Displaying 20 results from an estimated 900 matches similar to: "Half Solved - Fail over to Pri on VoIP connection failure"
2006 Feb 09
0
re: voipjet -- Workaround if needed
Same thing here. I had this problem awhile ago and made this
workaround.
Going to another trunk does not work because they are answering and not
sending a error code. If you are using AAH code then this waits 10
seconds on your Voip then times out and goes to PSTN. You can modify
for your needs
The pertinent line is 14 in macro-dialout-trunk
I am going to clean it up and repost under my
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
2005 Aug 24
0
SIP trunk rollover problem
Hello,
I've got an Asterisk system with 3 SIP trunks configured. Each SIP
trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound
call routing (via AMP 1.10.007a) uses the 3 trunks in descending order,
all set with max channels to 4. Unfortunately, when the first trunk
reports a "480 Service Unavailable" (all ports in use), Asterisk reports
congestion without
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to
2006 Mar 17
0
Call transfer problems, SOLVED
Hi All, in regards to my previous queries about call transfers not working from inside, several days of searching turned up this posting:
I got this to work by editing the line
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf
seems like many people have had this issue in the past, I guess it's AMP related, as
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 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
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2006 Jan 26
6
Fail over to Pri on VoIP connection failure
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
Any point in the right direction would be great
Thanks,
CLI output (cleansed to protect the
2008 May 20
5
Server recommendation help
I am having a issues with adding a analog card to my dell 2800. I
already have a t1 card installed and running fine but when I install the
analog card asterisk will not start (ztcfg fails). I have determined it
is because of a IRQ problem and have decided to get a new server. Can
anyone suggest a server grade setup that supports this? I would rather
not buy a machine that will be unstable. I
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 Apr 20
0
Re: Asterisk-Users Digest, Vol 21, Issue 113
Hi List!!
Thanks for the colaboration, especially to Richard Cavanna who gave me the
necessary support.
I followed your indications and the comunication was better for the test
users. The warning indication is no jumping anymore and the voice is not
delayed. This is my sip.conf:
[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
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
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 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)
2007 Jan 25
1
IAX softphone fails through PRI trunks with Hangup
I've a call center using IAX softphones provided by a third party.
We've observed problems where the IAX phones seem unable to use our PRI
trunks. A sample anonymized call is provided below with the PRI debug
calls embedded. Any thoughts,
comments or suggestions would be welcome. In anonymizing it, I preseved
the format
and number of digits sent.
-- Accepting AUTHENTICATED
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
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
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 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,