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
>
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 tha
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 th
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
201
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 makin
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 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
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 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
> s
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]: chan_sip.c:
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 provider
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
trunks
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
http://sh
Quoting Matt Riddell :
> 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 fields, a new fiel
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 Manage
Quoting Warren Selby :
> 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 Asterisk
even re
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)} = "anon
Quoting Tilghman Lesher :
> 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 standard.
Well,
Quoting Tilghman Lesher :
>> 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 the correct
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:
CSU
Quoting Alfredo Peña :
> 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 INVITE" errors
Quoting Motiejus Jak?tys :
> 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 reach my second PBX --
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
h
Quoting Jaap Winius :
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
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, bec
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)
Spa
Quoting James Lamanna :
> 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 first thing I
cha
Quoting RESEARCH :
> 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:
==
# asterisk -rx 'pri show s
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 sy
Quoting Jaap Winius :
> 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 run
Quoting Matt Riddell :
>> [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 set to bind t
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 serv
Quoting Jorge Mendoza :
> 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 ISDN
connection f
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 willing
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 di
Quoting Jaap Winius :
> 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
minleng
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
unit
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 a
Quoting Michiel van Baak <[EMAIL PROTECTED]>:
>> Would anyone here happen to know of the existence of a Dutch Asterisk
>> mailing list? If so, where can it be found?
>
> Not that I know off.
I didn't think so. Either that, or its existence is a well kept secret!
> I can start it if you want.
Ye
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 TD
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 stil
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 wit
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
th
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 avail
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"
setti
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
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
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 SPA30
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
correctly
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 (voipsoluti
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
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,
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 le
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 n
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.
I'm
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 ${M
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
previou
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 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 wo
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
${MONITOR_EXEC_ARGS}, but not the
Quoting Steve Langstaff <[EMAIL PROTECTED]>:
> The "481 Call Leg/Transaction Does Not Exist" response to the
> NOTIFY makes me think that you might need to configure the
> phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages
> from the phone when it is booted?
Yeah, sure. And there are s
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${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 s
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 @
if the context is 'default'. But, I
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 n
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?
>
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
>; whe
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 t
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.
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
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 indicat
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.
&g
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 th
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 user,
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 w
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
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 n
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 i
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 HDLCFC
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
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 (v1.4.
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 fi
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
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 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 que
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 CL
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
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 configuration.
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
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
>> <
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 rea
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
ring in my 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
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 avail
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)
>> Spa
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_ter
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
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 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
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