similar to: 25 second delay, then busy...?

In a system connected to a verizon T1, Digium TE411P (quad T1 echo cancellation), client is complaining it is "too quiet". The complaint regards calls over the T1, not in house SIP only calls. Their description indicates they want some earpiece feedback of themselves speaking. Also, they complain that it takes several seconds (3-4) for the other party to respond. That is kind of

I just wanted to see if any one else has seen this or could help point me in the right direction on this problem. I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I

Hi, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has

I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or . any other brand and model that is known to work well with the TE411P ? Will an old Proliant do? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a TDM400P with 2 FXO cards and I'm using Asterisk@Home 2.8 I noticed that when I place a call to the analog lines from outside, Asterisk takes a while to actually ring the extension the call is being sen to. I've been doing some tests, calling from my cellphone and here is what I see... - After the first ring on my cell, Asterisk logs to the CLI that is has an incoming call -

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf

Hi, Does a parameter exist for a queue to delay ringing/sending a caller to all agent phones after the previous call is answered by an agent? My queue ring strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones. And it looks like the KIRK wireless server may need a split send to realize all wireless phones are no longer ringing (busy) after 1 call rings & is unanswered,

Hi, I have a strange problem with our Asterisk installation. Outgoing calls are handled by the following lines: exten => _0[2-9]X.,1,Set(CALLERID(num)=09999403${CALLERID(num)}) exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} = 0999940321]?099994030:${CALLERID(num)})}) exten => _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr) exten =>

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

I'm trying to pass a 65K DATA call in one channel on my Digium TE411P to another channel on a different span. Any idea what could keep this call from going through? -- Accepting call from '' to '5444' on channel 0/1, span 1 -- Executing Goto("Zap/1-1", "sendto-definity|5444|1") in new stack -- Goto (sendto-definity,5444,1) -- Executing

We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1, Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two TE120P cards and everything was working fine. Since they needed to add a third E1 line we decided to change one of the TE120P cards with a TE210P. After the change we had a couple of crashes (server

I'm new to asterisk and already a fan. Please forgive me if my questions are covered by some FAQ and thanks in advance for any pointers anyone can give me. The basic problem that I'm having is that sometimes outgoing calls result in a busy signal when the outgoing line is free. I'm thinking that the channel is timing out or something but haven't figured out how to debug or gather

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Hi! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>

We have a customer who wants to forward an extension to their cell phone, if and only if that extension is "unavailable", or when the Dial() command times out. However, should the Dial() command return "busy" it should go to voicemail instead. As far as I know, the dialplan doesn't support this. Certainly not natively or in any particularly easy or obvious way, and I

Hello, Has anybody seen that Audiocodes gateway is replying with "486 Busy here" when it's actually not (last call ended ~15 seconds ago). I see this quite often. From other logs i see that previous call ends at 11:13:01, then app_queue tries to dial at 11:13:14 and fails numerous times, before succeeding at 11:14:02 I have attached sample SIP debug log: Any ideas what i could

Hi, I'm using R to collect data for a number of exchanges through a socket connection and constantly running into memory problems even though task I believe is not that memory consuming. I guess there is a miscommunication between R and WinXP about freeing up memory. So this is the code: for (x in 1:length(exchanges.to.get)) { tickers<-sqlQuery(channel,paste("SELECT Symbol