Displaying 20 results from an estimated 3000 matches similar to: "Fail over to Pri on VoIP connection failure"
2007 Jul 24
2
Dial out through multiple Zap groups
Hi,
I'm trying to set a rule to dial out through multiple
Zap groups so that, say, g0 is the cheaper POTS lines
group
and must be used first. However, if g0 is busy or
disconnected then try dialing out g1.
My g0 group is made up of 4 analog lines connected to
a 4-FXO card. I disconnected the RJ-11 wires from the
FXO card
to simulate a line disconnection. So theoretically all
calls should
2015 Mar 25
2
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
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 all numbers just the numbers are configured in thses ip phones.
but when i configured the same trunk in x-lite i can call theses ip-phones
without
2011 Sep 28
2
PSTN connectivity
Hi All,
I am trying to connect my asterisk box with freepbx to PSTN. I
have purchased x100p FXO card and installed in my asterisk server. My
freepbx detected the x100p FXO card and i can see the card specific details
in command line. I have configured the following things.
1. OUTBOUND caller id and Dialing rules in Freepbx.
2. INBOUND route
When i call to the PSTN number before
2015 Mar 27
2
call between snom 300 and aastra 6731i
You would need to give more information really.
Your sip.conf file listing the entries for the phones especially which
codecs are permitted.
A copy of the 'asterisk -rvvv' console output when you make the call.
On 27/03/15 17:05, Salaheddine Elharit wrote:
> please no body has som with aastra can help me in this issue
>
> 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit
>
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
2008 Oct 28
1
Multiline Analog Setup
What is involved in provisioning Asterisk to use a multiline analog service from our local telco?
I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk
interprets and deals with two incoming calls and/or two outgoing calls?
Thanks!
jlc
2010 Nov 03
1
Ring back problem on SIP calls. Order of 183 Session Progress and 180 Ringing
Hello everyone!
I've had this problem for a while and cant figure it out. When an outside
caller calls an extension on my asterisk system, they do not hear any sort
of ringing. Inside extensions calling other extensions do hear ringing. We
have 3 other asterisk systems that are configured the same way, but do not
have this problem. We think it has something to do with asterisk 1.6. The
other
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
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
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
2010 Mar 26
1
send a call from A to B use sip trunk prablem
i have a prablom here,
i want to send a call from A to B use sip trunk ,
the call can sended B,but not work to PSTN.
the message from B server. help pls,what's rong?
>
> <--- SIP read from 192.168.0.176:5060 --->
> INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport
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
2006 Feb 10
0
Half Solved - Fail over to Pri on VoIP connection failure
I want to say thanks to everyone for the help so far. I figured out a
way to modify some AAH code that worked for me (well sort of). The line
I modified is s,14 in macro-dialout-trunk. Then I just added a variable
and passed it from 9_outside.
I just have one last problem. This waits for an answer not ringing. So
if the called party has a long ring to voice mail the call is dropped
and goes
2005 Mar 09
3
NuFone + VoIPJet = busy busy busy
Hi List,
I'm using VoIPJet and NuFone as a fallback, and it seems that both of
them are circuit busy!
Also it seems that VoIPJet takes forever to return 'circuit busy' while
NuFone does it instantly.
At any rate, is there like a reliable third VoIP provider I can use for
fallback when the two others are busy?
Cheers,
Jean-Michel.
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:
2005 Mar 03
3
Why ${EXTEN} variable changes after Goto ?
Hi,
I'm trying to implement dynamic routing of incoming calls to local extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in advance,
regards,
Rob.
[outbound-capi-ISDN]
exten => _0.,1,NoOp(Calling ISDN
2006 Jan 27
7
AAH out bound routing problem
Hi all
I have installed AAH 2.2 in my P4 PC
following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as per the guide says
and downloaded SJ Phone, and registered user
and when i try to dial the 19197543700
i get message that, all circuits are busy now, please try your call later
and when i see in the console i get this mesage
any help
Called easycall/19197543700
2005 Jun 22
1
missing cdr records
I am experiencing a very wired problem.
Some of my cdr are lost.
I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning.
I am running asterisk 1.0.7; this is simple configuration file:
extensions.conf
[general]
static=yes
writeprotect=no
[macro-gw-voipjet]
exten =>
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
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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