similar to: DTMF from SIP phone to FXS/FXO

Displaying 20 results from an estimated 1000 matches similar to: "DTMF from SIP phone to FXS/FXO"

2009 Sep 24
1
rtp.conf dtmftimeout
What unit is dtmftimeout measured in? The sample configuration is provided below. Does it mean to say that the sample configuration file's dtmftimeout=3000 equates 1/8000th of a second? ; The amount of time a DTMF digit with no 'end' marker should be ; allowed to continue (in 'samples', 1/8000 of a second) ; ;dtmftimeout=3000 -- Brian Camp IT Freedom direct
2005 Aug 25
0
Internal FXS to SIP problem
I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and a couple computers with eyebeam. I have one small. I cannot call the eyebeam clients from the phone connected the fxs port. I can call the phone from the eyebeem clients. And, I get both the fxs phone and eyebeam clients to ring when a call comes in through the fxo port. I have been trying to get this straightened out
2007 Jun 25
0
four ringing and hangup with error
Dear All I have this setup [asterisk]----[mediant2000]-------E1 Trunk----------[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip
2007 May 06
2
Were i make mistake
I've found some manuals and tried this to do : Sip.conf [test] type=friend username=test1 secret=test1 host=192.168.1.238 context=tutorial fromuser=SIP Phone callerid=101 nat=no canreinvite=yes dtfmode=info disallow=all allow=ulaw [test] type=friend username=test secret=test host=192.168.1.240 context=tutorial callerid=100 nat=no canreinvite=yes dtfmode=info
2009 Jan 21
0
About Asterisk 1.6.0.1
Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and
2008 Nov 10
3
directrtpsetup without reinvite
Hi, I want to be able to bridge two sip channels using direct RTP between my endpoints (Audio IP : not local) but without using reinvites. So I set up my asterisk sip endpoints as follows: [test1] type=friend host=dynamic username=test1 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes [test2] type=friend host=dynamic username=test2 dtmfmode=info context=test_rtp
2009 Jan 22
0
Query About Asterisk 1.6.0.1 Dialog Event Package.
Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and
2012 Jan 13
1
Sporadic one way audio problem
Hi all again, I've got a problem with sporadic one way audio calls, which means sometimes I can't hear the calling party (call is established, but audio is missing). Today I received ~90 calls, one of them got this problem. I've got two networks involved, without NAT: - 192.168.1.X, in there one nic of my server and all the phones - a private net to my provider, in there a nic of my
2007 Jun 12
2
Softphone behind NAT issues
We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching
2013 May 24
1
Registration timed out - for created users
Hi all ,? I have managed to install and configure the? 1. asterisk-1.8-current 2. dahdi-linux-complete-current I did not faced any issues during the installation. After that I installed X-Lite soft phone in two different PCs and tested the setup. every thing was success. I was able make calls from each?extensions. But when I observe the log files , i could see some messages ......
2009 Aug 27
1
Bad Gateway
Hey guys, I've been having a very odd problem that happens intermittently. I've had this happen with only a couple of providers and somewhat rarely but its to the point now that we need to fix it to be able to do business. The scenario is as follows: We have a DID provider that routes calls to our asterisk boxes and we have an outbound provider to whom we send the calls of the person
2008 Dec 01
2
Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12
2008 May 25
3
trying directrtpsetup
Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron
2009 Jan 11
2
sip peer permit/deny - Need some explanation
Hi all, I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result. Here is the problem: I have a peer -which is peer AND user- setted up like this [MyPeer] ; type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 context=from-MyPeer dtfmode=auto disallow=all allow=ulaw,alaw
2010 Sep 27
1
propagate sip reinvites with directrtpsetup=yes
is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't
2014 Jul 02
1
Webrtc Not acceptable here
Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF
2015 May 11
0
"Retransmission Timeout" results in dropped calls after 32 seconds
----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:24:53 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > Could this
2005 Aug 08
2
[OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
With the lack of info on Yoda Communications in Taiwan and their hardware, I thought I'd post my experience. I got my hands on a few H.323 VG-400's and VG-100TA's. http://www.yoda.com.tw/model.php?type=VoIP_Solution&pname=VG400 2 of the VG-400's were 2FXO/2FXS models. A couple were deployed to SE Asia, where we planned to offer our services. Originally, I ran a GnuGK server
2005 Jan 03
0
SPA-3000 as FXO Gateway for * (Was: Qs about FXO/FXS cards)
Thanks Rich, I have an SPA-3000 laying around, so I will attempt to set it up in a little more conventional manner (although your method looks like a winner for a home test PBX). Would you mind posting or PM your current config to me, maybe screenshots if you PM. If I start with that it will take less time to get to the point where the SPA-3000 is a true FXO-FXS gateway for *. I will be happy to
2007 Aug 19
1
Someone please explain FXO/FXS module/channel relationships in Zaptel configuration to me
Please explain the relationship between modules from the driver (wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if I have a FXS module 0 and FXO module 1, what should be used in zaptel.conf and what should be used in zapata.conf? Then finally, in extensions.conf, what is the Zap channel for dialing out? Zap/? % dmesg Module 0: Installed -- AUTO FXS/DPO Module 1: