Displaying 20 results from an estimated 6000 matches similar to: "PRI Cause Code Help"
2008 May 19
2
Recording problems, reinvites
Hello,
I'm wondering if anyone else has been observing problems lately with
1.4.18 and higher releases of asterisk not properly recording calls.
When using MixMonitor, the resulting file is only a few bytes long.
I think this is because asterisk is doing Native bridging even though
MixMonitor should block that.
Did something change around the release of 1.4.18 that would have
changed
2007 Aug 07
3
test the email-list
test only. good luck!
james.zhu
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2005 Jan 15
2
IAX2 one side loses audio
It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop
audio on one side. I place a call out through voipjet, and call
quality is flawless. However a few minutes later the person who I'm
talking to can no longer hear me. I can still hear them.
What should I look for to resolve this? Has anyone else had this problem?
Using last night's CVS this problem still exists.
2007 Jan 30
3
Toll-free dialing via PRI problem
We have a PRI from Telepacific. Asterisk 1.2 and a Sangoma A101 T1 card.
Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but
the calls are never answered. All other calls, and most toll-free numbers
are not affected. The numbers that are affected are all travel related
companies (United Airlines, American Airlines, US Air, Starwood Hotels,
etc.) we cannot connect to
2004 Jul 19
1
Flash Zap trunk from a Sipura
Hello,
In my quest to create several proof of concepts for what can be done
with Asterisk, I've run into a bit of a problem. I have a pair of
SPA-2000's acting as off premise extensions for an analog line. When a
call waiting call comes in, the caller id information makes it though
the ULAW codec and displays on the caller id box, however asterisk
doesn't seem to want to pick
2005 Feb 25
1
Transposed ringing
I don't suppose anyone might know why I hear ringing transposed over
itself when I place a call out via PRI?
SIP to SIP is fine
SIP to IAX is fine
SIP to PRI is always transposed
I mean sometimes you don't notice it much because it's lined up right,
but other times you'll hear a really long ring (starts sounding normal,
then sounds "weird" -- like two rings played at
2007 Mar 01
4
Cannot hear ringback music from telco
Hello,
We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to
the telco, users mainly use snom 320/300 SIP phones.
When dialing to an external phone number with custom ringback music, users
reported that they could not hear the music but can only hear the standard
ring tone generated by the system.
Is there any kind of settings need to allow the ringback music pass to the
2004 Jun 06
2
Analog Bridged Calls Pulsate
Hello,
I've been playing around with two generic X100P analog cards to create a
proof-of-concept system before we go ahead and hook up a PRI. I'm
running into a reproducible problem with sound quality of bridged calls,
and am hoping someone will be able to point me in the right direction.
I have in my dial plan a _9. extension so outgoing calls can be made...
the first thing is
2004 Aug 01
2
Parking & SIP Phones
Hello,
I know not too long ago I saw /something/ _somewhere_ about an
adjustment to call parking that allowed blind transfers from SIP phones
to park a call and still be able to hear the parking lot stall number.
Unfortunately, I have no idea where I saw that (google turned up little,
couldn't find it on the list either). I'm using Sipura SPA-2000
adapters and it doesn't seem to
2004 Nov 28
1
SetVar ALERT_INFO
Hello,
I've got my dialplan configured to do a double ring when a customer
service call comes in, and a normal ring when an extension is dialed
directly. When a customer service call is transferred, I want to ring
to revert back to normal.
In the local extension macro, I have the following
; make sure ring is set to default
exten => s,n,NoOp(${ALERT_INFO})
exten =>
2005 Mar 22
4
OT: does Sipura SPA 3000 support UK caller id?
Hi,
the topic says it all really.
Does the Sipura 3000 detect and report UK clid correctly?
thanks
Mike
2005 Sep 11
1
Presence Fully Supported?
I've seen lots about presence and Polycom phones recently. I've got one
here for evaluation but noticed other phones only seem to appear busy
when they initiate a call. If they receive a call, they still show as
available.
Is this a config problem on my part, or is that as far as presence is
working right now?
Thanks!
Trev
2007 Aug 25
2
Asterisk 1.2 + Zaptel 1.4 + HPEC = Crash?
Hello,
Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and
HPEC 9.00.003?
In particular, with a hardware configuration similar to:
Module 0: Installed -- AUTO FXO (FCC mode)
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Not installed
Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules)
I have two fully independent systems
2009 Jun 20
1
PRI cause codes
I am trying to retrieve the cause code of a outgoing call over a PRI
where the number called is out of service. When an out service number is
called I get a recording that the number dialed is not a working
number. I see cause code 1 in in the CLI as soon as the call is dialed
the Telco recording goes on for 30 sec. then hangs up. Any idea on how
retrieve info that the called number is is
2004 Aug 19
1
Debit/Credit Card Terminals
Has anyone tried using a debit/credit card terminal as such:
Terminal <-> SPA-2000 <-> Public Internet <-> * <-> PRI
I'm hoping someone will tell me they have done this successfully and
rarely experience dropped calls. Though I'd like to hear from anyone
who has tried and failed as well.
Thanks,
Trevor Peirce
2005 Aug 25
1
PRI signaling experts please help
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocent):
-- Called 12345678@sip-outbound
-- Got SIP response 486 "Busy here" back from
2006 Apr 10
4
callerid name inboune from PRI
I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?"
AT&T Local did not, But XO communications said they did.
Before I call to complain, is there an setting to turn this on in asterisk?
I want to make sure that I have my side covered before I call XO.
My current zaptel.conf is:
context=from-pstn
switchtype=national
2005 Dec 05
3
PRI indications.
Hello,
i have succesfullu setup asterisk with Sangoma E1 card, evrything works well
but i don't know how to pass indications from telco switch to the user - when
users call bad number telco switch shuld talk "unallocated number" but its only
send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients?
My /etc/zaptel.conf:
span=1,0,0,CCS,HDB3,CRC4
dchan=16
2005 Jan 27
2
Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi,
well, most of the things work right now due to the help of peter
svensson, but after heavy use of our ericsson BP250 today several
problems appeared.
i split into several mails as they are seperate problems.
* i can't signal Busy to the calling party.
asterisk receives busy from the ericsson PBX but does not forward
this to the external caller. i tried with exten =>
2005 Feb 23
1
Request for PRI Dump
Hello,
To assist Matt's efforts in bug 3554 (2BCT & CNAM), I'm hoping someone
can provide a dump of the setup and related messages from a PBX that
supports outgoing Station Name to the CO.
As suggested in the bug, I tried to ask my telco for a dump of the setup
messages for a client that supports this but was told to contact my
vendor as they cannot provide that information.