Hi!
I've suffering cut offs after 6 or 7 seconds a call is answered,
incoming calls are working fine, but outgoing ones show the gollowing
messages when are being dropped
[...]
It seems the SIP ACK is not being received properly.
I can confirm this issue: In my case it happens with calls
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,GoTo(${vgLabel})
But in stead of vgLabel becoming the SUM of 2 numbers, it is just a
string :
Use the MATH function.
Philipp
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Hi!
It is causing an issue for me. One SIP UA works fine - ring, forward, etc.
While the other does not.
Make the UAs listen on different ports (for example 5060 and 5062) and
see if that solves your problem - if you can't make them have different
IPs, that is.
Also be sure to fully
Hi!
2. Add BRI card(s) to the computer to run Asterisk and somehow hook
up the Samsung.
Do-able. Connect Asterisk to your ISDN2, then host the Samsung off the
asterisk box. But then, might as well dump the Samsung and just put
VoIP phones on everyones desks.
If you decide to go down
Hi!
Side note: Stay away from solutions that use mISDN, instead go with
Zaptel (DAHDI), Woomera or CAPI.
Interesting.
I've been usng mISDN for some years now without issues. Why should I
migrate to DAHDI?
None - if you are happy then don't touch it. :-) Otherwise search this
list's
Hi!
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
6839, digest has 3169
You most likely have two SIP UAs that use the same IP, of which the 6839
account is listed last in sip.conf while 3169 is trying to auth
(unsuccessfully).
Philipp
--
Hi!
I have asterisk 1.4
I want to make a MGCP trunk as a client to connect to a provider who is
using MGCP protocol, he provided me with user password,
You will most probably need 1.8 for this, with 1.4 you will certainly not
be able to succed. Read more:
Hi!
I've turned off t.38 and all of the codecs except ulaw; I still have the
same problems. SOMETIMES it works. Other times, the sniffer clearly
shows that the media simply isn't being sent. NOTHING is being sent.
Anything else I should check?
Look at the firewalls and possible SIP ALGs
Hi!
In this case I will want to use Snom phones. TFTP is available, but no FTP
(with indeed then a username and password). FTP would be great...
You could also consider to use the SNOM Redirection Service for
provisioning:
http://wiki.snom.com/PROVISIONING
Remark: TR-69 provisioning
Hi!
but all of a sudden we have all calls origination from one sip
extension opening channels which have the name of another sip extension
in the channel name.
Do the devices of this extension happen to have the same IP address?
Philipp
--
Hi!
Is there a way/software which can act as a middle man between asterisk
and the ethernet ports, and by default sends registrations to asterisk
only from eth0, and if this port fails, sends registration coming in
from eth1?
Spanning Tree (STP, RSTP, MSTP)
--
Hi!
Can someone suggest where to look? Could this be the ITSP?
- turn off IAX trunking mode
- test with SIP to find if it IAX causing the trouble
- capture the RTP traffice on the other side and let wireshark have a
look at that stats (loss, jitter)
Philipp
--
Hi!
Trunking only reduces overhead after 4+ calls, so that shouldn't help
either. (Since this occurs at 2 calls)
Trunking requires a timing source, and you might have trouble with your
timing, that is why I suggested this (and because you did not tell us
wether you have trunking enabled or
Hi!
I'm having difficulty with registering a SIP account in a Snom 320 IP-
phone.
Do a SIP trace on your SNOM phone, and you will most probably see that
the 401 reply of Asterisk does not arrive on the phone. Then check your
STUN/ICE settings on the phone in combination with the nat=
Hi!
Can you tell me how I can get my Snom 320 auto-answer the call when I
use the Page()-command ?
Configure a special identity on the SNOM that is set to auto-answer in
the phone's configuration. Or consider to use multicast instead of Page()
if your network topology doesn't stand in the
Hi!
I've also been lead to believe that I can set a URL for a custom
ringtone in the Alert-Info header but can't find the exact syntax
anywhere.
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+snom
http://asterisk.snom.com/index.php/Asterisk_1.4/Ringtone_Configuration
Philipp
Hi!
Just out of interest, have you ever got this working?
Yes, sure.
Mine just isn't but I'm starting to think that my mp3 to 8000Hz Mono
16 bit wav files is a bit dodgy
Very well possible. Also look at the individual identity x
configuration and consider to select Custom ringtone, then
Hi!
Looks like I still don't understand how SHARED works :-(
exten=6052,n,Dial(SIP/6052,,M(test))
exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL}))
Please check if in Asterisk 1.6 the Syntax for passing arguments to the M
option of Dial() has changed.
[macro-test]
exten =
Hi!
traffic to an IP address - then, rather than me manually analysing with
wireshark, will analyze the cap file and produce stats on jitter, lag,
delta etc.
This is what RTCP was made for.
Philipp
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Hi!
Why is it a problem? It sounds like Asterisk does silence suppression.
1) With no rtp traffic, the nat device will drop the connection in it's
nat table and thus disconnecting the softphone from Asterisk. (after
the router's timeout period of course)
2) The other issue is you are
Hi!
And the third hit in my google result is this:
http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
Since I mentioned in my previous message that you will find the answer
in the archive of this list you could have found that even without
google. gmane.org for
Hi!
There are 2 things I can't understand
- 1. how can I know channel name?
${CHANNEL}
2. where should I call this SHARED function? before Dial, after Dial?
Either In the macro that you specify using the M option of Dial() or in
the h extension. You will, however, have trouble treating the
Hi!
I see. I want to use SHARED function!
Do you have example how to
to export them to the local call leg/channel ?
Have you considered using Google (or your favourite search engine)?
The search terms asterisk function shared will surely help you, and in
fact point you to the very archive of
Hi Dmitry!
Have you considered using Google (or your favourite search engine)?
Shure, I searched and find nothing.
The search terms C will surely help you, and in
fact point you to the very archive of this mailing list.
Don't know where this quote comes from, but C is absolutely not
.slin is not .wav
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asterisk-users
Hi!
I need the system to be resilient to any network partition, so that
anyone can send announces from any mic to all the reachable clients.
I'd need also to page a subset of all the speakers.
Most of the major phone vendors (that are employed by the users of this
list) have support for
Hi!
Could somebody tell me how to use SHARED function?
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared
I want to get RTCP stats from SIP, but current channel is DAHDI.
How can I get SIP channel?
If you have one DADHI and one SIP channel bridged together, then only for
Hi!
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
Does this mean that when I mark this as yes, a phone that already has
taken a call will be send a second and third call ?!
No, not directly: This setting is only
Hi!
Does this shine new light to the problem ?!
No. Once more: Go and read doc/backtrace.txt.
And check if you have any meaningful information in /var/log/messages for
the timestamp when asterisk crashed.
Philipp
--
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--
Jonas,
everyone here supports you in your effort to get a good Asterisk
installation going, but could you ... maybe restrain yourself a little
bit and reduce the number of hasty postings you are sending to this
mailing list?
Thank you,
Philipp
--
Hi!
I know I post a lot concerning this issue, but this is because this
problem occurs on a production system and I feel very hot breathing down
my neck.
Why not reduce the pressure and revert to 1.4.30 for the production
system until you have figued out the issue? That will give you more
Hi Jonas!
It indicates to be a binary file, however I have not found instructions on
dealing with this @ the link you gave me.
Can you give me instruction on how to handle the core.pid file ?
Could I ask you again to make an effort to reduce your number of daily
postings to this list? If
Hi!
DEVICE_STATE function is not available in asterisk, even DEVSTATE does not
work for me in asterisk 1.4.35. Any other method function to check the
channel status
There is a backport available for 1.4:
http://www.voip-info.org/wiki/view/Asterisk+func+device_State
I assume that with does
Hi!
I am running asterisk ver 1.2.4 and have faced this error:
Try a downgrade to Asterisk 0.7.1 ;-
Philipp
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Hi!
is their a way to keep having a dialtone for the calling party when the
macro is executed ?! Or not ?!
Consider using either FollowMe() or using the G option of Dial() instead.
Philipp
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Hi!
1. Do you have any experience with receiving incoming SMS on an analog or
ISDN landline ? How can then you differentiate an SMS call from a voice
call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems
the way to tell an inbound call is an SMS one is to read the callerid
Hi!
Yes, typically there is only one SMSC that can send you SMS on a fixed
line; look at its Caller ID to identify a SMS call.
Even when the call is coming from a cellphone ?
A SMS is not really a call (at least not in the mobile world), and the
cellphone cannot directly send a SMS to a
Hi!
We're using firmware 7.3.30 on an installation of Snom 300 phones.
Should we stick with it, or do the newer firmwares have better support
for Asterisk?
So what is it that you are missing that firmware 8 does offer? 7.3.30 is
rather stable and therefore a good choice.
Actually I would
Hi!
After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
finishes. On the Asterisk console, I can see that the sound file is indeed
playing, but we can't hear it. [...]
I have tried so many things that I have lost count, and I humbly ask the
collective intelligence of
Hi!
My question is this. Is it possible to tell Asterisk to execute part
of a macro as a block without allowing any other commands to be
processed during that time?
What would be a correct way to do this in 1.4.x?
*CLI show application MacroExclusive
Philipp
--
Hi!
* Remove current STUN support from chan_sip.c. This change removes the
current
broken/useless STUN support from chan_sip.
(Closes issue #17622. Reported by philipp2.
Review: https://reviewboard.asterisk.org/r/855/)
What you do not see mentioned here, and that is a bit
Hi!
How can I remove the Playing digits from parkcall application?
In general you can address problems like this by creating your own set of
sounds files where the obstructing files are either simply missing or
replaced by silence. Use Set(LANGUAGE) right before the action (here:
parking the
Hi!
By a mixed environment I mean some Asterisk servers running on AMD and
some running on Intel
If it was possible for that to matter, then the software would be very
poorly written indeed. As another poster said, the only way that would
have any effect is if you compiled binaries
Hi!
Does anyone know if the behavior of 'r' has changed but was not
documented? If yes, then how does one inject ringback audio before the
call is answered on the called end?
Search this list for progress or progressinband, and look at the voip-
info wiki.
Hi!
Is there ANYWAY to find out which party hang-up the call or if the call
was cut-off due to other reasons?
The only way - apart from putting DAHDI or SIP into debug mode - I can
think of is to use the 'g' option of Dial(): If the remote side hangs up
then 'g' will come into effect; if
read the value of var ${HANGUPCAUSE} next line to dial command.
So how can you be sure this has been populated by PRI and not by SIP?
This will not tell you which side hung up.
Philipp
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Hi!
Although my previous posts in this forum have not received satisfying
answers, here is another question from me.
You might want to consider to reqest a refund. ;-
my question is can i use ChanIsAvail function to get the status of a user
in the format SPI/user-id if i provide user in
Only when I configure my Grandstream to use only G726 (I have 8
choices), I see that the g726-codec is used.
When I configure 7 x g726 and 1 x alaw, then again alaw is used !
Is it normal that Asterisk has such a great preference for alaw ?! The
moment the peer suggests codec alaw (even if
Hi!
Let's say I call by SIP/trunk1/number and the proxy server is
down, is there a way to getCHANUNAVAIL?
*CLI core show application Dial
Unfortunatelythe timeout parameter will not do the job for me. I need
somethingequivalentto qualify to monitor the outboundproxy.
Why not qualify and
The mapping in Asterisk 1.4.24 is the problem: 402 Payment Required
is mapped to 16 Normal termination instead of 21 Call Rejected.
Could you direct me to the relevant file of code where these mappings
are done? Before reporting a bug I would like to confirm whether this
issue has been
Ok, here's the challenge:
I would like to be able to find, match - and then react - upon prompts
that are presented by the outbound/remote side of a call. Think mobile
phone and This user is temporarily unavailable.
Collecting a limited number of known prompt snippets should not be a
problem,
You might be able to record these snippets then pass them through the
Vestec or Lumenvox Speech engine to get what you want.
Unfortunately that won't work because:
* the containing recordings/feeds can be quite long, can be
embedded/surrounded by silence, ringing tones, music or special
Hi!
Is there a way to change the mappings of disconnect reasons to certain
SIP messages? E.G. I need to change the mapping for SIP 402 Payment
Required from 16 (normal termination) like it is in 1.4.24 to 21
(call rejected) as defined in RFC 3398.
* if you think the mapping is wrong, then
Hi!
Question 1 :
[Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
why is combined alaw|g726 and not g726|alaw (reverse) ??
Guess: Here the order presented has no meaning for the order of codec
Also:
There are at least two implementations of the g726 codec, i.e. g726 and
g726aal2. For this also look at the g726nonstandard setting in sip.conf.
It is quite possible that your problem is here.
For quick testing to see if the codec works at all: Configure your phones
to do g726 only (so
Hi!
In the [general] section of sip.conf I have :
disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm
So change the order there and see what happens.
* look at the variable SIP_CODEC for the inbound (first) call leg, and
in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
Hi!
i want to get channel-id before dialing so that i can dial using that
channel id.
I am afraid that is not going to work. Maybe you should take a step back
and describe what it actually is that you are trying to accomplish.
Philipp
--
Hi there!
David has put up a patch to fix the STUN issues that has plagued Asterisk
1.6 ever since that feature was introduced. Now we need testers to verify
the patch, so please grab the patch (or check out the SVN branch) and add
your comments:
Hi!
I have Asterisk 1.4, trought ODBC I'm savind CDR to MSSQL. How can I
change the fields name of database?
You will want the adaptive CDR backport to Asterisk 1.4:
https://issues.asterisk.org/view.php?id=1
http://svncommunity.digium.com/view/tilghman/branches/1.4/
Philipp
--
Hi!
- upgrade to a current 1.4 version, 1.4.17 is very old (you probably run
this because of the zaptel -- dahdi change, but still)
- do you have a SIP proxy or any SIP-aware hardware in your network
that might play tricks on you, e.g. a SIP ALG (application layer gateway)
on your Internet
Hi!
Three notes:
* as others have already mentioned: personally I would not Dial() from
within AGI using EXEC, but rather set extension and context and then let
the dialplan handle the Dial, and therefore complete that AGI before the
Dial; then possibly run another AGI after the call in the h
Hi!
I'm working for Zoiper, you can contact us directly on supp...@zoiper.com
Zoa
I will do a test call from a soft phone to my mobile. I can speak into
my headset and the audio is heard instantly. But if I speak into my
mobile there is a 1-2 second delay in the Audio. I am using SIP.
We are running asteriskNow 1.4.18 and after a few days it becomes
unresponsive and inbound INVITEs timeout.
Search this list for DNS.
Philipp
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Hi!
Great, but how exactly do i find that channel - that is my question -
which command.
For the third time: Use the M option to Dial() and create a Macro. In
that macro use the SIPCHANINFO() or CHANNEL() function to get what you
want to get. No AGI (and AGI is a protocol while Ruby is a
Hi!
7) if john doe want to speak with caller assistant bridge the two
lines using the transfer function of GXP2000 phone (REFER).
After the transfer in the CDR i can't see the callerid of the caller,
only data of the bridged call is reported.
Any idea on what i can do to keep it ?
Hi!
Depending on the version of Asterisk you are running you can call a macro
or an agi as option to dial. These will be called when the line is
answered and you can find the channel name of who answered.
Do as he says, look at the M option to Dial.
Philipp
--
The problem we are having with Asterisk is when we initiate a call via a
Zap line and it goes out on a Sip line. When it goes out via Sip we hear
no sound until the party we are calling answers the line.
Search for progress and/or progressinband.
--
Hi!
I've worked with these before. They are designed to run a whole
hospital shift, so there should be no worries regarding the battery.
Sounds good. The speaker phone quality is acceptable (the speaker is
quite small and points forward, not upwards in the direction of the ear),
or would
Hi!
I´ve seen them at trade shows, I think I remember it being proprietary.
What about using Dect handsets?
That Star Trek device has always interested me. Too bad they chose WiFi
over DECT, though.
Vocera badge:
* WLAN b/g
* Talktime 2-2.5 hours, standby 20-27 hours
* headset jack
* OLED
Hi!
I've been asked to implement the following transfer workflow in an
asterisk system, and I'm not seeing an easy way to do the bolded steps
below (steps 4 and 5 for those with a text-only email client):
You could create a dynamic meetme room for the 3 legs and drop out when
done. Or do
Hi!
I got some reports of (Debian Testing/Unstable) systems where the
timerfd timing didn't work properly and the workaround was reverting to
the pthreads one. I have not yet managed to reproduce them here.
I wonder if this is the issue.
How about this:
Hi!
2. my SOX (1.14.0) on CENTOS doesn't handle alaw files.
It surely does, only that you need to tell it explicitely to:
Use -t ul or -t al and you are fine.
Philipp
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Hi!
I'm experiencing a problem with my SIP channel's. When I have an
external connection for one of my SIP carrier's, I can listen to the
client and the client listens to me normally. The problem is when I will
transfer this connection, the call is mute for the extension I have
Hi!
Nobody uses chan_local
Absolutely nobody. Except you. ;-
Maybe this will help you: Search for Asterisk timing, consider to not
run Asterisk in a virtual environment, and do not run X on the same box.
Makre sure to turn off silence suppression in your SIP client(s).
Search for
Hi!
client listens to me normally. The problem is when I will transfer this
connection, the call is mute for the extension I have transfered. Only the
client hears normally.
I *think* there is/was an entry in the bug tracker on this. You might
want to search https://issues.asterisk.org (also
Hi!
I am working on calling 2 registrations of same user on 2 different ip or
ports. It works fine and both phones ring simultaneously. the problem is
that there is one way audio, calling party can hear me but i can't hear
calling party.
You need to make sure that these two phones use
Hi!
'Fax for Asterisk' is a commercial application sold by Digium. This is
not their official support channel. Since you paid for the product, why
not contact them directly about your problem?
Maybe because having to deal with Digium support is an ... uncomfortable
experience that I've made
Hi!
Provided the user doesn't have access to the firewall (eg. corporate or
hotel), and the firewall doesn't allow dynamic port opening through UPnP
or NAT-PMP...
For those interested, I was tipped through private e-mail about using
OpenVPN to open a steady tunnel between the client and
SEND 0.0.0.100:5060
?!
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Hi!
Is there a way to log the negotiated codec that was used for each call
in CDR or in a separate log file?
Use CHANNEL(audionativeformat) - and do the same with the help of the M
option to Dial() for the remote call leg. Store that info in the CDR
userfield, or create your own field if you
Hi!
The 'M' in PtMP stands for 'Multi'. Basically PTP is the standard ISDN
protocol, and PtMP is an extension of its logic to make ISDN (BRI) phones
behave somewhat like analog phones: allow you to connect several of them
on the same line.
In other words:
While you *must* have exactly one
Hi!
Has anyone had experience installing it?
yum install asterisk-chan_misdn
I'ts the latest Trixbox Distro version and same issues exists if add in
the Trixbox repo. FAILS as per below
Please search this list for recent messages on mISDN, or Google it.
You will find that mISDN v1 does
When I dial a peer with no registration, SIP channel currently waits
many seconds before returning ${DIALSTATUS} CONGESTION - how can I
shorten this timeout?
Look at qualify=yes for that peer.
Use ChanIsAvail() before you dial.
Use SIPPEER(peername|status) to check registration status.
Use
Hi!
For codecs use CHANNEL function, but you will only get CallLegA
codecs. Without hacking Asterisk, you will not be able to get CallLegB
codecs. Patch for Asterisk 1.4.33.1 attached to get such info.
Thank you! In the meanwhile I found that with the help of the M option to
Dial (macro
Hi!
Sometimes there is a long gap between Asterisk starting and devices
being able to register.
First you should check your DNS setup - it has been discussed many a
times on this list.
Philipp
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Hi!
The network setup is :
analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP
-- other networks
Do it step-by-step: Take the Asterisk server out of the equation, i.e.
call the destination directly with your softphone or the Grandstream ATA
and see if that removes the
Hi!
i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?
Use option a of Authenticate together with ${CDR(accountcode)}
Philipp
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Hi!
Do you already have script to capture user's IP address? If not, check
it here how I am capturing it:
http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-
within-the-dialplan
Or simply use one fo these:
${SIPCHANINFO(peerip)}
${SIPCHANINFO(recvip)}
Hi!
Because the codec is already chosen before the call is made, and you
told it that g722 is permitted.
There are all sorts of discussions in play about codec negotiation,
but at this point in time, if you want different behaviour you'll need to
go and code it yourself
Look at the list
Hi!
Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??
Most probably - who on this list would you like to test it for you? ;-
Philipp
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Well, I¹ve tried this, and something just isn¹t right.
Look here:
Event: Hangup
Channel: SIP/ShoreTel-1-0004
Cause: 17
Cause-txt: User busy
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Hi!
but i want to answer the channel when dial someone and pick up the
phone.not play a file.
Search this list for early media and maybe also for progress.
Philipp
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Hi!
A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip
phones, and we had the opportunity to unregister user by typing *-a
number and -* again, ex * 99 *, and then the phone number/sip extension
was unavailable
It is entirely up to you to design the Asterisk dialplan this
Hi!
Can anyone think of a way to play IVR after conversation initiated by
Dial() terminates?
You will most probably have to prevent the hangup to happen in the first
place:
You could, for example, join the two callers by the help of a dynamic
MeetMe room, and then take action when the
Hey Gilles,
for whatever reason your messages appear twice twice on this list.
Philipp
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Hi!
But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip.
No iax. Why does the asterisk machine have to resolve any address?
Probably because you have one or more register = statements in your
sip.conf and Asterisk is trying badly - but without success - to register
Do this, for example:
exten = 1234567,1,NoOp()
exten = 1234567,n,Set(_PICKUPMARK=${EXTEN:5})
exten = 1234567,n,Dial(SIP/IPphone-1)
exten = 3456789,1,NoOp()
exten = 3456789,n,Set(_PICKUPMARK=${EXTEN:5})
exten = 3456789,n,Dial(SIP/IPphone-2)
[example-pickup]
exten = **XX,1,NoOp()
exten
Hi!
exten = **XX
-- This is a local extension, a certain phone which is monitored with
BLF-lights. So if I press the button I want the phone call that made this
phone ring, not another phone.
This is NOT a local extension: It is a special local PICKUP extension
(you even named it
Hi!
suppose ${SIPaccounts}=SIP/testcorp5SIP/testcorp6
exten = group,1,Set(_PICKUPMARK=${SIPaccounts})
If I was doing this I'd rather do
Set(_PICKUPMARK=group)
or
Set(_PICKUPMARK=${EXTEN})
but that is probably just me. But let's look at two of your lines:
Set(SIP/testcorp4-
Hi!
I understand that SfA is a binary module? There are processors it will not
work on, correct? Are there limits as to operating system or distros?
Requires kernel 2.6, does not work on 2.4 - as I had to find out the hard
way (this is not documented - so embedded people: be aware!).
Philipp
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