similar to: 1 second delay when connecting calls

Displaying 20 results from an estimated 7000 matches similar to: "1 second delay when connecting calls"

2008 Dec 03
6
Call parking
Hi, Been playing with Call parking, and I can`t help but wonder if I am doing something incorrectly. The way I understand it (using default config in features.conf), is I would transfer a call to extension 700, which would park the call, tell me "701". I could then hang up, go fetch the fright person and tell him "call 701 you have a call waiting for you". The way I
2008 Oct 17
1
Strip prefix
Dear All, i have the following context defines in etensions.conf: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup exten => _X.,21,Playback(AR_GetGiveToID) exten => _X.,22,Wait(2) exten => _X.,23,Record(/tmp/asterisk-recording:ulaw,,5) exten => _X.,24,Wait(2) exten =>
2008 Nov 12
1
How to get correct dial result for outgoing calls thru ISDN?
Hi everyone, Currectly I'm having some troubles to get correct status of my calls throug ISDN lines, when outbound calls don't get its destination I always receive NO ANSWER as ${DIALSTATUS} despite the fact I know the target number doesn't exists or is busy at that time. Maybe there is something I must change in my zaptel.conf or zapata.conf, current configs follows: ####
2008 Oct 30
3
SIP # DTMF
Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only "333" What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica Descentralizada y Lucha
2008 Nov 20
4
Using MAC or extension number as SIP identifier
Hi, For a long time, I was wondering if I should use MAC address instead of Extension number to identify SIP endpoints (as I'm mostly not using softphones). Before diving into this, I wondered how people using MAC address are using CLI as it seems more natural and simple to type "sip show peer 4566" as opposed to "sip show peer 00147F784512". Is there something obvious
2008 Dec 03
3
canreinvite=yes problem
Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you -------------- next part -------------- An HTML
2008 Dec 04
2
Packet size limit for HDLC?
Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with
2008 Nov 12
4
E1 PRI to and from SIP screeching
Hi all, We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching
2008 Oct 17
4
srv records not being honoured properly
Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen
2010 Jan 28
4
Latency and Rsync Transfers
Hello, Working a few servers that are transferring data across country with a 75ms delay on a GIGE connection. We can tune the tcp buffers on linux to improve the connections using iperf. Does rsync use the tcp buffers of the OS or does it override these settings? Thanks, Neal -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Dec 04
5
We think we are cpe but they think they are cpe too
Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but
2006 Jun 13
7
delay in MeetMe
Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Jul 12
0
No subject
That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >
2008 Oct 10
3
Got event 17 (Polarity Reversal)...
Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]:
2008 Sep 29
3
Knowing incoming call technology and channel [SOLVED]
2008/9/29 Alex Balashov <abalashov at evaristesys.com> > Try this: > > exten => _XXXX,1,Set(THISTECH=${CUT(CHANNEL,/,1)}) > exten => _XXXX,n,NoOp(Technology is ${THISTECH}) > exten => _XXXX,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)}) > exten => _XXXX,n,NoOp(Channel is ${THISCHANNEL}) Hi, I don't have any spare zaptel enabled system I could try this on, but I
2006 May 02
4
Under which project , auto-dial feature comes
Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___________________________________________________________ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.
2008 Oct 10
3
Question about echo cancelation
Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,
2007 Jul 12
0
No subject
like this: print STDOUT "EXEC background /var/lib/asterisk/sounds/wait-moment \n" system("program2.agi &") exit; program 2 would run while the sound played. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: Tuesday, December 02, 2008 3:20 PM
2007 Jul 12
0
No subject
like this: print STDOUT "EXEC background /var/lib/asterisk/sounds/wait-moment \n" system("program2.agi &") exit; program 2 would run while the sound played. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: Tuesday, December 02, 2008 3:20 PM
2008 Mar 22
3
G723 on asterisk 1.4.1
Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE