Displaying 20 results from an estimated 8000 matches similar to: "call not routed"
2006 Apr 30
1
newbie-too much latency
I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS.
The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log :
====
Apr 30 10:26:50 DEBUG[3050] manager.c: Manager received command 'Command'
Apr 30
2010 Aug 26
1
Timecondition fallthrough on 2nd GSM Modem, First modem and ZAP's are all fine
Hello,
we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards.
No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt:
[Aug 26 11:04:36] VERBOSE[3112]
2006 Nov 08
1
Delay between DTMF Down & Detected Digit
Good Morning,
I've recently gotten Asterisk installed and configured our IVR using
FreePBX. Things seem to be going well except a few of our inbound
callers are ending up in the wrong place when trying to connect to a
specific extension. The example I had this morning was someone trying to
call extension 212 and getting connected to the Sales queue which is
option 2 on the IVR. I looked in
2008 Jul 08
0
Trouble with faxing using iaxmodem / hylafax
Hi all,
I have just setup a trixbox system and I am implementing
hylafax/iaxmodem solution for the faxing.
When i send a fax to it by phoning in listening to the IVR and manually
pressing start to initate the fax, the call gets picked up correctly as
a fax and everything works well.
When I try sending a fax by entering the phone number and pressing start
to initiate dialing it sounds like
2006 May 02
0
Telasip config problem/question
I seem to be getting a connection from telasip but instead of dialing my
default extension, nothing happens. I listen to dead air.
I have a fxo card configured and working on both inbound and outbound
calls. Telasip is working outbound. I put in the recommended (by telasip)
changes to the trunk for incoming, e.g.
host=gw4.telasip.com
insecure=very
qualify=yes
type=user
context=from-pstn
Then
2007 Dec 22
0
Dead Incoming call - Sangoma A200
Hello List,
I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up.
A bit of background:
The client actually has two systems install (one at
2006 Nov 05
1
asterisk DTMF detection
Hi,
Hi All,
I've just delved into the world of asterisk and I'm having a few dtmf issues.
Internally, amongst sip phones, dtmf is fine.
Externally, if you ring from a GSM mobile, DTMF is fine, however if
you ring from a standard phone, DTMF fails to register.
I am attempting to use a quad port HFC-4S Beronet Card. I've been
searching the web most of the last week and
2013 Jun 24
2
Asterisk-11 loop behaviour
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4.0
FreePBX = 2.11.0.2
Snom870 Handsets
We are in the process of moving to an Asterisk based PBX. At the
moment most things work as we wish. However, I have just notices that
when I force a reload using 'amportal a reload' I see this loop start
in 'asterisk -rvvvvvvvvvv':
> Channel Local/s at tc-maint-000002a4;1
2006 Feb 16
1
Problem making outbound calls on TE210P using NFAS
Hello,
I'm running Asterisk@home 2.5
asterisk 1.2.4
zapatel 1.2.2
libpri 1.2.2
on a Dell Poweredge 2850 (1 CPU) with a TE210P
I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound
calls on all channels and can only make outbound calls on channels 25-48.
Attempting to make an outbound call on channels 1-23 results in congestion.
2005 Mar 10
2
NVFaxDetect errors on make
Hi All,
I am trying to add FAX to my SIP confiig and I am getting some errors, any help would be great.
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\"CVS-v1-0-12/23/04-22:36:11\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"
1999 Jul 14
0
Samba or mutt problem
Hi,
I read about your samba vs mutt problem on the samba mailing list.
Unfortunately I accidently deleted the digest before replying so I can't
include a copy of your original question. I don't have a good answer to
your original question, but have an alternative. The answer, if you want
to try something new, is IMAP. This is an alternative mail client
protocol to the standard POP3
2005 Aug 03
0
fax <--> grandstream 286 <--> asterisk <--> pstn
Hi all,
Im having problems using a fax machine conected trough a grandstream
286 sip ATA, it must be able to send and recive fax from pstn, but fax
always ends with communication errors 252/244/232 and others.
Im using alaw/ulaw codes on pass trough mode, also have tried asterisk
faxdetection, nvfaxdetect, disable echo cancellation by hand always
with same results.
Grandstream ATA is using
2005 Jun 21
5
NVFaxdetect
I have googled this and come up empty. Has anyone had any problems
compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am getting
when I run make.
app_nv_faxdetect.c: In function `nv_detectfax_exec':
app_nv_faxdetect.c:210: error: structure has no member named `cid'
app_nv_faxdetect.c:227: error: structure has no member named `cid'
app_nv_faxdetect.c:265: error:
2006 Mar 08
0
Random Zap port going crazy When channel released after a flash.
On 1.2.x I have a random problem when a Zap/x channel flashes to transfer
or make a three way call.
The Zap/x-2 channel is created and the transfer or three way proceeds, but
on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk
goes crazy logging the problem. Here is an example debug log.
This happens only once a day or so, with 100 or so users transfering and
three
2007 Jan 05
2
chan_zap.c: Failed to read gains: Invalid argument
I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P
card in E1 mode. I've recently noticed in my logs the following
Jan 5 01:27:11 VERBOSE[22490] logger.c: [chan_zap.so]Jan 5 01:27:11
VERBOSE[22490] logger.c: [chan_zap.so] => (Zapata Telephony w/PRI)
Jan 5 01:27:11 VERBOSE[22490] logger.c: == Parsing
'/etc/asterisk/zapata.conf': Jan 5 01:27:11
2006 Mar 15
3
Failed to read gains: Invalid argument
Hello,
When I start Asterisk, I get the following in my log (/var/log/asterisk/full):
Mar 15 17:16:55 VERBOSE[4242] logger.c: == Parsing '/etc/asterisk/zapata.conf': Mar 15 17:16:55 VERBOSE[4242] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found
Mar 15 17:16:55 DEBUG[4242] chan_zap.c: Failed to read gains: Invalid argument
Mar 15 17:16:55 DEBUG[4242]
2006 Jun 08
1
zap calls drop suddenly + tremendous noise when answering a call
We have an asterisk box with the following configuration:
- AMD Athlon XP 2400+
- 512 MB RAM
- SUSE Linux 10.1
- a Digium card TDM400P with 3 FXO
- another Digium card TDM400P with 4 FXS
- asterisk 1.2.7.1
- zaptel 1.2.4
I already checked that those cards aren't sharing interrupts (by cat
/proc/interrupts):
0: 14119786 XT-PIC timer
1: 10 XT-PIC i8042
2:
2008 Mar 28
1
PRI error cause hangup calls
Dear all,
When I make a call using my PRI line, all goes well, but suddently the
call hangs up.
I searched the asterisk logs, and I found that.
Write to 55 failed: Unknown error 500
Short write: 0/15 (Unknown error 500)
What does this mean?
Why this occurs?
How could I solve that?
Someone could tell me if it was a primary error (the primary shows red
alert in all its channels) or it could be a
2008 Dec 18
3
Asterisk AGX addons compile issues
Has anyone seen this before, and know what is happening?
USER at HOST:~/asterisk/agx-ast-addons# ./build.sh
-- Configuring done
-- Generating done
-- Build files have been written to: /root/asterisk/agx-ast-addons
[ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o
Linking C shared module dist/app_devstate.so
[ 11%] Built target app_devstate
[ 22%] Building C object
2008 Feb 04
0
PRI ISSUE
hello everyone,
Last week I installed asterisk 1.2.24 with digium TE220B card. I have a problem with our PRI and Asterisk: the call be interrupted.It happens either PSTN-to-SIP or SIP-to-SIP,almost every call.
After spending several days searching on internet, I found a lot of
discussion about this issue and I have tried many,
but it still.I am totally new to Asterisk environment and suspect I