Displaying 20 results from an estimated 1100 matches similar to: "Asterisk-Users Digest, Vol 22, Issue 1"
2006 Apr 30
2
PRI Issue: D-Channel woes
Hi,
I am about to pull my hair out after trying to get our PRI up and working.
We are switching from a Cisco gateway to an Asterisk box which provides
the 23 phone lines for our office. So, because the Cisco gateway is
working I can assume I have all the settings right (b8zs, esf, dms100,
etc) and the PRI is live (because we are switching over). When dialing
from PSTN, I get busy signal. When
2005 Aug 23
1
Asterisk 1.0.9: TE411P replacement for TE410P 1stgen causes crashes
Hi all,
I replaced a TE410P Rev C 1st Generation Firmware with a TE411P
without any cfg changes (zaptel/zapata).
As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls:
Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a
UA, but i'm in state 1
Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a
UA, but i'm in state 1
2004 Jun 16
0
(no subject)
Hello!
We are using the Digium 405PP card, and getting the following messages:
Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event:
6 on Primary D-channel of span 1
Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event:
8 on Primary D-channel of span 1
My config file is below. We are trying to set up D-Channel on channel 24,
1-23 in trunk group 1,
2005 Feb 11
2
Question about DID
Hello Group
I have a Asterisk server running with 2 Digium T1 cards installed. 1
card connects to Telco via a PRI. The 2nd card is connected to a fax
server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to
have Asterisk route the calls based on DID or FAX tones. Everything is
working great so far. The only problem is the Fax server does not see
the DID. How can I tell if Asterisk
2004 Dec 03
3
Two zaptel T1 cards: no clock from one
List,
I have a TE410P (T1 mode, all PRI) and a T100P (fxoks, for fxs channel
bank). I cannot seem to get the T100P to send any clock to the
channel bank. I prefer that it use the same clock source as the
TE410P, but it doesn't matter if it's not in sync just as long as it's
there.
The TE410P is configured 3x pri_cpe, 1x pri_net. The three cpe go to
XO Sonus switch, the net to
2006 Feb 09
0
re: Polycom IP501 with Asterisk - distinctive ring
The answer is yes, I think, but I don't recall precisely how off the top of my head, and I'm walking out the door in a moment. The phone will hold more than a dozen distinct ring tones which you can create for yourself, and you can have asterisk direct it to use a ring tone independently of line appearance. The most direct way would be with the SIP Alert-Info field, but the phone itself
2003 Nov 07
0
RE: Asterisk-Users digest, Vol 1 #1808 - 13 msgs archives gsm of asterisk ???
Hello.
The procedure so that it works you can find in:
http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris
k
a the files .wav
chmod 755 file.wav
sox file.wav -r 8000 file.gsm resample -ql
chmod 755 file.gsm
in extensions.conf
xxxx=> xxx,x,playback(file)
Ing Javier Rios
Ing de Proyectos
04167285748
212 2637246 /2637187
-----Original Message-----
From:
2006 May 13
0
Spam? Re: Cisco 7970 problems
I don't see that anywhere. Here is my zapata.conf This is only happing
on my 7970 all other phone are working without trouble.
[channels]
context=pri
signalling=pri_cpe
switchtype=dms100
group=1
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=incoming
musiconhold=default
echocancel=yes
2005 Sep 28
1
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unknown signalling method 'pri_net'
Any ideas?
51] logger.c: [chan_zap.so] => (Zapata Telephony)
Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found
Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing
2005 Sep 28
2
Sep 28 00:42:35 ERROR[5151] chan_zap.c: Unkn own signalling method 'pri_net'
Did you compile and install libpri *before* Asterisk? I had same problem
(among others) b/c I didn't install in the correct order. Try the awesome
asterisk_update.sh shell script.
Are you trying to emulate CPE or NET? Try signalling=pri_cpe
Check for whitespace behind the statement, zapata.conf seems bitchy about
whitespace.
hth
-----Original Message-----
From: Steve Totaro
2004 Nov 28
4
PRI Dialing failure?
So I reached the point where my PRI is accepting incoming calls, but I
cannot dialout. I must be doing something stupid, but I can't figure it
out. The Asterisk box is sitting between the Mitel and the phone company,
and has PRI lines to each. Asterisk was built from CVS r1-0
Log for a call from mitel heading outbound:
-------------------------
-- Accepting call from '' to
2005 Feb 10
0
Asterisk 1.0.5 won't pick up incoming calls
Hi All,
I have just migrated from Asterisk 1.0.0 to Asterisk
1.0.5 and I have an X100P installed. The old asterisk
was working, but now the new version isn't picking up
any calls! However, I did notice that after
installation, I performed modprobe zaptel and modprobe
wcfxo and they worked fine, but when I executed ztcfg,
I get the following errors:
ioctl(ZT_LOADZONE) failed: Invalid
2004 Sep 16
2
No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS?
I have a PRI link up and running between Asterisk and a Nortel Norstar MICS
v4.1 . I'm having a problem getting the textual Caller Name across the link
from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns
to Ast both elements come through fine. I'm forcing dummy values for testing
using:
exten => s,1,SetCIDName(Test)
exten => s,2,SetCallerID(1234561234)
2009 Jan 30
0
Can't hear audio when Playback(something, noanswer) on Zap
Hi
I have this escenario:
|SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR,
A2Billing, etc...
The problem is that I can not hear any audio when call from 'sip or H323
phone' and configure something like: exten =>
_01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ...
It works if I remove the 'noanswer' parameter but in this case it connects
2006 Feb 09
0
Busy problem
Hello,
I have a busy problem with Asterisk when I try to transfer a call from PRI
directly to IVR.
This problem appear sometime after 2 hours or 2 minutes.
The log file contain :
Unable to create channel of type 'Zap' (cause 34 - Circuit/channel
congestion)
When this problem appear I must restart Asterisk to solve it.
Another thing, I don't know why the alarm is set to NOP on SPAN
2004 Aug 31
4
T100P No D-channels
Hi
Last week I installed Asterisk (release1) with digium t100p single span T1
(wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is
currently the only user of the t1 line. All worked well for about a half a
day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I
consistently get multitudes of blue alarms on all b-channels followed by a
loss of d-channel:
Aug 31
2003 Dec 08
1
Re: Asterisk-Users digest, Vol 1 #2120 - 14 msgs
In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger:
Andrew,
I modified the exten line in extensions.conf as you suggested.
Unfortunately,
It still does not work...
Ernest,
I spent approx. 4 hours reading list archives (and anything else Google
served up) on
how to configure iax.conf and extensions.conf to work with Voicepulse.
Then, I sent
an email to voicepulse
2003 Nov 20
2
TE410P ERRORS under load
Hi all-
HELP!
This is actually a revisit of a problem that I had earlier with E400P's at a
customer site. Customer still gets locked up channel problem, but has
learned to live with it (channels clear themselves after several minutes).
The symptoms, which I believe are directly related:
I'm having problems with tons of framing and "read" errors on my E1
connections (and
2004 Nov 27
1
Interfacing T100P with Definity PBX
Hi,
I have two t100p cards installed. One card connected to the PSTN,
the other card is hooked to the Definity PBX with a T1 cross over
cable. The card connected to the PSTN works fine.
The problem is with the card hooked up to the PBX. The light behind
the card is green, but when I dial from this card I get an error. Its
says it could start the D-Channel. Could someone help me out with this
2006 Mar 06
1
PRI CID signalling not working?
Hello - I have (finally) obtained a fair amount of success in
connecting my Intertel Axxess PBX to an asterisk box via a T1 PRI. I
can place calls from the Intertel side through the T1, out to an IAX2
softphone and the calls get routed correctly and all of the CID
information stays intact. However, when I call from the IAX side to
an extension which should route back through to the Intertel