similar to: CallerId CheckSum Errors on x100??

I know a few ppl have those CAC Adit 600's with t400 I can't seem to get my second span up on the T400 connected to the second spand on the adit (A:2) A:1 seems ok Can someone post they zaptel.conf span defintions And maybe a "print config" from the adit 600 cli I think my issue is timing srcs the coding, framing. bld out are all matched thx -------------- next part

Callerid issue 1) if you run ztmonitor on the fxo line & call in do you hear the fsk tone if yes then we beleive the CAC is passing fsk 2) in chan_zap->ss_thread around line 4154 (current cvs) if you get to the callerid_feed at least once then if you get to chan_zap->ss_thread->callerid_get around line 4163 (current cvs) does this parse fail

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

I am a little confused about the FXO ports on the Carrier Accesss Bank 1. The wiki, and people on the mailing list have stated that they do not have impedance matching, but Carrier Access states in their documentation that they do have impednace matching. Did Carrier Access make a typo, or does it depend on the version of the FXO card? I current have a CAC AB1, and I am having echo problems

hi: in our country callerid is sent with fsk. but it will sent DTAS(dual tone alerting signal) first, then fsk callerid, then first ring. I search google, but didn't find the configuration method or patch for this. any good suggestion for asterisk to detect this? I have trid asterisk 1.4 and 1.6.2. thanks for help!! Regards, tbskyd

Hi ! I am connecting to Adit 600 thru a T100P card I have configured 1-16 FXS channels and 17-24 FXO. Everything looks fine on Asterisk side I get a tone on all FXS channels, but when I try to dialout thru one of the FXO channels 17-24 it doesn't connect to the POTS line and echoes back my voice. I use fxsls and fxols for the T1 channels and ls on Adit side. Whats wrong here ? here is my Adit

Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much

Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no

Jon, I actually had some more discussions with Tim on this issue, and it seems that the channel bank would still be a good option to choose for internal purposes. I would not see any other solution than a channel bank to connect many 2wire phones into one asterisk box. I had a talk today with Carrier Access, and it seems that the adit would do us fine. The fxs cards of the adit 600 are

>> I'd have to agree with the channel bank plan. You don't even need to >> buy a used channel bank. You can get a new Rhino for $1500: >> >> http://www.channelbanks.com/ > > I gather you've had experience with Rhino. > > How does it work with Asterisk? > Does it provide all features? Caller-id, echo cancellation, how's the > voice

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

As a follow-up to my previous post where I stated the IAD would be a Vina model, after some more prodding to XO, they have told me it will either be an Adtran TA-600 or a CAC Adit 600. These products are covered pretty well on the web and I have manuals on both. So, if those knowledgeable folks had to use one of these to attach to an Asterisk box, what interface would be best or at least

At the risk of being flamed.... Has anyone had any success get the 'El cheapo' Wildcard W100P clone's (?20 flavour) to work with UK Caller ID? I'm not sure what the status of Asterisk 1.6 is with respect to UK caller ID, being that we have an odd method of sending the FSK ahead of the ring, but I'm guessing I can't be the first to ask this? Keeping in mind that cost is

I have an adit 600 with one fxo card connected to a Digium single span T1 card. CallerID, disconnect supervision work perfect, however the users complain that they have some sound quality issues, after testing it I realized that whenever one is in a phone call they get like silence between the sounds coming from the other party, almost like a cell phone, in other words if there is no sound coming

Follow this diagram: Many POTS lines <-> Many Channel Banks <-> Mux <-> DS3 <---cloud---> DS3 <-> demux into T1s <-> Many Asterisk's If someone calls into 1 of the asterisk boxes (via PRI or VoIP), and I send the call back down the line above to a POTS phone, who will provide caller id to the POTS line? I think its called FSK signaling. Anyone know

I thought I'd share my Asterisk experience, which hasn't exactly been as pleasant as I would like but now seems usable in most ways and more then I expected in other ways. I wanted a home PBX system, that would let me treat different callers different ways depending on CID. I initially bought the Digium developer's kit to try things out. That's a single port TDM400 and a X101P.

Hello, I'd like to add rules (level workstation) for ignoring wlan reconnects/disconnects. The rules would ignore messages like this: May 12 11:02:16 t400 kernel: [13254.204101] No probe response from AP <MAC> after 500ms, disconnecting. May 12 11:02:16 t400 kernel: [13254.213029] cfg80211: Calling CRDA to update world regulatory domain May 12 11:02:16 t400 wpa_supplicant[3277]:

Hi guys, I wanted to fetch some info from webpage and use it in my db. I read hpricot is one way to do it. But the problem is I''m unable to use it with my app. For some reason after successfully installing hpricot it fails when I try to use it in the rails console. I''ve been trying to fix this for the last four-five days. I''m stumped ! I tried re-installing everything

I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

Hi all... I'm sorry for repeating my message. I have a problem with caller id on my asterisk server with xorcom astribank. here is my configuration : centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2 ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting) everything fine until I try to feed my app with caller id. here is the log : --