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All, We are in the process of trying to setup our network to use Verizon's SIP "trunking" product. They say that since Asterisk is not on their certified list of approved devices, we need to go through a field trial to get it approved before allowing us to use their service. Where we are at is getting the design approved. We are trying to watch our budget at the same time. We

...= MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs I'm going to make this change and reload at lunchtime, I'll document it and post it to the list if it works. thanks for the good eye. -----Original Message----- From: Michael L. Young [mailto:myoung@acsacc.com] Sent: Wednesday, January 03, 2007 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts < Zaptel.conf: < loadzone = us < span=1,1,0,esf,b8zs < span=2,1,0,esf,b8...

...n the PRI's and we get no more artifacts. So, : wanpipe1.conf: TE_CLOCK = NORMAL TE_REF_CLOCK = 0 wanpipe2.conf: TE_CLOCK = MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs -----Original Message----- From: Michael L. Young [mailto:myoung@acsacc.com] Sent: Wednesday, January 03, 2007 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts < Zaptel.conf: < loadzone = us < span=1,1,0,esf,b8zs < span=2,1,0,esf,b8...

I need some help getting a system running for one of my company's plants. I am running AsteriskNow 1.7 with Asterisk 1.6.2.10 and FreePBX 2.8.0.2. My D-Channel keeps bouncing. The telecom tech told me he thought that I might be using the wrong sync source, and I think I might have been, but I changed DAHDI system.conf to "span=1,1,0,ESF,B8ZS" (from

----- Original Message ----- > From: "sean darcy" <seandarcy2 at gmail.com> > To: "Asterisk Users Mailing List, Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Tuesday, January 21, 2020 9:22:28 PM > Subject: [asterisk-users] permission woes on systemd [..] > So why would starting asterisk as user asterisk work, but fail using

> From: "John Hughes" <john at calva.com> > To: "Asterisk Users Mailing List, Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Sent: Thursday, May 14, 2020 2:10:45 AM > Subject: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and > alaw; asterisk wants to translate g729 -> alaw. WHY? > I am having a

We are in the process of installing a PRI line and we are going to connect it to an Asterisk Box. Verizon called us today to find out some information. I am surprised that they have never heard of Asterisk or Digium. But anyways, they needed some information in order to set up the circuit. Does the TE110P support NI1 or NI2? (I think the answer is both) What is the number of digits

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