On Sat, 23 Mar 2013 00:49:31 +, Jaap Winius wrote:
There is a file, called /etc/sysctl.d/bindv6only.conf, on my system that
sets /proc/sys/net/ipv6/bindv6only to 1 and it's been there since June
2010 when I installed Debian squeeze on my server machine (while squeeze
was still in its
On Fri, 22 Mar 2013 10:07:57 +0100, Jakob Hirsch wrote:
This is well explained here: http://serverfault.com/a/39561
Indeed, that's the solution!
There is a file, called /etc/sysctl.d/bindv6only.conf, on my system that
sets /proc/sys/net/ipv6/bindv6only to 1 and it's been there since June
On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote:
... For example, if my server sends it a SIP packet with a
register request and a Call-ID that looks like this:
Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a]
... somewhere along they line they end up changing
On Fri, 22 Mar 2013 02:46:43 +, Jaap Winius wrote:
Of course, an even better solution would be if Asterisk had a variable
with which to alter the Call-ID string format so that I could omit the
IP address. :-)
It turns out that there in a variable that can do exactly
On Tue, 19 Mar 2013 02:15:10 +, Jaap Winius wrote:
Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9
to 1.8.13, my server is no longer able to register a connection to a SIP
account at my ISP (XS4ALL in the Netherlands). At the same time, it is
still able
Hi folks,
Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1.
As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can
support IPv6. However, it seems that I can't get it to support both IPv4
and IPv6 at the same time. For example, if in sip.conf I set the
On Thu, 21 Mar 2013 12:21:44 -0700, Michael L. Young wrote:
How are you determining that it is not listening on IPv4?
bindaddr=:: should allow you to support dual stack.
That's what I thought would happen. When I set bindaddr=:: and use
'netstat -lpn |grep 5060' it shows:
udp6 0 0
On Thu, 21 Mar 2013 16:02:17 -0700, Michael L. Young wrote:
Let me try to understand this. With bindaddr set as bindaddr=::, upon
starting Asterisk, you are fine and all your IPv4 peers connect
properly. Therefore, dual stack is working at this point. ...
You minunderstand. When I start
On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote:
Hopefully, my ISP will see fit to squash this bug ASAP.
Well, I got my answer from them quickly enough: Nope.
Luckily, somebody was kind enough to suggest a workaround. Unfortunately,
it involves, downloading the source code and making
On Tue, 19 Mar 2013 11:58:22 +0100, Asghar Mohammad wrote:
try srvlookup=yes
Already tried that, but enabling DNS lookups makes no difference when
establishing the SIP connection. The error message that I keep seeing at
the console looks like this:
[Mar 19 12:47:21] NOTICE[7494]:
Hi folks,
Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9
to 1.8.13, my server is no longer able to register a connection to a SIP
account at my ISP (XS4ALL in the Netherlands). At the same time, it is
still able to register a different account with another SIP
Hi folks,
How can I take advantage of a high-bandwidth CODEC, like G.722, for
internal communications at my site, but use G.711 (alaw/ulaw) for all
other outgoing calls? I need G.711 to support Inband DTMF signaling.
As my site has multiple locations that are tied together with IAX
Hi folks,
What methods are available for testing IAX2 service availability? I
know about iax2 show peers and iax2 show registry, but I'd like
some alternatives.
Tcpdump shows a little more about what's going on, but a handy test
using nmap doesn't seem to work anymore (see
Quoting Matt Riddell li...@venturevoip.com:
Maybe you could do:
Set(CDR(userfield)=${CALLERID(num)})
Before dialing SIP/1000
That looks so simple -- and it actually works! -- although exactly not
in the way that I was expecting. Instead of replacing the contents of
one of the existing
Hi folks,
My v1.6 Asterisk system logs all Call Detail Records to a PostgreSQL
database, including those handled by the Privacy Manager.
Unfortunately, even though I can use the CLI to see the information
being submitted by anonymous callers to satisfy the demands of the the
Privacy
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with Asterisk 1.4 is the PrivacyManager. However, this was not
straightforward, because anonymous SIP calls arrive with
${CALLERID(num)} =
Quoting Warren Selby wcse...@selbytech.com:
Try removing the quotes in your n(true) priority.
From FAILED? That makes no difference: with or without the quotes,
the result is always 0, which leads in the Dial() rule being executed.
Actually, though, that's not even relevant, because before
Quoting Tilghman Lesher tles...@digium.com:
http://www.digchip.com/datasheets/parts/datasheet/222/82V2088-pdf.php
See pages 17-18 of the associated PDF. While this is not the T1 framer chip
used, the values are identical, which leads me to believe that these values
are actually industry
Quoting Tilghman Lesher tles...@digium.com:
The value selected should almost always be zero. However, if the cable
is of a significant length, another value must be selected, but which
one? There are two columns: CSU and DSX-1. When is it appropriate to
use the one or the other to determine
Hi all,
When configuring Asterisk with an ISDN card, it will at one point
become necessary to select the LBO (Line Build-Out) value. This is an
integer (0-7) that is determined by the length of the cable and is
selected from the following table. Many of us are familiar with it:
Quoting Alfredo Peña arp...@gmail.com:
Try using this line in the [general] section of sip.conf in your
simulated SIP provider machine:
realm=sip.provider.com
No, that didn't seem to make any difference. However, this did:
insecure=invite
This prevents the Failed to authenticate on
Hi all,
The kind of configuration that I use in my sip.conf to connect to
various commercial SIP providers looks like this:
[general]
context=incoming-calls
canreinvite=no
qualify=yes
register = jwinius:pass...@sip.provider.com/0201234567
[provider]
type=peer
Quoting Motiejus Jak?tys desired@gmail.com:
If I understand well - you want second PBX to act as your sip.provider.com
add this to your /etc/hosts (on primary pbx):
10.10.10.10 sip.provider.com
No, I'm afraid you misunderstand. This has nothing to do with DNS and
not being able to
Quoting Jaap Winius jwin...@umrk.to:
Being both impatient and charitable, I'll try answering this myself:
ISDN uses LAPD for the D-channel and LAPB for data connections over
the B-channels. However, LAPB is irrelevant for Asterisk, because when
the B-channels are used for voice they carry
Hi all,
Thanks to Russ Meyerriecks for his previous reply in this thread,
which was very informative. I'm now hoping that someone will comment
on the following:
ISDN uses LAPD for the D-channel and LAPB for data connections over
the B-channels. However, LAPB is irrelevant for Asterisk,
Hi all,
On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is:
~# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
Span
Quoting RESEARCH resea...@businesstz.com:
Can you post outputs for the following commands;
#asterisk -rx 'pri show spans'
#asterisk -rx 'zap show channels'
#wanpipemon -i w1g1 -c Ta
Sure thing! Here they are in succession:
==
#
Quoting James Lamanna jlama...@gmail.com:
I would call KPN Telecom and ask them for help as well.
They will have much more sophisticated tools for debugging PRIs and also will
be able to check on their end if they see the D-Channel as up.
After studying the configuration more closely, the
Hi all,
My problem boils down to these errors:
... Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)
== Everyone is busy/congested at this time
This is triggered by lines in extentions.conf such as:
exten = _X.,1,Dial(ZAP/g1/${EXTEN},,W)
The
Quoting Jaap Winius jwin...@umrk.to:
The question remains: how can a remote Asterisk server be receiving
SIP packets that still contain the private net IP address of a client?
Okay, I fixed it: by installing siproxd on the firewall system of the
local network. With the Debian systems I'm
Quoting Matt Riddell li...@venturevoip.com:
[Nov 11 14:29:47] WARNING[6365]: chan_sip.c:1787 __sip_xmit:
sip_xmit of 0xb63d5694 (len 444) to 192.168.8.30:5060 returned -1:
Operation not permitted
Are you binding to an address that the box doesn't own?
Check the top of sip.conf.
It's
Hi all,
My Asterisk problem today involves getting a SIP client on a private
net to register with a server somewhere else on the Internet. This
worked for me about a year ago no problem, but now I see an error
message on the remote server every time the client attempts to connect
(the
Hi all,
For a while now I've been using Asterisk together with HFC-PCI cards
(Cologne chipset) for Euro-ISDN BRI support. However, I do not
consider this to be the most reliable solution and believe that the
most stubborn problems have always been software related.
If my clients are
Quoting Jorge Mendoza mend...@tcc.com.pe:
We use Patton BRI gateways. No problems so far.
If possible, we prefer to keep telephony interfaces out of Asterisk box.
What a great idea! I'm going to remember that. Unfortunately, I
believe that would be of no use if you also wanted to use your
Hi all,
Is it possible to display or print variables in Asterisk (e.g. in the
CLI) for debugging purposes?
For example, I'm using two different types of SIP phones: the Snom M3
and the Siemens S675IP. However, when anonymous callers submit a
number to the PrivacyManager, only the Siemens
Quoting Jaap Winius jwin...@umrk.to:
Previously, I had the PrivacyManager working for me exactly as would
be expected, but after upgrading the OS to Debian lenny and Asterisk
to v1.4.21.2 that's no longer the case. Anonymous callers are still
confronted with the PrivacyManager, but now
Hi all,
Previously, I had the PrivacyManager working for me exactly as would
be expected, but after upgrading the OS to Debian lenny and Asterisk
to v1.4.21.2 that's no longer the case. Anonymous callers are still
confronted with the PrivacyManager, but now no matter what I set the
Hi list,
Are there any reliable wireless SIP phones available on the market yet?
Six months ago I went for the Siemens Gigaset 675IP. Although there
was a bug in the MWI support, unit #1 seemed fine for the first few
weeks, so I bought #2 and #3. Then the problems started. Of the three
Quoting Erik de Wild: Tripple-o [EMAIL PROTECTED]:
What is the most reliable method for Asterisk
to detect the Called ID for incoming calls on
an analog line in the Netherlands?
In Holland you have to pay to receive cid info on the incoming line.
I've got that and I've tested
Hi folks,
Would anyone here happen to know of the existence of a Dutch Asterisk
mailing list? If so, where can it be found?
It's not that I'm unable to pose my questions here in English, but I'm
hoping that I may sooner find an answer there to the following question:
What is the most
Hi list,
Has anyone here used one of these cards and got it to recognize
incoming CIDs in Denmark, Sweden, or the Netherlands?
I'm still looking for a way to attach an analog line to my Asterisk
system in the Netherlands that recognizes incoming CIDs. I've now
purchased a Digium Wildcard
Quoting Marco [EMAIL PROTECTED]:
* The firmware and ALL of the pre-recorded messages are in german. I
had some customers a little scared about this!
I have German units too (of the S675IP), but it's easy to switch the
menu language to English. If the pre-recorded messages are still
Quoting Jerry Harshany [EMAIL PROTECTED]:
There is an additional alternative for a ringback to a caller, which
is to use the Call File capability as noted in Van Meggelen's
Future of Telephone; 2nd ed, p306.
As it says in the book, call files allow calls to be created through
the
Quoting Michael Graves [EMAIL PROTECTED]:
in case anyone is interested, I've just taken ownership of a small home
network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone.
It works great with Asterisk. ...
Sounds great, especially where you say that you got MWI to work with
Hi list,
Regarding callback functionality, it seems that Asterisk only includes
a provision for callback in the voicemail configuration, for
authorization purposes, but not an actual callback mechanism. For
that, there are various
free 3rd party AGI (Asterisk Gateway Interface) scripts
Hi list,
The Linksys SPA-3000 and SPA-3102 are often used as PSTN gateways for
Asterisk. They're cheap and convenient to use. Both have worked fine
for me, except I've never been able them to pass on incoming Caller
IDs. I know about the PSTN CID For VoIP CID and Caller ID Method
settings
Hi list,
On my system, PrivacyManager is not reacting to anonymous calls.
Whenever I dial into my system with my mobile phone's number hidden,
the CLI message CallerID Present: Skipping shows up and and my SIP
phone rings anyway.
Perhaps the cause is due to the fact that when there is no
Hi list,
After successfully configuring Linksys SPA3000 and SPA3102 devices as
Asterisk PSTN gateways, the only thing I can't get working is the PSTN
Caller ID. The analog and SIP phones I've used can both display CIDs
for internal calls, while the analog model also displays CIDs
Quoting Tim Johnson [EMAIL PROTECTED]:
Your caller ID is probably being over-ridden by the settings in your
sip.conf file. Remove the caller ID from your PSTN section of the
sip.conf, and the CID should be passed on from the POTS line.
That sounds like a good idea regardless. On the SPA3000
Quoting Tim Johnson [EMAIL PROTECTED]:
What do you have for your PSTN Answer Delay (in PSTN tab)? I had to
set mine between 3 to 5 to get reliable CID from the POTS line. This
was for a SPA3102, not a 3000. I've never had a 3000, but everyone
says they are nearly identical.
I normally have 0
Quoting Jaap Winius [EMAIL PROTECTED]:
My problem is that normal SPA3102 configurations just don't seem to
work. I can't even get the FXS port to register. I'm beginning to
suspect that my unit is defective.
Today I called the vendor (voipsolutions.be) and was passed on to a
knowledgeable
Quoting Mandeep Singh Bhabha [EMAIL PROTECTED]:
what i did to configure SPA3102 is ...
My problem is that normal SPA3102 configurations just don't seem to
work. I can't even get the FXS port to register. I'm beginning to
suspect that my unit is defective. Here's why:
If I configure
Hi list,
After failing to get a Sipura/Linksys SPA3000, which I've configured
as a PSTN gateway, to pass on the Caller ID, I decided to try my luck
with a Linksys SPA3102 after hearing some promising stories.
Unfortunately, I've run into a completely new problem: it seems
Asterisk won't
Quoting Tim Johnson [EMAIL PROTECTED]:
I see you put a password line in your sip.conf, but I do not see a
username line. Also, you might want to check the port #'s for both the
Line 1 and PSTN line. I use 5060 and 5061, respectively. Hopefully
this either helps, or puts you on the right
Hi list,
The default automon (touch monitor) output file name format is:
auto-epoch-caller-callee.wav
A variable is available to modify the second half:
auto-epoch-${TOUCH_MONITOR}.wav
But, I can't modify the first half, 'auto-epoch-', with any variables
that I know of, including
Hi list,
Hopefully, some of our Dutch members can help with this one. I'm also
based in the Netherlands and am using a Sipura (Linksys) SPA-3000
(firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test
system. It works fine, except that the Called ID (CID) is not working.
Quoting Tim Johnson [EMAIL PROTECTED]:
I have a SPA3102 which is supposed to be similar. Make sure you leave
the PSTN -- Subscriber Information -- Display Name blank. Also, in
your sip.conf file, do not specify any callerid= value. ...
It was worth a try, but unfortunately it makes no
Quoting Razza [EMAIL PROTECTED]:
I'm running - 2.6.23.15-137.fc8 which appears to be supported at ATrpms.
I built a F8 box, added ATrpms to the repository list, executed yum -y
install fcpci, which forced a kernel upgrade.
Unfortunately, I can't test any of this since I'm running a Debian
Quoting Axel Thimm [EMAIL PROTECTED]:
There are patches inside that will work on Debian as well, just get
the src.rpm and pick out the patches.
Now, why am I not surprised? Actually, if I had known this back in
early December, I'd be following your advice and thanking you now. I
previously
Quoting Razza [EMAIL PROTECTED]:
Hi list, i'm keen to move to Asterisk 1.6, so really need to update my
system which is running Mandrake 9.2 although it has been solid for years,
fo Fedora 8. I have a Fritz! card for ISDN BRI, ... I 'modprobe capi' and
'modprobe fcpci' which appear to work
Quoting Mojo with Horan Company, LLC [EMAIL PROTECTED]:
Will Set(MONITOR_FILENAME=/blahblah/filename) work for you?
No, that doesn't work. ${MONITOR_FILENAME} can influence the filenames
in the string that you can tack onto the somix sequence using
${MONITOR_EXEC_ARGS}, but not the file
Quoting Steve Langstaff [EMAIL PROTECTED]:
The 481 Call Leg/Transaction Does Not Exist response to the
NOTIFY makes me think that you might need to configure the
phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages
from the phone when it is booted?
Yeah, sure. And there are some
Hi list,
Before purchasing a number of Siemens DECT SIP phones, the Gigaset
S675 IP, I read that the problems with MWI had been fixed with the
latest firmware version (see
http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm
not so sure that's the case.
After setting up a
Quoting Henry Devito [EMAIL PROTECTED]:
Try adding [EMAIL PROTECTED] (or what ever your voicemail
contexxt is) I've had to add the voicemail context to get MWI
to work correctly in the past.
According to the documentation, you shouldn't have to add @context
if the context is 'default'.
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${EPOCH}-caller-calee
It's possible to use the ${TOUCH_MONITOR} variable to change the
'caller-calee' part, but what about the 'auto-${EPOCH}-' part?
I've been trying to use ${MONITOR_EXEC_ARGS} to add
Quoting Jaap Winius [EMAIL PROTECTED]:
After wrestling with the voicemail system for a while (Asterisk 1.4.14,
Debian etch), I got it to work, but I still have lots of questions,
like:
* Why can't I delete any voicemail messages?
(Response: Message undeleted.)
* Why can't I
Quoting Drew Gibson [EMAIL PROTECTED]:
We made this function reliable by including the word quickly in our
instructions for pressing the keycode to start the recording. ...
Indeed, but somehow I don't think my users will be satisfied with that.
Although a private confirmation beep to the
Hi list,
Can someone please explain how to get one touch recording (automon) to
work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My
current configuration includes the following settings:
In /etc/asterisk/sip.conf:
[2000]
; Siemens Gigaset S675 IP wireless SIP phone.
Quoting Doug Lytle [EMAIL PROTECTED]:
featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation (default is 500 ms)
courtesytone = local/stutter ; Sound file to play to the parked caller
; when
Hi list,
After wrestling with the voicemail system for a while (Asterisk
1.4.14, Debian etch), I got it to work, but I still have lots of
questions, like:
* Why can't I delete any voicemail messages?
(Response: Message undeleted.)
* Why can't I listen to the messages in the
Hi list,
After recently setting up voicemail for Asterisk 1.4.14 on my Debian
etch server, I noticed that I can't delete any old voicemail messages.
The voicemail menu option Press 7 to delete this message is
available, but when I press 7 the response is always message
undeleted and the
Quoting Michiel van Baak [EMAIL PROTECTED]:
On 00:38, Wed 06 Feb 08, Jaap Winius wrote:
Hi list,
After recently setting up voicemail for Asterisk 1.4.14 on my Debian
etch server, I noticed that I can't delete any old voicemail messages.
The voicemail menu option Press 7 to delete
Quoting Andy Doss [EMAIL PROTECTED]:
File permission error?
That is just my first guess. I am kind of new to Asterisk myself.
The files are all in /var/spool/asterisk/voicemail/ where the asterisk
user has read/write access to everything. Also, I see no error
messages that would indicate a
Hi list,
Recently I figured out how to automatically record (Monitor) both
incoming and outgoing calls, which is handy. However, since this is
not always desirable (or legal), can Asterisk be configured to start
recording at some arbitrary point during a call, to be determined by
the
Quoting Jaap Winius [EMAIL PROTECTED]:
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro,
kernel, modules, versions, config files).
Thanks to the support I received here I now have a working system, so
I
Hi list,
My Asterisk v1.4 system now has two ISDN channels and two SIP
channels. The idea is to make a dialplan that mostly uses the SIP
channels for outgoing calls, but I'd like those to fall back
automatically to ISDN if the SIP channels aren't available, possibly
in combination with a
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What is the Dial command you use?
Can you post the relevant part of your diaplan?
exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r)
In addition: are you sure that there are channels set for group=0 ?
Maybe try a channel directly: Zap/1 or Zap/2 instead
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/1000-081f68b0,
Zap/g1/[EMAIL PROTECTED]||r) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/[EMAIL PROTECTED]
Again, you're calling an incorrect number. You dial
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
... I get the wierd impression that either both modules somehow
get interrupts from the two cards, or each module handles a
different card. This hsouldn't happen.
So try blacklisting one of them:
I've already done something like that: removing the
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What do you mean by In Use?
# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In
Hi list,
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config files).
I've tried to get it to work on a Debian etch system with an HFC-PCI
card and the zaptel package
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
(if you set in /etc/default/zaptel: ZAPBRI_SIGNALLING=bri
you'll get that from genzaptelconf.
If I create a file like this, I end up with signalling=bri_cpe instead of
signalling=bri_cpe_ptmp.
Anyway, either you use zaphfc or vzaphfc. The first one
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What is the output of:
pri show spans
PRI span 1/0: Provisioned, Down, Active
PRI span 2/0: Provisioned, Down, Active
Do incoming calls work?
Negative, and nothing shows up on the CLI. And that's after creating
separate contexts
Quoting Michiel van Baak [EMAIL PROTECTED]:
I don't know about NL but in the UK, multiple ISDN2e lines have to be
configured as bri_cpe_ptp not bri_cpe_ptmp. Have you tried this mode?
It's the same here in .nl
Interesting, but I would think this to be unnecessary in my case,
since I have
Hi list,
Attempting to get an ISDN-BRI line connected using an HFC-S PCI card
together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch
system, I find that I can't access the card's resources because the
channels are always be busy. An attempt to call out results in the
following
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What is the output of:
pri show spans
PRI span 1/0: Provisioned, Down, Active
PRI span 2/0: Provisioned, Down, Active
Do incoming calls work?
I haven't configured that yet.
Interesting... which one of those two is used?
Good question.
Hi list,
After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error
messages related to my HFC-S PCI card disappeared, but now I can't
access the card's resources because it always seems to be busy. Any
idea why?
Thanks,
Jaap
PS -- Below is some info regarding my
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What do you mean by busy? What exactly do you see?
This kind of thing:
# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
... this error that keeps appearing in my syslog and kern.log:
zaphfc: empty HDLC frame or bad CRC received
Try using the zaptel packages from:
deb http://updates.xorcom.com/rapid etch main
This upgraded Asterisk from v1.2 to v1.4.14 and the
Hi list,
After a recent upgrade to Asterisk v1.4.14, my message log is now
filling up with
the following error messages:
-
[Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645
determine_firstline_parts: Bad request protocol Packet
--- (1 headers 0 lines) ---
bitis*CLI
--- SIP read
Quoting Michiel van Baak [EMAIL PROTECTED]:
-
[Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645
determine_firstline_parts: Bad request protocol Packet
--- (1 headers 0 lines) ---
bitis*CLI
--- SIP read from 82.101.62.99:5060 ---
Cirpack KeepAlive Packet
-
Are you
Quoting Michiel van Baak [EMAIL PROTECTED]:
Sounds like a good idea, but I'm having trouble getting the source
code for Debian etch from xorcom.com to compile regardless.
I have no idea.
I got it to compile. My mistake; I had attempted to modify chan_sip.c
directly. It then refused to
Hi list,
Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run
into some serious problems. The first thing I noticed was this message
that would show up every five seconds on the CLI:
Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No
D-channels
syslog and kern.log:
zaphfc: empty HDLC frame or bad CRC received
Any idea how to get rid of it?
Thanks,
Jaap
==
Quoting Jaap Winius [EMAIL PROTECTED]:
Hi list,
Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run
into some
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 DEACTIVATED
(F4) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
Span 2: ZTHFC2 HFC-S PCI
Hi all,
Could someone please point me in the direction of some reasonable instructions
for setting up ISDN BRI support for Asterisk 1.2 (Debian etch) with
HFC-PCI cards (Cologne chips) and/or cards with Winbond W6692CF chips?
I keep finding solutions that involve running misdn-init. However,
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
Could someone please point me in the direction of some reasonable
instructions
for setting up ISDN BRI support for Asterisk 1.2 (Debian etch) with
HFC-PCI cards (Cologne chips) and/or cards with Winbond W6692CF chips?
apt-get install asterisk
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
However, after installing the zaptel package, I get these
errors:
# genzaptelconf -sd
Stopping Asterisk PBX: asterisk.
cat: /tmp/tmp.uiMna12463/span_termtype: No such file or directory
cat: /tmp/tmp.uiMna12463/span_termtype: No such
Hi all,
Perhaps someone here could help me with this. I'm new to Asterisk, but
have already met with some success at getting my first system to work
with two different VoIP (SIP) providers: XS4ALL and InternetCalls.com.
The config
for the former works fine, but my InternetCalls.com config
Quoting Justin Case [EMAIL PROTECTED]:
What comes up in the Asterisk CLI?
When it's not working, nothing appears in the CLI even though I've used
set verbose 10.
Also it can be a NAT issue?
How can that lead to this intermittent behavior? I've already set
nat=yes. Also, I'm using an ADSL
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