Displaying 20 results from an estimated 20000 matches similar to: "Mulig_SPAM: More than one outgoing call"
2004 Aug 25
1
Individual call-forwarding on ISDN
Hi,
I'm looking forward to install an asterisk-server to perform ISDN-to-SIP-
bridging. But for the times when I'm not available via SIP I also need call-
forwarding-features. Afaik it is possible to directly forward a call on ISDN
instead of opening a second ISDN-channel additional to the incoming-ISDN-
channel and forward the call "by hand", right?
What if I need features
2006 Nov 10
2
Outgoing problem on PRI
Dear All,
I have an asterisk server version 1.2.12.1 along with trixbox and I am
having this nasty problem, I have a TE200P and have an E1 pri attached
to it and zttool says it's OK, I have configured the whole 31 channels
into one group as follow:
Zapata-auto.conf:
callerid=asreceived
signalling=pri_cpe
switchtype=euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
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 May 26
1
Not able to make any calls
Hi All,
I have registered "abhijit" for SIP in asterisk Server.
I am able to register my softphone (SJPhone) to the server using the
name "abhijit".
But whenever I try to make any calls I am gettinh the following error
message:-
*CLI>
-- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120
May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:
2008 Jan 15
0
busy/congestion random
Hi, I use:
Trixbox-2.2.4
FreePBX-2.3.1.0
Asterisk-1.2.17
BRIstuffed-0.3.0-PRE-1y-e
Zaptel-1.2.19
..with two ISDN cards, often but occasionally the dial out failed but is
possible to receive external call.
My zapata.conf conf is:
[trunkgroups]
[channels]
language=it
context=from-pstn
signalling=bri_cpe_ptmp
rxwink=300
pridialplan=unknown
prilocaldialplan=local
switchtype=euroisdn
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:
2015 Mar 25
0
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
<salah.elharit200 at gmail.com> wrote:
> hello list,
>
> i have asterisk 11.15.0 and i have some trunks sip from my provider
>
> we have some ip phone astra 6731i
>
> each Ip-phone is configured with trunk and we call
>
> no ihave configured another trunk from the same provider in my asterisk
>
> i can call
2008 Apr 28
2
PRI hangup certain outgoing calls
I have a problem calling a certain number from our PRI line. Calling the
number from a separate PSTN phone works fine.
The remote number seems to have some funny call redivert setup, when you
call it, it answers immediately, makes some kind of beep and then starts
to ring.
Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing
calls work without a problem. The server is
2015 Mar 27
0
call between snom 300 and aastra 6731i
thank you for your response below the asterisk -vvvr
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192",
"user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192",
"TOUCH_MONITOR=1427481319.470") in new stack
--
2015 Mar 20
0
outbound calls
I am making some assumptions, but assuming the 217.195.xx.xxx is your
provider, you are getting this back from them:
"Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060"
Are you sure that "0033149xxxxxx" is the format the provider is expecting?
You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what
the INVITE looks like, but
2006 Feb 15
1
problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion)
Nik,
Looks like you're making some progress. When I first started using A@H
I had trouble getting the outbound dialing to work. I wasn't sure where
to start, so what I did was skip the macros in the dial plan. I wanted
to play around with exactly what digits the telco wanted to see. So I
put a specific extension in my [default] context like this:
exten =>
2015 Mar 20
0
outbound calls
thanks for your response
i noticed that when i active the voicemail in the IP-phone where the number
0033149xxxxxx is configured i can call this number without issue
the server asterisk and the ip-phone where the number is configured are in
the same network 192.168.1.X
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xxxxxx
== Begin MixMonitor Recording
2007 Jan 29
3
Pickup() ringing extension and call waiting
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
application.
from my dial plan:
[incoming]
exten => s,1,Dial(SIP/somebody1|60|tTrR)
[internal]
include => outbound-local
include => parkedcalls
2005 Aug 10
0
tdm400p / outbound zap prob
I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p
w/ 4 FXO. Incoming calls work fine, outbound I get this:
-- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack
-- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("SIP/231-af2b", "0?19") in new stack
2015 Jun 19
1
outgoing calls not working on sangoma A200
Hi.
I have 3 lines connected to a SangomaA200. All was working normally,
but 3 days ago we experiment this issue. The outside callers can
call us normally; but when we try to do an external call, just receive
the message "all circuits are busy".
Only happen with outgoing call. Incomming calls work fine.
if the conf was not modified, what can i check to search the reason of
that
2010 Mar 26
1
SIP/2.0 403 Forbidden
hi,all
when i send a call to other server use SIP trunk,
i got this below,
<--- SIP read from 222.46.18.52:5060 --->
SIP/2.0 403 Forbidden
what's rong with is?
> Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
> Created by Mark Spencer <markster at digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
2015 Mar 25
2
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
tnaks for your response but the number dialed exist and i can call this
number when i configure the trunk directly in x-lite and i call call also
this number from my cell phone .
any help
thanks and regards
2015-03-25 12:59 GMT+00:00 Matthew Jordan <mjordan at digium.com>:
> On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
> <salah.elharit200 at gmail.com> wrote:
> >
2007 Apr 11
0
Asterisk Pickup with more than one argument
Dear all,
I tried to use the following sintax to implement call pickup in Asterisk
1.2.17 with no success:
exten => _**5219/5215,1,Pickup(5219)
exten => _**5219/5215,2,Pickup(220408108)
exten => _**5219/5215,3,Hangup
Asterisk seems to just do the first priority command (Pickup(5219)) and
if the ringing call comes from the channel 220408108, it doesn't jump to
the second priority
2007 Aug 10
2
Ordering BRI From AT&T
Hello everyone,
I'm hoping someone can help me with this. I have a business customer in
the U.S. (Michigan, AT&T Territory).
I need to get 4 trunks into an asterisk Box. My intention is to use an
Eicon Diva Server card with 2 BRI Circuits. The reason for this is that
the business needs DID's on the trunks (20 of them). A full or fractional
PRI is overboard for them, as they will
2015 Mar 20
3
outbound calls
hello list
i have an issue related to outbound calls i can contact all the number
except on number given by our provider in trunk
the issue just when i configure my trunk in our server but when i configure
the trunk directly in x-lite i can contact this number without issue
below the cli
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0149xxxxxx at