Displaying 20 results from an estimated 10000 matches similar to: "cannot set outgoing cid"
2006 Mar 30
3
asterisk doesn't wait for whole extension
Hi,
maybe a dumb question, but it seems that some calls are directed to our
central dial in number despite the extensions the callers say they dialled.
E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown
extension, where it is right, and redirects the call to the central dial in
extension 1234-0. This only seems to happen when the numbers are dialled
manually. When
2006 Feb 13
0
problem with outgoing calls Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)
hi
i've configured a TE205P on asterisk at home
this is my
zaptel.conf
span=1,0,0,ccs,hdb3,crc4,yellow
span=2,0,0,ccs,hdb3,crc4,yellow
bchan = 1-15, 17-31
dchan = 16
bchan = 32-46,48-62
dchan = 47
loadzone = it
defaultzone = it
and my zapata.conf
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
2006 Feb 22
0
Outbound problem sip chanel
I setup my aah box with a sip trunk at irisxa.iristel.net
Incaming it is ok but when I try to dial 8 and the nr where I want to call I
get all line is busy.
In my log I have these:
Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command'
Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command'
Feb 22 14:33:19 VERBOSE[2721] logger.c: --
2006 Feb 13
1
problem with outgoing calls Unable to createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)
Nik,
Just curious - what is your telco setup? Do you have PRI with the
specified D channels? You need to make sure that your telco is set up
to have the D channels on 16 and 47. When you first start Asterisk, or
when you log on to the CLI, do you ever see messages stating the B
channels are successfully started?
Let us know.
-MC
-----Original Message-----
From:
2010 Jun 21
1
How to tell if a dropped call is my fault
I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.
2006 Feb 21
1
Outbound Routing does not use Multiple Trunks
Hello,
I have a TDM400 and currently have 2 of the ZAP Trunks configured
on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4
with AMP version 1.10.010
In my Outbound Routing I have the Trunk Sequence set up so that 0 is
Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is
full, it does not open Trunk Sequence 1. I have found that this is true
even if I
2006 Feb 06
12
Cisco 2620 as PRI gateway
I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make
this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like
it should bee useful for something!
I'm perfectly happy to do my homework, but also don't feel thee need to
reinvent the wheel! So, links with relevant info would be appreciated. If
there is a config for a 2621 being used as a gateway
2006 Apr 05
2
can't start chan_capi with asterisk group
Hello,
While upgrading * from 1.0.9 to 1.2.5, I have installed chan-capi-head
and I can't start asterisk under asterisk group
"asterisk -vvvvgc -U asterisk " and "asterisk -vvvvgc -U asterisk -G
dialout" work well but "asterisk -vvvvgc -U asterisk -G asterisk" fail.
I am thinking about a group permission configuration but I have exactly
the same one
2006 Apr 07
1
Inbound PRI calls drop after 5 seconds using Sangoma A101
Hi Folks,
I'm have Asterisk version 1.2.1 with a A101 PRI card. I'm working with the
CLEC to bring up the PRI and inbound calls are hanging up at his end after
a few seconds. I ran PRI debug but it only gives me minimal insight.
" Ext: 1 Cause: Unknown (16), class = Normal Event (1)"
He ran a trace and the only difference he is seeing is a
"ISDN interface explicitly
2010 Jan 05
5
CallerID on Indian PSTN is not working.
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing "Unknown" when there is an incoming call. I think the
same problem listed here: https://issues.asterisk.org/view.php?id=6683
There is one patch on this link but i don't know how to apply patch on
asterisknow.
2010 Jun 11
2
Call ended after 31 seconds
Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call.
This is the log, but I've not been able to find something wrong...
Any ideas?
[Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf
[Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [s at macro-dialout-trunk:16]
2009 Mar 17
0
Weird issue with outbound calls and MOH
Hi,
We have a PRI Trunk (physical E1) and we are getting
some rather weird and very isolocated problems. On outbound calls to
specific numbers, it would seem to me that DTMF from the remote side is
affecting the local asterisk system. Basically what happens:
- We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System
- Remote Answers, and converse
- Remote sends DTMF on their site to
2006 Jan 10
4
Help with amportal: asterisk ended with exit status 127
Greetings. I am trying to get AMP up and going on my Asterisk server. I can
access the admin pages on my asterisk server via a web browser. I can add
and edit things via the web browser and it edits the database accordingly.
Everything seems fine except when I try to run 'amportal start'. Below is
what I get (Plus tail /var/log/asterisk/full, but the tail of the 'full' log
2005 Jan 17
1
Attempting native bridge
ERROR CONDITION
---------------
-- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered SIP/2001-f6c4
-- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead
Have searched web and archive w/o good results.
Thks in advance for any help,
Dave
sip.conf
--------
[general]
port =
2009 Feb 05
2
TDM400P Circuit/channel congestion problem
Hello,
I have an issue with Digium TDM 400 card series. When I try to make
outgoing call (PSTN call) for example, the Zap channel could not be
created and busy channel message appeared. Below is the full log :
[Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro-
dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|")
in new stack
[Feb
2006 Jan 06
5
3RD REQUEST - Any Help Is Appreciated
Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know".
My third repost:
Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has
2006 Jan 05
1
Bizarre Answering Behavior
Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened.
I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring
2008 Feb 25
2
cannot dial out with latest zaptel and kernel 2.6.24
Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4
(latest as of today) and I cannot dial out using my 400P card with one
fxs module and one fxo module. I am using kernel 2.6.24 and get the
following log entries:
[Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [s at macro-dialout-trunk:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack
[Feb 25
2006 Mar 10
0
Voice Mail woe
Hi
i have installed AAH 2.6 and configured some extensions
the calls are working fine. but if i dont answer a call then
it says " the person at extension " and hangs up .
it doesnt spell out the extesion number nor it goes to voice mail box.
*************************** Asterisk CLI log ****************************
dialparties.agi: Extension 200 is available...skipping checks
--
2006 Jan 05
0
Bizarre Answering Problem - 2ND REQUEST
Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened.
I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring