Displaying 20 results from an estimated 600 matches similar to: "chan_zap.c - echo cancelation getting in the way of dialing????"
2004 Nov 29
1
Outbound E&M?
I've got a new setup (different building) where Asterisk is sitting
between the PBX and phone company on a E&M T1 line.
Mitel PBX <-> Asterisk <-> Phone company
Inbound works. Asterisk gets the in-band digits from the phone company
and hands the call off to the Mitel just fine.
Outbound is weird. Asterisk seems to expect that the mitel will send
routing information
2005 Jun 01
1
rxfax problems - cont.
Well, my faxes passes through asterisk successfully, however I still have
some problems about fax reception by rxfax.
The softfax answers, and negotiates transmission, however then as some stage
of communiation something is wrong.
But I have nothing more but this log:
Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on
Zap/10-1
Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242
2003 Dec 16
1
DISA - Zap/DTMF Problem
Hi guys,
I am trying to use DISA. The scenario is - I call my home number (where
X100P seats) from mobile phone, enter the password, enter international
number and get connected via voiptel. It works perfectly when I call
extension setup with DISA from X-PRO SIP phone, but when I dial into
Zap, It seems that it does not detect DTMF tones. Here is a log and
config files
Please help
2004 Dec 22
2
txfax failure
Hi list,
Just installed spandsp. In my limiting testing, I have an issue on a
Philips fax machine (HFC21) directly connected to my * server through
TDM400, reception with rxfax works fine, but txfax always fails. Below
is a transcript of failed transmit.
This is with asterisk-1.0.3 (with native moh patch but I don't think it
is the source of the problem). I already tried libtiff 3.5.7,
2004 Oct 05
1
difference between dtmf digit 8 and 9
Hello,
this is an example extensions.conf.
[default]
exten => 500,1,Answer
exten => 8,1,SetGlobalVar(firstdigit=8)
exten => 8,2,Goto(process,s,1)
exten => 9,1,SetGlobalVar(firstdigit=9)
exten => 9,2,Goto(process,s,1)
I call extension 500 and send dtmf digit 9. This is printed to the
CLI:
-- Executing Answer("Zap/20-1", "") in new stack
-- Accepting
2003 Nov 18
0
Bad DTMF detection
We're still having problems with DTMF detection on our X100P cards.
Incoming callers that hold down the "1" button for too long are being
connected to extension 11. One would think fat fingers were uncommon,
but it happens to alot of people.
I suspected this was related to our having to increase the txgain, but I
tried turning it down with no effect. I also tried disabling the
2005 Jun 28
2
Trying to get *8 call pickup to work
I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When
I call from a zap channel or a SIP phone to another SIP phone, then dial
*8 from a third SIP phone, I get 503 Service Unavailable on the
third phone and I get this at the Asterisk console:
Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible...
Jun 28 09:01:24 NOTICE[16774]:
2005 Sep 13
1
wctdm, issue w/outbound calls
Hi all,
I've been running Asterisk with a TDM400P for about 6months, no problems.
2 in/outgoing analog lines, one analog phone. Recently I was messing with
the XTEN client, got to finagling with things, and not knowing what was
wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was
testing various things, and found everything worked except outgoing calls.
So I checked
2010 Aug 04
1
Asterisk not working with Festival
Hello,
I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk
1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine
with SIP channels without Festival. I have written following context in
extension.conf:
[connect-to-me]
exten => s,1,Answer
Exten => s,n,SayDigits(?1?)
exten => s,n,Festival(hello john)
exten => s,n,Hangup
I use call files to
2003 May 22
2
new DTMF tones
I just loaded from CVS this afternoon and in the debug output I see...
DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: m on Zap/16-1
DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: u on Zap/16-1
I knew about DTMF 0-9, A-D, *, and #, but I didn't know about m and u :-).
2004 Aug 12
1
AgentLogin issue
Hi
i have an issue getting agentLogin working
/etc/asterisk/queues.conf
member => Agent/1001
member => Agent/1002
extension.conf
exten => 110,1,Wait,1
exten => 110,2,AgentLogin()
now, i call 110 by a firefly client, trying to login in as 1001 agent:
Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060>
--
2003 Sep 03
8
Asterisk Jitters
Hi,
Every time I dial into my asterisk box i hear nothing but asterisk
jittering.
The following is an example of what I get on the asterisk CLI
Thanks
*CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT
on RTP
to 0
DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res
DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user
2006 May 26
0
SIP call problem
Hello,
I have problem to make SIP calls, i have asterisk +
PC InterP4 + Digium TDM400P
here is the content of the sip.conf:
[SIP_PROVIDER]
type=peer
fromuser=testcomclient
username=testcomclient
secret=testr
host=IP_SIP_PROVIDER
;dtmfmode=rfc2833
context=interne
canreinvite=no
;allerid=Beer
disallow=all
allow=ulaw
allow=gsm
allow=g723.1 ; Asterisk only
2006 Feb 19
2
spandsp 0.0.2pre25
Hello,
Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or
1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax,
and it builds, but I'm not having any luck getting it working. 99% of my test
faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate.
I've bumped the console debugging level in logger.conf to include debug
2004 Aug 25
3
Fax detect
I have found that fax detection is returning an error saying that no fax
extension is present when I have defined one.
The console returns this error:
Aug 26 10:58:41 NOTICE[1112745536]: chan_zap.c:3989 zt_read: Fax detected,
but no fax extension
extensions.conf has:
[default]
exten => fax,1,Hangup
exten => fax,2,Congestion
exten => fax,102,Congestion
exten => f,1,Hangup
exten =>
2003 Dec 17
5
ALL incoming Zap channel calls are getting picked up as FAX calls!
All,
I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all*
of my incoming calls are coming up as FAXes. I had to disable my fax
extension because every call to my POTS line was getting redirected to my
FAX machine. After removing the FAX extension, if I call my POTS line from
my cell phone, I get the following:
*CLI> -- Starting simple switch on 'Zap/1-1'
2005 Jul 08
1
Help needed - Zap Transfer Failing...
Hi.
I have the following line in the default context of all my internal
extensions:
exten => 9876,1,Transfer(125)
When I dial extension 9876 from any sip phone, * dutifully transferrs it to
extension 125, which is just what I want.
Unfortunately when I dial 9786 from my Zap connected analogue phone, the
transfer doesn't go through and the dialplan drops through to a hangup.
debug
2003 Nov 18
4
Help with Warnings
I'm trying to clean up some notices/warnings that are repeatedly logged
in *.Any Help would be appreciated as I'm not sure of the cause
/solution.
Here are the errors:
Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321
(zt_call): cidspill already exists??
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
/* Don't send audio while on hook, until the call
2004 Sep 22
1
(euro)ISDN: complete silence / can't hear a word.
Hello,
I just got my isdn-card working together with i4l and asterisk.
Everything seems to be working fine: I can accept calls coming from the
outside and I can dial out. Even setting the msn works like charm but my
problem is that I cannot hear a word. There's complete silence in both
directions.
Any idea what could be the cause?
Thanks for your help,
Gunther
Lspci:
0000:01:07.0 Network
2005 Aug 08
0
Asterisk-to-IVR Problem
This was submitted to the Dev list last week, but there was no response, and
perhaps it wasn't the right group.
I am developing an application in which I need asterisk to pass on an
incoming call to a separate IVR server. The problem is that asterisk appears
to hang up while the IVR is playing back a sequence of recorded voice and
systhesized voice prompts.
My setup is:
Analog line