Re: [asterisk-users] X100M never goes on-hook state
Hello, On Mon, Apr 14, 2008 at 10:30 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: It's possible the 'line relay' on that card is not a physical relay, but electronic, and that its sensitive to too much loop current -- and the DSL filter drops the current far enough for that 'relay' not to pull in spuriously. Yes, I was suspecting it was something like that. I bought a 33-ohm resistor (1/2 watt) and it fixed the problem. Both modules are working properly now. According to the Ohm's law, the loop's overall resistence was 188.46 ohms, once I was measuring 4.9V and 26mA in the circuit. By adding 33 ohms, it went to 221.46 ohms, bringing the current down to 22 mA. Telecom guy Mike Sandman has a paper on loop current on his website: Excellent article and telephony stuff. Bookmarked! Thanks for the reply. Have a nice day. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100M never goes on-hook state
Hi guys, I've been experiencing a very strange issue with my Digium Card TDM400 as of this week. It has two FXS and two FXO. The FXO modules (both of them) never goes on-hook after hanging up in Asterisk. It had worked perfectly well for over four years. I put an ammeter in series with the line and the card, and immediately after plugging the connector to the card, I got 26mA in the circuit and a dial tone from the carrier, where it should be zero amper (on-hook state). If a Dial() something, it works perfectly. I can Hangup() the call, freeing the channel in Asterisk, but the hardware keeps off-hook forever, locking the line. If I Dial() again, Asterisk opens the line, sends the DTMFs normally, but it doesn't work since the carrier thinks I'm still holding the first call. It behaves exactly the same way with another analog line. If I plug either of the lines and my other Digium card (TDM2400), it works ok. The same with my Brazilian DigiVoice FXO card. Ok, you all might say: your card is damaged, throw it away. Ok, I could do it, but now comes the funny part: If I put an DSL filter in series with the line and the card, IT WORKS PERFECTLY!!! The filter imposes 25 ohms over the circuit. Maybe that's causing the card to work. When I put the filter and the ammeter in series, I get zero amper when on-hook and 26 mA when off-hook, that's the expected behaviour. I'm not an expert in electricity, so I really don't know why the card is behaving that way. What does that resistance make for the card to start working ok? I know the DSL filter isn't only a resistor. Maybe it has another electrical component that's helping more than the resistor. Just a guess. Tomorrow I'll buy a 30-ohm resistor, take the DSL filter off, and test the card only with the resistor, to check it out. In order to isolate the problem even more, I plugged the FXO port in one FXS port. Immediately after plugging it, Asterisk announced at the console that someone went off-hook at the FXS port. So, it's not really a carrier issue. The FXS port is perfectly -48V on-hook, and about 20 mA in the circuit when off-hook, closer than the carrier to the standard values. Any clue is welcome. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Keeping queue counters after restarting
Hello, Every queue has some status counters (completed, abandoned, hold time...) that are very useful for statistics. The problem is that those counters are reset every time Asterisk restarts. Is there a way to keep those counters, maybe in astdb? Also, is there a way to reset the counters through a cli command? Thanks. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keeping queue counters after restarting
Hello, On 8/24/07, James FitzGibbon [EMAIL PROTECTED] wrote: Not sure about restarts, but trunk keeps them through reloads. How often are you restarting? My Asterisk has been segfaulting a few times during the day. I couldn't figure out why that's happening. safe_asterisk restarts Asterisk immediately, but all my calls are dropped and I lose the queue stats. I'll check that 'keepstats' option. Thanks. Regards. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent Channel SIP transfer
On 11/22/06, Xue Liangliang [EMAIL PROTECTED] wrote: Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer call using SIP phone's transfer feature, he is always in busy status and cannot answer any more incoming call from queue until the transferee hang up the call. I'm experiencing the same problem here with Asterisk 1.4.5. Is there a solution for that? -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linking Asterisk with PBX through E1
Hello,
I linked an Asterisk server to a Brazilian PBX (Leucotron) through an E1
connection, using MFC/R2, that's common down here. The connection works
properly. I'm able both to dial and receive calls through that link,
among their extensions.
The problem is that the PBX configuration is very tough. Just a few
options in the GUI software and I cannot play with it in lower level.
That PBX has two E1 interfaces. One of them is connected to the PSTN and
the other to the Asterisk server. Both connections are working ok.
I need to make calls from the Asterisk server to the PSTN, i.e., coming
from an E1 and going through the other one. Here is my pain. That PBX
assumes that an E1 connection is always PSTN, so an E1 link doesn't
need to talk to each other. Zero flexibility.
The manufacturer support gave me a solution. Coming from Asterisk, I
can dial a special code, then I get a simulated dial tone, and then I
dial (through DTMF) the number I want. That's odd, but it works.
In my case, that code is . Since E1 is digital-signaled, the best to
do would be dialing just like I do between two Asterisks:
exten = _,1,Dial(Unicall/g1/${EXTEN})
But it doesn't work. The PBX just ignores the numbers after and
gives me a dial tone.
Another way would be dialing and then sending the number to dial
through DTMF tones, with something like this:
exten = _,1,Dial(Unicall/g1/|20|D(w${EXTEN}))
That would work, BUT a little detail broke my legs. The Dial
application only sends the DTMF tones after receiving the channel
answered signal from the E1 channel, and that PBX only sends that
signal when the remote party has answered the call, what's useful for
accounting purposes. So, when I dial something using the above dial
plan, Asterisk dials and I hear the dial tone. If I dial something
in my phone (DTMF), the PBX hears that and makes the call. When the
remote party answers the call, the Dial application releases the DTMF
tones.
Possible solutions:
1) Finding a way that Asterisk sends the DTMF tones immediately after
opening the channel, without waiting the answer signal.
2) Making the PBX works the way it should do, receiving all the numbers
in the digital channel and making the call without simulating any dial
tone.
I'm not hopeful that the manufacturer will be able to change the way the
PBX works, so I better keep looking for the first solution.
Any help is pretty welcome.
TIA
--
Marlon Dutra
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