The error lies here:
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory. Stop.
do you have the kernel-headers installed? (e.g.
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora)
Alyed
I've made a very simple one time ago I
could share with you, it's made on bash, takes as input a CSV file,
places the calls using the /var/spool/asterisk/outbound directory, and
restricts the number of calls to a given number at a time (say 10)
I can share it with you only if you
Had the same issue time ago, but Eric shed good light on it,
have a look at:
http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html
Summary: sorry, no nice work around.
Alyed
Return-Path: [EMAIL PROTECTED]
Carlos: Had you tried re-compiling wanpipe?
Had a similar problem, and eventhough I'm pretty sure I compiled
wanpipe and it did the zaptel patch succesfully, once I finished with
the Unicall installation, somehow the patch was not wotking correctly.
So after a couple of days of
Hello!
I'm tryuing to bring up an R2 connection but eventhough I've followed
the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems
to be missing.
When an incomming call is generated I get:
Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report:
First look if you have the libidn.so.11 library. if you don't
then install it, otherwise you can simply copy-paste it into the /usr/lib
folder where Asterisk is looking for it or make a symbolic link to it.
Alyed
Return-Path: [EMAIL
Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is when dialing to the Harris PBX, it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing folder I
Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is that when dialing to the Harris PBX it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing
If your Asterisk is behind a NAT you can use externip=x.x.x.x (sip.conf)
If your Sipura is behind a NAT you can use nat= yes (sip.conf)
Btu I'm really afraid that unless you use a SIP
proxy (e.g. Portaone) you won't be able to succesfully connect both
elements if they both are behind
My advice is to first make some tests to see if a reload is enough for Asterisk to read any group definitions change in zapata.conf, otherwise no on-the-fly change will workAlyed
Return-Path: [EMAIL PROTECTED] Mon Oct 30 13:23:36 2006Received: from
Isn't your problem more about NAT traversal rather than the phones themselves?if so better use some iax softphone, have a look at: http://www.voip-info.org/wiki-VOIP+Phonesof course you can use SIP based hard/soft phones but using iax based ones is cheaper and faster.Alyed
Echo is generated by the analog end to where you place the call, not the IP side of it.
As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it)
I'm afraid there is little you can do to here.Alyed
you can also try using
busydetect=yes
busycount=4
in your zapata.conf
Hopefuly you won't start getting sudden hang ups, due to false positives and it will be helpful enough.
Alyed
Return-Path: [EMAIL PROTECTED] Tue Oct 17 14:30:11 2006Received: from
calling search space, and then give them access to
each other you can do that.In the attachment, I circled the calling search space field I see on my Add NEW SIP TRUNK PAGE.Hope this helps. -- Original message ------From: "Alyed Tzompa" Many thanks for
Hi!
I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in
http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration
but still not able to make Asterisk communicate with Cisco. I keep on receiving ---
SIP/2.0 400 Bad
risk trunk on the Call manager is in the same "calling
Search Space" as the phones are in, or make sure there is access
between the "calling search spaces"-Eric -- Original message ------From: "Alyed Tzompa" Hi! I'm trying to communicate a C
What I want is to transfer some calls to a Cisco extension, so think I don't need to do the upgrade to CM5.I'm I right?AlyedOn Tue, Oct 10, 2006 at 01:21:28PM -0500, Lacy Moore - Aspendora wrote: I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a
Be careful when using heavily ChanSpy. We
did couple of weeks ago and the result was having Asterisk crashing
almost once every day. How heavy? around 4 people using it 8 hours a
day, each one using ChanSpy every 3-5 mins.
we were not able to find the exact reason, so just stop using
I'm experiencing the same problems, but
unfortunatelly haven't been able to associate them with any number
since they appear to be random. But maybe we can do a little research
about it, and hopefully find teh solution for both:
are your PSTN lines POTS or E1/T1? can you make a couple of
I'm curious... why will this work??
busydetect will just cut the line if there are 4 tones (les or more
depending the busycount param), and call progress will in fact try not
to cut the call due to false hangups.Alyed
Return-Path: [EMAIL PROTECTED] Wed Sep 27 16:12:13
Hi guys!Can someone give advice on nice H323 IP phones brands?? I'm looking for some H323 IP phones for a customer. Diving in theinternet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard about them. Can someone give feedback experince about it??, configease, sound quality, visual
Hi guys!
Can someone give advice on nice H323 IP phones brands?
??
I'm looking for some H323 IP phones for a customer. Diving in the
internet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard
about them. Can someone give feedback experince about it??, config
ease, sound
;Thu, 21 Sep 2006 11:27:21 -0700Received: by nz-out-0102.google.com with SMTP id z6so390195nzd
didn't work :(Regards,SantiagoOn 9/20/06, Alyed Tzompa <[EMAIL PROTECTED]>wrote: Not an expert at reading Polycom config files, but guess g729 and ulaw are both preference 1 isn't it? hey... yo
in order to makeg729 the first choice:voice.codecPref.G711A="3" voice.codecPref.G729AB="1"voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2"voice.codecPref.IP_4000.G729AB=""/Cheers,SantiagoOn 9/19/06, Alyed Tzompa wrote: Make su
Since the phone is the one behind a NAT,
and the registration is done only with SIP packages, setting or not the
"nat" is not an issue (ONLY for registration purposes). You can see
this since Asterisk is receiving the registration. Why is it denying
it?... wel, that's something that will
We can help guy, but pls send detailed info regarding a specific problem. The more info you provide, the best we can help.
As for g729 codec instructions look at
http://www.digium.com/en/supportcenter/documentation/viewdocs/G729
there you'll find detailed and updated information about it.
Make sure the codec used by the Polycom
will be only g729 via the phone's web interface, as far as I remember
Polycom will try always to use ulaw or alaw first unless it is
configured to use only or as first choice the g729 codec.Alyed
Return-Path: [EMAIL PROTECTED] Tue
Hi there!
I'm having lots of problems with an Asterisk used by a customer.
Got hundreds (yes hundreds, about 3-4 per minute) of this messages
every hour:
WARNING[12685] channel.c: Avoided initial deadlock for '0x96dee78', 10 retries!
The hex number changes with every message. The warning
from
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
qualify=xxx|no|yes
where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds.
If you turn on qualify in the configuration of a SIP device in sip.conf,
Could you please explain what the network configuration you want to try? it would be really helpful.
you can be as simple as: SIPphone-- internet -- NAT-- asterisk
or whatever your particular scenario is.Alyed
Return-Path: [EMAIL PROTECTED] Mon Jul 31 11:43:16
As far as I there is no free softphone that can handle G729 codec. So you will need a licenced one.
Have a call center working with eyebeam from counterpath (previously
known as Xten) for about a year with no problems. Don't know if it
supports
the URL option, but I'm pretty sure it will.
first: download the latest version 1.2.5 had some bugs and is already several months old.Depending on how you want your asterisk to behave will be the amount of files you'll need to mess with. Let's say you want a very basic installation with some SIP phones (hard or soft), then you'll have to
As of Asterisk 1.0.X a "#" was recognized as a pattern not as a digit, hence in order to use it at the begining of an extension you should use "_" before it. I guess this is still valid in 1.2.X versions.i.e: use _#31#0046011 in your extensions.confAlyed
Return-Path:
like if some one speaks loud, he would get a low volume.I'm sorry, but this goes far beyond Asterisk (at least for the moment) :)Anyway you can still play with rxgain and txgain in zapata.conf, but this will increase/reduce the overall volume gains and can also affect echo perception.Alyed
You have a little confusion:
friend = can GENERATE and RECEIVE calls
peer = can only GENERATE calls
user = can only RECEIVE callsAlyed
Return-Path: [EMAIL PROTECTED] Fri Jul 07 09:27:13 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number (CallerID)into an "Additional calling party number". Guess it
refeers to a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf
2 things might worth having a look:
a) set up in your zapata.conf:
musiconhold=default
b) You say the asterisk version is 1.1, but 1.1 is developement
version, maybe was just a typo, but you should be using either a 1.0.X
or 1.2.X versionAlyed
Return-Path: [EMAIL
Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number into an "Additional calling party number". Guess it
refeers to a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf but
have
Have you tryed phoning a fixed line instead of a cell phone?is this giving the same result?I assume your outgoing call to a the cellphone goes through a Zap channel. Try another one (e.g. Zap channel 2), and let us know the result.Alyed
Return-Path: [EMAIL PROTECTED] Sat Apr 01
I used g729 couple of times in the past and got the warning messages ONLY when I was trying to use more channels than the total amount of licenses I'd got.If you are sure you are using only one device that needs the license, I would suggest to check out how it is communicating with Asterisk.
That was a bug fixed in Asterisk version 1.2.3 recently version 1.2.6 was released, so don't worry you can try the latest one without timing fears :DAlyed
Return-Path: [EMAIL PROTECTED] Sat Apr 01 15:42:39 2006Received: from digium-69-16-138-164.phx1.puregig.net
I use Portaone's PortaSIP for everything related to LCRAlyed
Return-Path: [EMAIL PROTECTED] Wed Mar 29 16:48:54 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Wed, 29 Mar 2006 16:48:54 -0700Received: from
I may add a very nice configuration:
- Use two (or more) Asterisks to create your own VoIP network
Very useful if you have broadband and several facilities spread out in distant geographical locations.Alyed
Return-Path: [EMAIL PROTECTED] Wed Mar 29 16:32:16 2006Received:
As I understand this, it's a problem of redirecting the call to the same FXS channel.
To replicate this behaviour in the Asterisk you could try the following in the extensions.conf:
(suppose your FXS channel is group 1 in zapata.conf)
exten = 100,1,Dial(Zap/g1/${EXTEN},20)
exten =
Think a zaptel recompile is just what you need.Alyed
Return-Path: [EMAIL PROTECTED] Thu Mar 23 17:05:27 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 23 Mar 2006 17:05:27 -0700
i've got a
Polycom's can work in one of two ways:
a) using self configuration
b) downloading it from a ftp server
To make your Polycoms work with Asterisk you actually don't need the
phone to download any configuration, with the one embeded is ok. In any
case, when turned on, the phone searches for
if you want to use the defaults unavailable and bussy just put it in /var/spool/asterisk/voicemail/default/100/unavail.wav
The "100" extension will automatically be created after you leave your
first voicemail message, change "unavail.wav" for "busy.wav" to use
them as
Voicemail(u100) and
Have a customer running some 25-28 concurrents calls (with about 35 agents logged in)without problems with a P4 2.X Ghz, 1GB RAM,I'm doing no transcoding btw.Alyed Return-Path: [EMAIL PROTECTED] Sat Feb 04 16:59:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
Also get the book (again I dont have the URL if some one does please post it). http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Alyed If you are new I would reccomend using [EMAIL PROTECTED]http://asteriskathome.soundforge.net . It is a greatresource for beginers. Also get the
I would be useful if you could post your config files and the pri debug as well. check your zapata.conf or paste it here so we can take a look.AlyedReturn-Path: [EMAIL PROTECTED] Thu Jan 12 10:04:28 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP. just add the range ports tih a ":" e.g 192.168.1.2 1 : 10007(4)Please,I know alot of
I had a very similar problem some months ago, was using a Sangoma A101 card though. The problem was something related to the card's memory and was able to solve it by updating the driver. It was caused due to I was using a brand new card with a not so updated driver (I was using one that I thought
can't you ask the users to dial a prefix? that can solve your problem. btw, which provider are you using for your calls to the USA? Alyed Return-Path: [EMAIL PROTECTED] Wed Jan 11 09:47:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with
Don't know if you can actually adjust the volume in any of them, but you can try from the asterisk with rxgain / txgain in your zapata.confAlyed Return-Path: [EMAIL PROTECTED] Mon Jan 09 16:27:24 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
== Primary D-Channel on span 4 down for TEI 65 == Primary D-Channel on span 4 down for TEI 64 == Primary D-Channel on span 4 down for TEI 66 This looks like a signaling problem, check out your configuration in zaptel.conf and the log messages you get at /var/log/messages You can also post them
I don't find the console output ugly, maybe messy, but never ugly :P If u don't like those NoOp, just take them away from ur extensions.conf. BTW, to save the console output to a given file, just edit your logger.conf file. Say you only want the console output, then just add to your filename the
tried using: mylogfile = verbose in logger.conf but all I got was the startup/shutdown asterisk messages. Besides, this isn't what I wan't. I don't want Asterisk internal generated log messages. I want my OWN log messages, that I specify. Doug-Original Message-From: Alyed Tzompa [mailto
. -Original Message- From: Alyed Tzompa [mailto:[EMAIL PROTECTED] Sent: Thursday, January 05, 2006 11:59 AM To: Douglas Garstang; asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk Debugging Then stop looking for easy solutions and get your hands dirty changing your c
r 1" 300 host=dynamic nat=yes dtmfmode=INFO mailbox=300 disallow=all allow=alaw allow=ulaw allow=g723.1 allow=g729 Many thanks, Aisling. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alyed Tzompa Sent: 04 January 2006 00:28 To: asterisk-user
made the changes in sip.conf so now it reads: disallow=all allow ilbc now I when the call is placed it is not hanged up, but I cannot hear anything. I think it's becasue Asterisk is sending the RTP's to a wrong address (my internal IP). Looked at the sip debug and got the following: --
, 2006 at 12:09:33PM -0700, Alyed Tzompa wrote: You are not gonna be able to modify this behaviour from the asterisk since in your case asterisk is only receiving the digits from someone else (an Avaya in your case but could be PSTN for instance) Just asked an Avaya support guy and told me you
References: In-Reply-To: Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-SmarterMail-Spam: SPF_None Alyed Tzompa wrote: sip.conf [general] port=5060 externip = www.theip.net localnet = 192.168.1.0 localmask = 255.255.255.0 allow=all Don't use
t know who can I address it in the right way. I'm using the default STUN config in the SJphone : STUN server address -- stun.softjoys.com :3478, refresh time out --120 conclusive response timeout--0 retrunsmissions number -- 13 and nat= yes in the sip.conf But still no sound in my endp
iLBC even in the free version ? Alyed Tzompa wrote: made the changes in sip.conf so now it reads: disallow=all allow ilbc now I when the call is placed it is not hanged up, but I cannot hear anything. I think it's becasue Asterisk is sending the RTP's to a wrong address (my internal IP). Looked
Just type in the asterisk command line: show channels or sip show channels type "help" also to take a look at the other commands availableAlyed How do you check whether a channel is active and the number of calls on it?Is it simple and
I'm a bit confused on how you get your calls to Asterisk, what I mean is: are you phoning into asterisk via a sip user? in this case, which one?, if not is it iax or though a zap channel?anyway, here some tips:For your first problem it seems it has to do with what I pointed above, check that the
You are not gonna be able to modify this behaviour from the asterisk since in your case asterisk is only receiving the digits from someone else (an Avaya in your case but could be PSTN for instance)Just asked an Avaya support guy and told me you should take a look at the ARS Digit Analysis Table,
Hi everyone My problem is the following: I'm trying to make a call from a sip phone (SJphone) behind a Restricted Cone NAT towards and Asterisk behind another NAT(a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the remote Asterisk. My soft phone registers without problems
Hi everyone My problem is the following: I'm trying to make a call from a sip phone (SJphone) behind a Restricted Cone NAT towards and Asterisk behind another NAT (a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the remote Asterisk. My soft phone registers without
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