Cisco ATAs come in two types ATA186-I1 with 600 ohm impedance and ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150 NF in parallel) What is the difference between the two ? Which one is suitable for Europe ? Thanks, Dave
Search the list - there's a detailed answer on it. I have two of the I1 version (at least that's what they say they are - ProductId: ATA186I1) and they work with UK spec phones. All you need to watch for is that UK phones are three wire and US phones are 2 wire. Maplin sells an adapter to sort this out (Part no. VD36P). Iain --On Wednesday, February 11, 2004 4:54 pm +0100 Dawid Mielnik <D.Mielnik@elka.pw.edu.pl> wrote:> > Cisco ATAs come in two types > > ATA186-I1 with 600 ohm impedance > and > ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150 > NF in parallel) > > What is the difference between the two ? Which one is suitable for Europe > ? > > Thanks, > > Dave > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? -- _______________________________________________ Get your free email from http://www.hackermail.com Powered by Outblaze
I had the same problem with a Mediatrix, it turned out to be a defective unit. No matter what we did the audio was very choppy, when I replaced the unit my problems went away. Are you running it as SIP or MGCP? Norm -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Gonzalo Gasca Sent: Wednesday, July 21, 2004 12:22 AM To: asterisk-users@lists.digium.comu Subject: [Asterisk-Users] Cisco ATA 186 Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? -- _______________________________________________ Get your free email from http://www.hackermail.com Powered by Outblaze _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Gonzalo Gasca wrote:> Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? > Any working configuration?Turn VAD off on the 1204. * can not clock itself. -- Bob Knight [-w] the work option bk@minusw.com 925-449-9163
I'm nothing understand now. I have Cisco ATA 186 with one analog phone and
the following problem:
The next config works just fine:
sip.conf:
[150]
type=friend
port=5060
context=officepbx-outgoing
qualify=yes
secret=password
user=150
username=150
fromuser=150
defaultip=XXX.XXX.XXX.XXX
host=dynamic
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=g729
allow=alaw
extensions.conf:
TEST = SIP/150@150
agent_150 = ${TEST}
exten => 150,1,Macro(callfullext,${TEST},,,30,N)
But if I rename 150 to "Cisco", by example, I get the following
error
message:
NOTICE[1174440880]: chan_sip.c:7519 handle_request: Registration from
'<sip:150@YYY.YYY.YYY.YYY;user=phone>' failed for
'XXX.XXX.XXX.XXX'
sip.conf:
[Cisco]
...
extensions.conf:
TEST = SIP/150@Cisco
...
Cisco configuration:
UID0: 150
PWD0: password
UID1: 0
UseLoginID: 0
What is going on?
--
Serge Matveev
Relcom Corp., St.Petersburg