Displaying 20 results from an estimated 700 matches similar to: "outgoing call routing"
2005 Jun 18
2
Unable to make outbound calls
Hi All,
I am a new bee to *. I just installed Asterisk@home on
FC3. I hv a FXO card. I hv configured two extensions
one x-lite and other iaxComm. I configured * using
AMP. The following setup works
- x-lite (x 200) to iaxComm (x 201)
- PSTN to x-lite
- PSTN to iaxComm
Voice mail, weather etc work fine.
When i try to make an external call i am getting
message "All routes are busy". In
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
2005 Sep 14
2
PRI to PRI passthrough with DID intact
I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink
T1 ---- Asterisk.
I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk
extensions over the T1.
I do not get DID nor CID on the Asterisk, so I want to use PRI between the
PBXs.
I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are
different cards)
I see this as my least
2005 Aug 09
3
SIP-Trunk problem, Please help!!!
Hi,
We are using VOIP-SIP gateway to route outbound PSTN calls.
Recently, I am getting == No one is available to answer at this time
message, after making 5 SIP attempts (Retransmitting #5 (no NAT):),
and the calls are going out through alternate Zap-trunk.
I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls.
Strange thing is that this is happening randomly,
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 =
2007 May 17
2
Quadbri Cellular Issue
Hello everybody, and first of all sorry for my poor English.
I'm having trouble with Quadbri installed on Asterisk
1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling
to switched off or "out of coverage" cell phones. In this case I have to
wait 40 seconds until Asterisk tell me that "all circuits are busy now"
instead of receive cell phone
2005 May 12
1
chan_capi and chan_misdn
Could someone please comment on the current state of chan_capi,
chan_misdn and chan_modem channel drivers in terms of functionality and
stability. Specifically, which channel driver would be best for a
passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that
chan_capi distinguishes between junghanns and non-junghans cards, and
that chan_misdn is better suited for general misdn
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
2007 Jan 16
2
IAX2 softphones can't (won't?) use PRI trunks....
I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons
that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give
them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3.
svn rev 47264.
I've appended a sample call trace. The
2006 Mar 31
4
cannot set outgoing cid
Hi,
sorry for the long debug output below. I configured Asterisk with AMP to send
the whole number including the extensions of the callers to the called party.
Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but
doesn't seem to work.
033811234451 is the call id i configured, and it seems to use them, but the
caller will only see a 0338189040 instead of my
2006 May 26
0
No sound when the call is diverted
Hi Guys,
I'm having sound problems when diverting a call using asterisk@home 1.5. I
am using the following configuration in extensions_custom.conf,
extensions_additional.conf and extensions.conf
[custom-Sales]
exten => s,1,SetVar(DivertNumber=02XXXXXXXX)
exten => s,2,Dial(SIP/116, 15)
exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1)
(i have replaced the diverted phone
2005 Jun 17
0
No ringing tone on outgoing SIP trunk
Hi!
I have configured a SIP trunk with a dialing rule.
The trunk behaves normally for incoming calls but when in used for
outgoing call a strange thing happens.
When I place a call I cannot hear the tone confirming that the remote
phone is ringing. I simply hear the voice as soon as the party picks up.
When the remote phone start ringing Asterisk receives a SIP packet
stating that the call is
2005 Sep 16
0
Unable to create ZAP channel - All circuits are busy
Hello,
I have *@Home 1.5 installed and all is working fine for incoming calls and
sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO
Ports)
setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that
order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2.
When i try to dial out to the PSTN from a SIP phone it sometimes works
(normally after a reboot)
2005 May 12
0
Cellsocket with @home
I am posting this in case someone need help..
=========================================================
YOU THA MAN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
No sure how I will repay you, but anything you need, just let me
know!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
Thank you, thank you, thank you!!!!!!!!
-- Executing GotoIf("SIP/2007-12c7", "0?4") in new stack
2005 May 26
0
capi dial in/out configuration
Hi all,
I've recentrly starting to play around with *, when all I wanted is to
configure an fritz ISDN card with A@H.
Currently I'm stuck at the phase of what do I do with capi after
everything is installed.
I'm trying to understand how to setup incoming and outgoing calls at A@H
since I'm getting a bit lost with the default dial plan.
It seems that * answers but disconnect
2005 Jul 05
1
Newbie question reg. Asterisk and Channel Access Bank I and TE110p
Hi,
I have some problem to get this setup working. I have a CAC Channel
Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2)
and I have a TE110p installed in this box.
What I want to do is, just to be able to dial one of those lines that
already are connected to the channel bank, and transfer that call
through TE110p and Asterisk to a user agent somewhere through
Internet.
2009 May 08
2
Configuring SIP Trunk
Hi All,
I have searched the various post and not able to find the solution.
I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same.
When i try from softphone i got error as "Call rejected" and in the asterisk i got error as
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
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
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