Glad I could help. One suggestion I may have is to use hardware echo canceller cards from Sangoma. They perform MUCH better that software echo canceller. And yes, you can have echo on a PRI circuit. Have experienced it and switching to a hardware echo canceller has made me sleep MUCH better.
--- Andre Courchesne - Consultant http://www.net-forces.com MSN: courc...@net-forces.com Skype: VoipForces L'information contenue dans le présent document est la propriété de Andre Courchesne. Et est divulguée en toute confidentialité. Cette information ne doit pas être utilisée, divulguée à d'autres personnes ou reproduite sans le consentement écrit explicite de Andre Courchesne. The information contained in this document is confidential and property of Andre Courchesne. It shall not be used, disclosed to others or reproduced without the express written consent of Andre Courchesne. On 2009-12-22, at 8:28 PM, Alex Robar wrote: > Andre, I owe you a beer! I made a very silly mistake in my dialplan, and the > dialed number wasn't actually getting passed to the trunk. Unfortunately I > figured it was something to do with the new technology I wasn't familiar > with, and one of the basics slipped by. > > Thank you everyone for your suggestions! > -- > Alex Robar > alex.ro...@gmail.com > > > > On Tue, Dec 22, 2009 at 7:34 PM, Andre Courchesne <courc...@net-forces.com> > wrote: > Looks like you are trying to dial 300 via SIP extension 50555 through the PRI: > > Dial("SIP/50555-00e73ea0","DAHDI/g0|300|") > > If that's the case and your PRI is connected to a telco, that is not valid. > > Also what is the output of: > > pri show spans > > Also looks like you are using freePBX so you could try the following. Add > this under [from-internal] in extensions_custom.conf > > exten => 1111,1,Dial(DAHDI/1/14087658080) > > Then from a handset dial 1111 > > That 408 number is INtel HQ in California. I use it cause it's easy to > remember and it always answer on an IVR... > > --- > > Andre Courchesne - Consultant > > http://www.net-forces.com > > MSN: courc...@net-forces.com > > Skype: VoipForces > > > > L'information contenue dans le présent document est la propriété de Andre > Courchesne. Et est divulguée en toute confidentialité. Cette information ne > doit pas être utilisée, divulguée à d'autres personnes ou reproduite sans le > consentement écrit explicite de Andre Courchesne. > > The information contained in this document is confidential and property of > Andre Courchesne. It shall not be used, disclosed to others or reproduced > without the express written consent of Andre Courchesne. > > > > > > On 2009-12-22, at 5:39 PM, Alex Robar wrote: > > > Hi all, > > > > We're very new to PRIs, and I'm having trouble getting one working properly > > on one of our Asterisk boxes. Inbound calls through the PRI work properly, > > but outbound calls are rejected immediately. Details below (if I've missed > > something important, please let me know... Not sure what's required to > > troubleshoot this yet). Hoping someone out there might have some hints. > > Willing to hire someone to take a look if that's what it takes. Thanks in > > advance! > > > > Asterisk version: 1.4.26.2 > > Hardware: Sangoma A102 > > Wanpipe version: 3.5.6 > > PRI coding: B8ZS > > PRI framing: ESF > > Switchtype: NI2 > > > > > > CLI ouput when placing the call: > > ----------------------------------------------- > > -- Executing [...@macro-dialout-trunk:26] Dial("SIP/50555-00e73ea0", > > "DAHDI/g0|300|") in new stack > > -- Requested transfer capability: 0x00 - SPEECH > > -- Called g0 > > -- Channel 0/1, span 1 got hangup, cause 90 > > -- Hungup 'DAHDI/1-1' > > == Everyone is busy/congested at this time (1:0/0/1) > > -- Executing [...@macro-dialout-trunk:27] Goto("SIP/50555-00e73ea0", > > "s-CHANUNAVAIL|1") in new stack > > -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) > > -- Executing [s-chanunav...@macro-dialout-trunk:1] > > GotoIf("SIP/50555-00e73ea0", "1?noreport") in new stack > > -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) > > -- Executing [s-chanunav...@macro-dialout-trunk:3] > > NoOp("SIP/50555-00e73ea0", "TRUNK Dial failed due to CHANUNAVAIL > > (hangupcause: 90) - failing through to other trunks") in new stack > > ----------------------------------------------- > > > > > > chan_dahdi.conf: > > ----------------------------------------------- > > [channels] > > context=default > > usecallerid=yes > > hidecallerid=no > > callwaiting=yes > > usecallingpres=yes > > callwaitingcallerid=yes > > threewaycalling=yes > > transfer=yes > > canpark=yes > > cancallforward=yes > > callreturn=yes > > echocancel=yes > > echocancelwhenbridged=yes > > relaxdtmf=yes > > rxgain=0.0 > > txgain=0.0 > > group=1 > > callgroup=1 > > pickupgroup=1 > > immediate=no > > > > ;Sangoma A102 port 1 [slot:4 bus:8 span:1] <wanpipe1> > > switchtype=national > > context=from-pstn > > group=0 > > echocancel=yes > > signalling=pri_cpe > > channel =>1-23 > > ----------------------------------------------- > > > > /etc/dahdi/system.conf: > > ----------------------------------------------- > > loadzone=us > > defaultzone=us > > > > #Sangoma A102 port 1 [slot:4 bus:8 span:1] <wanpipe1> > > span=1,0,0,esf,b8zs > > bchan=1-23 > > echocanceller=mg2,1-23 > > hardhdlc=24 > > ---------------------------------------------- > > > > > > -- > > Alex Robar > > alex.ro...@gmail.com > > --------------------------------------------------------------------- To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
