when I press reject on Firefly, another end hears congestion, but I'm still listening to ringing... Also how you determine which version of Firefly you have? I also don't understand licence completely and would like to get this answer - can I distribute Firefly to commercial users if I don't charge anything ? Regards, Rob.

Well in my setup I have a few IP phones connected to Asterisk as well as POTS phones on my analog line. When a call for my daughter comes in on the analog line (determined from callerID) I send it to her own voicemail after 20 seconds of ringing. It all works quite well. Here's a step-by-step of what happens below: 1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds.

Hello, I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows: sip.conf: [169] secret=111111 dtmfmode=rfc2833 directmedia=no directrtpsetup=yes canreinvite=no context=main host=dynamic type=friend port=5060 call-limit=5 nat=force_rport,comedia

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows:

Using a TDM400P in the UK nearly works fine, but I have a last remaining problem in that if the incoming is ringing and then the caller hangs up, asterisk takes another couple of rings before it spots the hangup. This is annoying in that I sometimes get phantom calls late at night (possibly due to call waiting or the exchange doing a half ring to see if we are live). Also I get phantom calls

Hi, I have a problem with dialing status. I'm using Asterisk 1.6 and a patton 4554 gateway for ISDN calls. When I call fixed telephone (not mobile phone) after few ringing the status change to "answer" but the phone is still ringing, so if I hangup before someone really answer, the call is reported as answered but it isn't. This gives me problem for call charge. Some I idea

I have a line in my house that goes to phones across the line BEFORE Asterisk. My wife uses them :) I plugged the line into an FXO port and pointed it at an extension, so I can pick it up as well. I can also pick it up with *8. One side effect is that once my wife picks up a call, the Asterisk extension continues to ring until the Zap card detects that it's not ringing any more. What

I have a bunch of grandstream gxp2000's in the office. When I Dial() multiple extensions at once, eg Dial(SIP/2000&SIP/2001&SIP/2002) and two extensions answer simultaneously, one of them gets the call and the other hears a ringing indication in the handset. Anyone else seen this bug? -Dan

Hi, I use asterisk with sip3000 device with "sip-aho" connected to PSTN and "sip-ahi" connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten => 99,n,Dial(SIP/sip-ahi,30,g) exten => 99,n,Hangup() The asterisk properly detects hangup of the caller as I

Hi, I've successfully used the 'd' flag in Dial() so that when I dial into my phone system from out there in the PSTN network I can press the 2 key while the phone is ringing to listen to my voicemail. It seems that one issue is that the public providers do not deliver DTMF, or anything, until the phone is answered. This is for security reasons and sounds like a good idea to me.

I ran into a couple of odd problems with a new Asterisk box I have set up... I have a T100P connected to a channelized T-1 (24ch) to the PSTN. The T-1 is set up properly so far as I know and seems to work perfectly with one exception: When a call comes in, asterisk starts a 'simple switch' for it, goes through my 's' extension progression until it gets to the first Playback

I am running into two problems: 1) The ringing stops during call screening once the extension picks up (but has not yet approved call) When a person calls and choose an extension, the Dial link is called and the person hears the ring -- but as soon as the receiving caller picks up (even though they have to approved the call), the ringing stops... the person calling hears silence. This is