Displaying 20 results from an estimated 900 matches similar to: "CallerID hosed"
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
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
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
2009 Jun 29
2
Calling Number Verification Number? for BellSouth/AT&T
BellSouth (now AT&T) has a number you can dial and it will play back
voice prompts with your calling number? It's used by their techs with a
buttset in identifying analog 1FB lines...
Eg Dial 704-210-3233, it answers
"Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two" if your dialing from
704-559-2122
-----------------------------------------
Disclaimer:
This e-mail
2005 Aug 23
1
Wait before dialing ( was Pause during dialing to enter another number)
Started a new thread as my problem is somewhat different than the OP.
Seems his somewhat different problem doesn't work as advertised either.
Eric Wieling wrote:
> I don't know what the problem is, but this is what I use and it works
on my analog FXO port.
> exten => _9NXXNXXXXXX,1,Dial(${PSTN}/w${EXTEN:1})
So, I modified slightly to fit my dialplan:
exten =>
2006 Jun 06
1
Problem with simple incoming calls
Hi all,
I must admit that I am stuck. I have a TDM400P card with two FXS and
two FXO modules which I had set up and configured so that it was
working beautifully. The only problem was that occasionally it would
get itself into a state where outgoing calls would simply be met with
a very loud static. A reboot would fix this issue and everything
would work fine for a while.
Recently however,
2003 Mar 06
2
Dial Problem
I have a simple problem with sialing a SIP device. I'm SURE
it's a syntax problem, but I dunno what it might be.
Here are the debug messages:
== Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>)
-- Executing Goto("Zap/1-1", "2111|1") in new stack
-- Goto (default,2111,1)
-- Executing Dial("Zap/1-1",
2003 Sep 13
5
Voicemail to a commercial PBX/key phone system
Hello.
I've seen some mentions of asterisk possibly being used as an inexpensive
voicemail attachment to a commercial PBX etc.
Does anyone here, have experience of using it in this fashion ?
What commercial systems have been successfully attached too ?
How is the attachment made ?
Analog, digital ?
If anyone has successfully accomplished this, I would like to hear the make
and model of
2003 Mar 06
1
Cisco SIP Weirdness (1750, not ATA)
I have the following in extentions.conf:
exten => 2111,1,Dial(SIP/2111 at gw1.langley)
exten => 2111,2,Voicemail(u2111)
exten => 2111,3,Hangup
exten => 2111,100,Voicemail(b2111)
exten => 2111,101,Hangup
I have the following in sip.conf:
; Cisco 1750
[gw1.langley]
type=friend
host=172.16.17.1
context=default
canreinvite=no
Like the ATA, lots of stuff doesn't work on the 1750
2003 Mar 07
2
Interesting VoIP device
There was some talk a couple of weeks ago about voip devices w/ ethernet
and fxo/fxs configurations. I found this while out and about and
thought it might be interesting.
Configurable fxo/fxs with dual ethernet.
http://www.tekdigitel.com/website/htmlPages/content/products/product_introductions/introduction_to_V-SERVER_iGATE_Dual_Ethernet.htm
--
Karl Putland <karl at
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
2003 Apr 07
1
chan_local segfault
Happened twice. There might also be a race condition and some bad
pointers in chan_local.locals_show.
First the segfault.
CLI> show locals
<unowned> -- 6001@default
Segmentation fault (core dumped)
[root@mars asterisk]# ll -tr
total 22260
[...]
Loaded symbols for /usr/lib/asterisk/modules/chan_local.so
#0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99
99 mutex.c: No such file
2005 Aug 12
1
Weird issues with TDM400P
We have a TDM400P installed here with four FXS modules. It works well
except for a couple of issues:
First, I have a Panasonic KX-TG2431 telephone (so others can reach me
when I am in o ther parts of the building) hooked up to one of the FXS
ports. When the other end hangs up, I get the usual CPC disconnect
signal. After the CPC, sometimes it will go to a dialtone, and other
times a
2005 Jun 29
5
Problems with OR Logic in the GotoIf Statement
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2006 Jun 13
1
calleridname.agi patch to only overwrite name if it is missing
I edited the calleridname.agi patch to only overwrite the name if it is missing.
The asteridex option still overwrites the name since it is our master list for known numbers.
--
Steven
calleridname.agi.patch:
--- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006
+++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006
@@ -16,6
2003 Apr 22
0
dlink updates
For those of you that have dlink VoIP hardware, I thought I'd post this
link. It's a whole lot easier that navigating their support site trying
to find anything. There is updated firmware for most of their products.
ftp://ftp.dlink.com/VoIP/
A word of warning about the DG_104S firmware. It's better than the
firmware that came with my box, but it still has quirks. Some settings
2003 Jun 06
0
Colorado Asterisk Users
Just a quick query to find out if there are any other Asterisk users in
Colorado.
If you're out there, drop me a line off-list. I'd like to start a user
group if there is anyone else out there.
--Karl
--
Karl Putland <karl@putland.linux-site.net>
2003 Jun 26
0
mec3 experiment
I was fooling around with echo cans today.
I found that I could reproduce this problem regularly
t400p
adit600 for fxs
mec3 enabled
Two speaker phones side by side within about 8" of each other.
Put phone 1 offhook with speakerphone
Dial phone 2
Phone 2 rings and is loud enough to be picked up by the mic in phone 1
By the second ring the echocan thinks it's figured things out, but the
2003 Dec 17
0
issue recording files in wav49 from AGI
Following is a log from an attempt to record and playback a file in
wav49 format from an AGI script.
COMMAND: stream file aa/after_the_tone "" 0
RESULT_LINE: 200 result=0 endpos=41920
RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')}
COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 "#" 20000 0
2005 May 25
2
Manager and Callerid problems
Guys.
Anybody knows why this is happening? Seems every time I make an internal
call, the manager shows this and I don't get the callerid on my identapop
but rather the calledid..
Event: Dial
Privilege: call,all
Source: SIP/intruder1-85f0
Destination: SIP/test-f037
CallerID: 201
CallerIDName: Anton Krall
SrcUniqueID: 1117038116.7
DestUniqueID: 1117038116.8
Event: Newchannel
Privilege: