[Asterisk-Users] Please Help: Trying to build Asterisk - bazillions of errors
Hi
I followed the instructions on the Asterisk website for download and
building Asterisk. I checked out a fresh copy from the CVS tree as
described and that went smooth, but when I try to build as described, I
get a truckload of errors and I have absolutely no clue what this all
means.
Can anybody please give me some hints or perhaps provide a link to a
pre-compiled version?
thanks in advance and apologies for the incredibly long error log ...
[EMAIL PROTECTED] zaptel]# make clean ; make install
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__
-DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I.
-Wstrict-prototypes -fomit-frame-pointer
-I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include
-I/usr/src/linux/include/net -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP
-DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTANDALONE_ZAPATA -c
zaptel.c
In file included from /usr/include/linux/prefetch.h:13,
from /usr/include/linux/list.h:6,
from /usr/include/linux/module.h:11,
from zaptel.c:35:
/usr/include/asm/processor.h:55: `CONFIG_X86_L1_CACHE_SHIFT' undeclared
here (not in a function)
/usr/include/asm/processor.h:55: requested alignment is not a constant
In file included from /usr/include/linux/module.h:20,
from zaptel.c:35:
/usr/include/linux/modversions.h:1:2: #error
"==="
/usr/include/linux/modversions.h:2:2: #error "You should not include
/usr/include/{linux,asm}/ header"
/usr/include/linux/modversions.h:3:2: #error "files directly for the
compilation of kernel modules."
/usr/include/linux/modversions.h:4:2: #error ""
/usr/include/linux/modversions.h:5:2: #error "glibc now uses kernel
header files from a well-defined"
/usr/include/linux/modversions.h:6:2: #error "working kernel version (as
recommended by Linus Torvalds)"
/usr/include/linux/modversions.h:7:2: #error "These files are glibc
internal and may not match the"
/usr/include/linux/modversions.h:8:2: #error "currently running kernel.
They should only be"
/usr/include/linux/modversions.h:9:2: #error "included via other system
header files - user space"
/usr/include/linux/modversions.h:10:2: #error "programs should not
directly include or"
/usr/include/linux/modversions.h:11:2: #error " as well."
/usr/include/linux/modversions.h:12:2: #error ""
/usr/include/linux/modversions.h:13:2: #error "To build kernel modules
please do the following:"
/usr/include/linux/modversions.h:14:2: #error ""
/usr/include/linux/modversions.h:15:2: #error " o Have the kernel
sources installed"
/usr/include/linux/modversions.h:16:2: #error ""
/usr/include/linux/modversions.h:17:2: #error " o Make sure that the
symbolic link"
/usr/include/linux/modversions.h:18:2: #error " /lib/modules/`uname
-r`/build exists and points to"
/usr/include/linux/modversions.h:19:2: #error " the matching kernel
source directory"
/usr/include/linux/modversions.h:20:2: #error ""
/usr/include/linux/modversions.h:21:2: #error " o Now copy
/boot/vmlinuz.version.h to"
/usr/include/linux/modversions.h:22:2: #error " /lib/modules/`uname
-r`/build/include/linux/version.h"
/usr/include/linux/modversions.h:23:2: #error ""
/usr/include/linux/modversions.h:24:2: #error " o When compiling, make
sure to use the following"
/usr/include/linux/modversions.h:25:2: #error " compiler option to use
the correct include files:"
/usr/include/linux/modversions.h:26:2: #error ""
/usr/include/linux/modversions.h:27:2: #error " -I/lib/modules/`uname
-r`/build/include"
/usr/include/linux/modversions.h:28:2: #error ""
/usr/include/linux/modversions.h:29:2: #error " instead of"
/usr/include/linux/modversions.h:30:2: #error ""
/usr/include/linux/modversions.h:31:2: #error " -I/usr/include/linux"
/usr/include/linux/modversions.h:32:2: #error ""
/usr/include/linux/modversions.h:33:2: #error " Please adjust the
Makefile accordingly."
/usr/include/linux/modversions.h:34:2: #error
"==="
In file included from /usr/include/linux/module.h:297,
from zaptel.c:35:
/usr/include/linux/version.h:2:2: #error
"==="
/usr/include/linux/version.h:3:2: #error "You should not include
/usr/include/{linux,asm}/ header"
/usr/include/linux/version.h:4:2: #error "files directly for the
compilation of kernel modules."
/usr/include/linux/version.h:5:2: #error ""
/usr/include/linux/version.h:6:2: #error "glibc now uses kernel header
files from a well-defined"
/usr/include/linux/version.h:7:2: #error "working kernel version (as
recommended by Linus Torvalds)"
Re: [Asterisk-Users] Grandstream BudgeTone?
Uriel Carrasquilla wrote: They are being distributed by a couple of folks, one being ovislink. I will get you some numbers for contact on monday. I just ordered my BT-100s with Ovislink, so I have got the details at hand ... Tel: +1 (626) 854-1805 Fax: +1 (626) 854-0835 they do have a website at www.ovislink.com but the phones are not yet on there, so you have to order by fax. hth bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Please Help: Trying to build Asterisk - bazillions of errors
thanks everybody who responded. I have reinstalled Linux from scratch using a different source and after that I was able to build Asterisk. thanks again ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gastman -- how do you build it?
Hi after I got asterisk to build, I am now trying to build gastman but it comes up with many errors, mostly complaining that things are missing. are there any instructions anywhere on how to build gastman? further, what are the prerequisites for building gastman? tia bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Webmin module
Hi on the download site for asterisk, there is one directory called webmin and I presume that this contains a webmin module for Asterisk, yet this doesn't seem to be mentioned anywhere. Is this unfinished work in progress? typically webmin modules are in form of .wbm packages that can be installed using webmin itself, but I couldn't find any such module in the webmin directory nor could I work out how to install this otherwise. is there any description how to make use of this? tia bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I make Asterisk login at/use VoIP provider?
Hi please excuse if this seems obvious, but I am new to this and the SIP section in the Asterisk handbook do not give any clues nor do the SIP examples in there seem to represent real-world situations. I am using Nikotel as a VoIP provider (for now) and I would like to configure Asterisk to sign on with Nikotel so that I can use the telephones connected to Asterisk to make calls using the Nikotel service. Checking the preferences in Nikotel's softphone to get a clue for what the settings are, here is what I found: Server: calamar0.nikotel.com Service: nikotel.com I assume that I have to use one of these in sip.conf with the register directive, but which one? Further, if my username at Nikotel is "fred", where do I specify this? Do I specify register => [EMAIL PROTECTED] or register => [EMAIL PROTECTED] ? and where do I specify the password? Do I do register => fred:[EMAIL PROTECTED] (as one would do with a web browser) Also, I don't quite understand the / at the end of the register directive. Do I *have* to specify this? And if I do, what does it do? If I have register => fred:[EMAIL PROTECTED]/ does that mean that only extension can use the Nikotel service? If so, do I have to specify multiple lines with register, one for each extension or can I just omit the / ? Further still, the examples in the handbook only show "friend" but not "peer", I assume I have to define Nikotel as a peer. How would I do this? Do I do ; [Nikotel] type = peer username = fred secret = blah host = nikotel.com or host = calamar0.nikotel.com ? and how does this relate to the above "register" business, why specify this twice? Finally, how do I dial? At present, if I dial 9, it goes straight out on channel Zap1-1 (FXO card). What is the best practise for alternative routes? Should I have it dial via Nikotel if one dials 8 instead of 9 for an outside line? Again, if I wanted to do this, I can't quite see from the examples and the explanation in the handbook how I would do this. I guess the example on page 37 is in principle the way how to tackle this, but there is no example for SIP providers to be used as an "outside line". I guess something like [voip] ignorepat => 8 exten => _8[1-9]XXX,1,Dial (???) exten => _8[1-9]XXX,1,Congestion is in principle how to do this, but how does one dial out using a VoIP service as defined in sip.conf ? Then, is there anything else that needs to be done for VoIP ? Note, my Asterisk is behind NAT, but I am not getting any incoming calls from Nikotel (at least I don't want them as they're prank calls from online voice chat lurkers) so I assume that I don't have to do any port forwarding (it works without that using the softphone). Again, my apologies if these questions seem rather stupid. I'd appreciate any help. Thanks a lot in advance. rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Sacrifice?
Hi is there any ritual sacrifice a newbie has to perform to be welcome on this list? I am new to this whole PBX thing in general and Asterisk in particular. I had hoped that the community on this list would welcome a newbie like myself and help me with some answers to my stupid questions, but somehow it seems to me that nobody likes to respond to somebody who appears to be a complete beginner -- too much bother and a risk to have to explain everything from scratch -- better not answer at all and all that. Well, it may appear that way, but I am not a complete idiot. I know a lot about mobile switching centres, HLRs, VLRs, IN service nodes, mediation devices, billing and settlement systems etc -- I just don't know much about PSTN and PBXes. I would appreciate it if somebody could help me out with a few hints on how to set up my Asterisk box, in particular in respect of VoIP as per my last posting. thank you very much in advance kind regards bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Sacrifice?
Hi Kelly, thanks a lot for your reply. On Friday, July 4, 2003, at 09:40 PM, Kelly McDonald wrote: I'm a newbie myself, but I have at least got * working with a sip provider, encouraging ;-) although the quality was not to my liking, I was hooking up with iconnecthere. Sorry to hear that. I have setup a BudgeTone-100 to connect directly with Nikotel (without Asterisk involved) and I am truly amazed at the quality. For US and European destinations it is just like ISDN. For other destinations it is still significantly better than most low cost long-distance services. I have been trying software based VoIP clients every once in a while for the last five years and always found that the technology was not yet ready for primetime, not even adequate for calling friends and family. Even now, I find that Nikotel's softphone is a disgrace - I have nothing positive to say about it - entirely unacceptable. But what a difference when using a hardphone. I hope the lack of quality you are experiencing isn't because of Asterisk. Perhaps you want to try a different provider. Nikotel don't charge a monthly fee, it's essentially prepaid and the minimum is $15. My roundtrip time to the Nikotel server averages about 170ms and I am on DSL. A friend of mine uses Vonage over the same distance with similar roundtrip times (but different DSL provider) and he is very pleased, too. Here's what I had in sip.conf: [iconnecthere] type=friend insecure=yes port=5060 username=xyz secret=abc host=natrelay.deltathree.com dtmfmode=inband callerid=15408675512 nat=yes in extensions.conf: exten => 8500,1,Dial(SIP/[EMAIL PROTECTED]) thanks, this looks very helpful, I am going to try this after breakfast (it's 5am over here and I just got up). This was just a test so I could dial 8500 and it would call my home phone. Probably have stuff wrong, but it seemed to work. For the rest, extensions.conf has enough stuff in it that you can go and make up your own stuff. Indeed, I will. thanks again regards bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Sacrifice?
Dear Steven, thanks for your reply On Saturday, July 5, 2003, at 12:14 AM, Steven Critchfield wrote: On Fri, 2003-07-04 at 07:23, BK [address only for mailing lists] wrote: is there any ritual sacrifice a newbie has to perform to be welcome on this list? Do not sacrafice your patience on this list. You email complaining of not being answered after only 30 hours that is leading into a holiday weekend for a large portion of the people who do the answering here. Oh dear, perhaps I should have included warning tags like ;-) but seriously, I am sorry if my post offended you. please accept my apologies. I would like to comment though that this wasn't the only post of mine to this list which did not receive any response at all, while others who seemed to ask more difficult questions did get responses right away, so I was starting to wonder if I needed to formally introduce myself or something ;-) Please remember that in the case of a purely VoIP asterisk setup, no one on this list gets paid to support you. Only if you are buying Zapata hardware do the Digium folks get paid for the support, and they even provide you with a preferential way of contacting them for it. I did indeed purchase a couple of interface boards from Digium and I am very grateful for the impeccable support they have provided, such as logging in to our Asterisk box, recompile Asterisk and configure a basic setup so we could test the Zapata hardware. After all they have done, it wouldn't seem appropriate to ask them to also set up the VoIP related stuff for me because it doesn't have anything to do with the hardware I ordered from them. Then again, I have to start somewhere and learn how to do these things myself, in particular as I will be passing on what I learn to others. Asking on the list did not occur to me as demanding support. So be careful to not seem like you are demanding support from a group of volunteers that are either taking time out of their personal life, or out of their work schedule to answer these questions. In fact if you look at the way I asked my questions, you will find that there is no such notion. You will find that I have made an effort on my own first to figure out what to do and the questions I ask are very specific to parts that are unclear to me, such as "what does the / in register do? etc" I did not come here to say "I have this and that, this is my provider, username etc, please provide me with a custom configuration file that allows me to do xyz". I have seen some posts here that had the air of asking along such lines, though, and I wouldn't be surprised if such a post was met with silence. However, I don't believe that my post falls into this category. BTW, whats up with your timezone. It shows +9 on your mail, but you are using a yahoo.co.uk address. Never mind. The time zone is correct. You may want to show all headers on my message and take a look at where it originates from ;-) UK is home base though. And don't get confused if my time zone is going to change to +11 some time, because we'll be building another Asterisk box in OZ when this one over here is done ;-) kind regards bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Information and Pipes
On Saturday, July 5, 2003, at 07:54 AM, Richard Smith wrote: Secondly and the reason for this email is that I'm looking to pipe the CDR information generated by Asterisk into a billing system... Can this be done? Yes, it can be done. If you want to use the CDR's for billing, you should set the parameter "amaflags" in the respective configuration files (ie zapata.conf) to "billing" and issue a reload command on the asterisk console, so it reloads the configuration files. Now, you have two choices how you want Asterisk to give you the CDRs: right into a MySQL database (see cdr_mysql.conf) or into a flat file in /var/log/asterisk/cdr-csv. If you have a billing system that can import the CDRs directly from MySQL, then you will probably want to use the database route, but most billing systems I know expect CDRs in a flat file. Typically, you have what is called a mediation device between the switch generating the CDRs (in this case Asterisk) and the billing platform. In general, a mediation device performs the following functions: Stage 1: call collection read the CDRs from the switch, typically by polling (ie periodically perform an ftp get) Stage 2: integrity check verify the CDRs are not corrupt or incomplete, generate alarms if any are Stage 3: filtering make sure there are no duplicates and omit anything that is not required for billing Stage 4: normalisation convert the CDRs into an output format that downstream systems (ie billing system) can understand Stage 5: distribution provide the CDRs in the proper format to any downstream system (billing system, settlement, fraud) Now, you can either let the billing system poll the CDR files off your mediation device, or you can push them into the billing platform. Either way, it is important to make sure that files are renamed after the transfer is complete so as to make sure that you don't start to process them before they are complete. I have seen the biggest telcos lose hundred thousands of dollars because there was no proper handshake and the billing system would assume that it had reached the end of a CDR file while there were still CDRs being written to it which ended then up unbilled. As to the format that your billing system will want the CDRs to be in, you need to consult the documentation or the vendor of your billing platform. Within limits, this is configurable. As to a mediation device, I only know the market in respect of big telcos and there the typical installation will set you back a million USD and more. There may be solutions for smaller installations at a more affordable price tag, but you may also be able to write your own. I have led many mediation projects for big telcos where we needed an interim solution during the time it took to deploy the solution of the selected vendor. For that we wrote various mediation devices ourselves and strange as it may seem they often performed better and were more flexible than the final solution. In many cases you will find that there are code snippets for parsing CDRs in open source libraries such as the PERL archive but also we came across various Java libraries for CDR processing. hope this helps regards bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi
thanks to everybody who responded to my earlier post. I have looked at
all the material and links provided and tried everything in there, but
it simply won't work for me.
My SIP phones register with Asterisk, but they cannot be called
(everybody is busy at this time) nor can they call anything (error code
4, whatever that means) not even internal (yes I did give them
appropriate context).
Further, Asterisk registers with my VoIP provider via SIP just fine, but
I cannot make any calls even from the analog phones.
sip show registry gives me
HostUsernameRefresh State
63.214.186.6:5060 myusername 120 Registered
sip debug also confirms successful registration.
I wonder what the syntax is to dial a number via a VoIP provider. This
appears to be documented NOWHERE.
I tried this:
; International long distance through VoIP service
;
exten => _00N.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr
exten => _00N.,2,Congestion
and sip debug tells me that the account doesn't match the one on record,
whatever that means.
I tried this:
; International long distance through VoIP service
;
exten => _00N.,1,Dial,SIP/[EMAIL PROTECTED]/${EXTEN:2},tr
exten => _00N.,2,Congestion
and this doesn't even show anything but immediately gives me a busy
signal. The fact that there is no debugging output leads me to believe
that Asterisk didn't even attempt to try talking to the VoIP server.
Does anybody know how to dial a PSTN number through a VoIP service?
Is this standardised, at least within SIP? Or does it vary from provider
to provider?
any hints appreciated
kind regards
bk
___
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Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi
thanks for your reply.
On Sunday, July 6, 2003, at 03:44 AM, John Todd wrote:
sip show registry gives me
HostUsernameRefresh State
63.214.186.6:5060 myusername 120 Registered
sip debug also confirms successful registration.
The command you will find more useful is "sip show peers".
show peers returns
Name/username HostMaskPort Status
Sip1/gsone 192.168.0.160 255.255.255.255 5060 Unmonitored
Sip2/gstwo 192.168.0.161 255.255.255.255 5060 Unmonitored
nikotel/myusername 63.214.186.6255.255.255.255 5060 Unmonitored
I don't think the problem is with registration. As I said, sip debug
confirms successful registration:
Sip read:
SIP/2.0 200 OK - 'Authenticated'
Via: SIP/2.0/UDP 192.168.0.111;branch=weoiuewotu
From: ;tag=xznxzcvnxzvn
To:
CSeq: 121 REGISTER
Call-ID: [EMAIL PROTECTED]
Expires: 120
Content-Length: 0
Contact:
If your hosts are "(Unspecified)" then your SIP clients are not
registering, and inbound calls will not work if you are using
"dynamic=yes" in your sip.conf. Possibly it may be helpful if you
would statically register your SIP phones until you get things working
better ("host=10.3.2.3" in sip.conf)
I had already done that. Part of the problem seems to have been the
length of the identifiers I had defined.
[Grandstream1]
username=grandstream1
appears to have been too long, so I shortened it to
[Sip1]
username=gsone
Now I can use the SIP phones internally at least. I can call from Sip1
to Zap2 for example. Dialing into the PSTN from the Sip phones is still
problematic, ie calls dropping, tiny volume etc etc.
However, for as long as I cannot even dial into Nikotel (nor FWD nor
ICH) using one of the analog phones on (Zap2 or Zap3) I don't even fancy
to fiddle with the SIP phones. The fever things there are in the call
chain that could be responsible for it not working, the better, so for
now, I want to concentrate on getting the dialing out via VoIP service
provider working from the Zap lines.
I wonder what the syntax is to dial a number via a VoIP provider. This
appears to be documented NOWHERE.
I would disagree. A VoIP provider is no different than a SIP phone;
they are treated the same.
If I dial into a SIP phone directly, I don't have to provide a third
party number because the SIP phone is the destination already. If I dial
into a VoIP service, I have to provide the number for my desired
destination in addition to my own credentials. As a result the two cases
are not the same as far as dialing syntax is concerned.
If you are looking for examples, please see
http://www.loligo.com/asterisk/ for my sample files, which contain
some VoIP provider dial statements.
Thanks, I had already been given the URL and I looked at it. One of the
problems I had is that I find it difficult to work out what the *naked*
dial string actually is because of all the macros and variables used in
there.
I would prefer to start as barebones as it can possibly be and get the
basics working. Bells and whistles can be added in later - one at a time.
exten => _00N.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr
exten => _00N.,2,Congestion
and sip debug tells me that the account doesn't match the one on
record, whatever that means.
I tried this:
; International long distance through VoIP service
;
exten => _00N.,1,Dial,SIP/[EMAIL PROTECTED]/${EXTEN:2},tr
You may be having at least one error due to syntax. The line above
should look like:
exten =>
_00N.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:2},100,r)
Thanks for the hint. However, I am not quite so sure that this is the
correct syntax either. When I try this with "sip debug" Asterisk does
not even make an attempt to contact the remote VoIP service. It chokes
on the dial string already and goes no further.
SIP Registration comes back with a SIP/2.0 200 OK (ie registration
successful) for both ICH and Nikotel. Thus, I am confident that
registration is not the issue. However dialing out (from Zap) with the
various possibilities of syntax I could think of, simply doesn't work.
Here is what I have tried so far
in zapata.conf ...
-
signalling=fxo_ks
context=internal
channel => 2,3
-
in sip.conf ...
-
register => user:[EMAIL PROTECTED]/asterisk ; already tried with
user instead of asterisk and blank
[nikotel] ; Service Provider Nikotel
type=peer
secret=pass
username=user
host=calamar0.nikotel.com
-
in extensions.conf ...
-
[globals]
REDPHONE => Zap/3
[voipintl]
;
; International long distance through VoIP ser
Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi Paul, thanks for your insights On Monday, July 7, 2003, at 03:59 PM, Paul Cheng wrote: To dial a PSTN number through Nikotel used to work from Asterisk, but they had a very serious security issue (you could make calls anytime anywhere and their billing wouldn't charge it) and after I informed them of this, they changed their authentication mechanism and since then I have not gotten it to work (they didn't even thank me!). This is what we have discovered last night. However, We have got it working now. I will document this in detail and make it available, but briefly here a quick summary ... First I had various glitches in my dial string. With the help of John Todd and some others on the IRC #asterisk channel I was able to fix those glitches. Thanks everybody who assisted. Then I tried a number of things I had already experimented with before. When I turned on SIP debug and watched the datagrams, I could see Nikotel's response "account name does not match address of record". Together with the "from" part, this led me to fiddle with "fromuser" again and when I set it to the actual login name, it worked. Their tech people said it should work with a slight change: "yes, we changed it yesterday. Now the user part of the From: address has to be the same as the username in the Proxy-Authentication line. I don't know if the Asterisk can do that. The ATA186 does it b[y] default." This CAN be done if you edit chan_sip.c, It would seem you can do it a lot simpler: in sip.conf - register => myusername:[EMAIL PROTECTED] [nikotel] username=myusername fromuser=myusername ... - but when I did this, it billed me a few times for unconnected calls Thanks for sharing this with us. I will watch this for a while and see if this happens here too. and I gave up trying to debug and switched to iConnect. iConnect is worse quality, but it is very easy to connect to. I had much better quality with calls via Nikotel than iConnect, but their support is non-existent/bad at best. I sent them 3-4 e-mails about their security issue before they even responded. Yes, support is not exactly their strength, is it?! FYI. Registering with Nikotel was futile anyways, because I never figured out how anyone could call into me. I don't want anybody to call in via Nikotel. Since they do not provide a telephone number for incoming calls, the only calls you could possibly get are from their public chat room. In the very best case you get a friendly test call from somebody who has just signed up and wants to try out the service, in the worst case you get prank calls in the middle of the night or indecent proposals and all the rest of it. I will have to find a way to disable incoming calls from Nikotel entirely. iConnect provides a PSTN-SIP dial in as an option, but I haven't tried it. Yes, I have seen that. And at $8.95/mth it would seem reasonably priced, too. Outbound calls do not require registering. I can provide examples of iConnect connection scripts if you contact me offline. Thanks, I will do that. again many thanks to everybody who has helped solving this riddle rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Follow-up -- Using Asterisk with Nikotel
Hi
thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service.
I have drafted a mini-how-to which is available at
http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf
This is a first draft, I will amend this further, in particular the "verify and debug" section which is blank right now.
here is a plain text summary:
in sip.conf ...
--
; SIP Registration with Nikotel
;
register => myusername:[EMAIL PROTECTED]
; SIP peer definition
;
[nikotel]
type=friend
secret=mypassword
auth=md5
username=myusername
fromuser=myusername ; IMPORTANT! Nikotel requires this!
host=calamar0.nikotel.com
--
in extensions.conf
--
; Extensions
;
[globals]
CC => 81 ; Local country code (here Japan)
PSTN => Zap/1 ; incoming NTT line
REDPHONE => Zap/2 ; internal analog line
FAXPHONE => Zap/3 ; internal fax line
IPPHONE1 => SIP/Sip1 ; IP phone #1
IPPHONE2 => SIP/Sip2 ; IP phone #2
; International long distance through VoIP service
;
[voipintl]
exten => _00N.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r)
exten => _00N.,2,Congestion
[voipnatl]
;
; National long distance and local throuh VoIP service
;
exten => _0N.,1,Dial(SIP/${CC}${EXTEN:[EMAIL PROTECTED],60,r)
exten => _0N.,2,Congestion
[international]
include => voipintl
[national]
include => voipnatl
[incoming]
exten => s,1,Dial(${IPPHONE1}&${IPPHONE2),20,r)
[internal]
exten => 1001,1,Dial(${IPPHONE1},20,tr)
exten => 1001,2,Congestion
exten => 1002,1,Dial(${IPPHONE2},20,tr)
exten => 1002,2,Congestion
include => national
include => international
--
regards
bk
[Asterisk-Users] Asterisk crashing after Voicemail box creation
Hi
I have just been struggling for four days to get SIP working and now as
I created a voicemail box, Asterisk has become very unstable and it
can't bridge SIP phone to SIP provider calls anymore.
Calling internally from one SIP phone to another works fine.
Calling internally from a SIP phone to an analog phone on a Zap channel
and vice versa works fine.
Incoming PSTN calls delivered to a SIP phone also works fine.
Dialing out from an analog phone on a Zap channel using a SIP provider
works fine as well.
HOWEVER,
when dialing out using a SIP provider (both Nikotel and iConnect)
Asterisk cannot bridge the two legs of the call and all I get is silence.
here is what the console shows:
-- Executing Dial("SIP/Sip1-1862", "SIP/[EMAIL PROTECTED]|60|r")
in new stack
-- Called [EMAIL PROTECTED]
-- SIP/nikotel-4815 is ringing
-- SIP/nikotel-4815 answered SIP/Sip1-1862
-- Attempting native bridge of SIP/Sip1-1862 and SIP/nikotel-4815
== Spawn extension (internal, 00442071231234, 1) exited non-zero on
'SIP/Sip1-1862'
NB: PSTN number edited
I have stop/started Asterisk and even rebooted the machine, but no
change.
This problem has popped up after I created a voicemail box. When I
tested the voicemail, at the moment when I tried to listen to the
recording (VoicemailMain) Asterisk crashed. After restarting, I can now
get into VoicemailMain without a crash, but there is now a problem with
SIP and Asterisk crashes once in a while.
Any ideas?
thanks in advance
rgds
bk
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Re: [Asterisk-Users] Lot's of errors and warnings.
On Wednesday, July 9, 2003, at 12:15 AM, marrandy wrote: [about the kernel sources on Mandrake 9.1] ZNot yet sure why it wasn't installed on this machine. Will have to look into that. This would seem to be a Mandrake thing. I had the same problem. The Mandrake installer would not install the kernel sources when doing a CD based installation. Although the kernel sources are on CD3, the package manager interface doesn't seem to find/show them. I did an FTP based install instead in order to get a complete system with kernel sources. rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using multiple iconnecthere accounts
On Wednesday, July 9, 2003, at 02:46 AM, Derek Beaumont wrote:
exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
What does EXTEN:1 do? Why is StripMSD not used?
EXTEN:1 expands into the extension dialed without the first digit,
that's why you don't need StripMSD.
rgds
bk
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[Asterisk-Users] ignorepat doesn't work
Hi
in order to keep the dial tone after pressing 9 for 'outside line' I
have this in my extensions.conf
[localpstn]
ignorepat => 9
exten => _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1}
exten => _9[123456789]XXX,2,Congestion
this is properly included in the handsets' context but the dial tone
disappears after pressing 9.
am I missing something?
thanks in advance
regards
bk
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[Asterisk-Users] It's true - Nikotel charge for not-completed calls
Hi A few days ago, Kelly remarked that he had previously observed that Nikotel charged him for calls he did not actually complete. I have made a number of test calls to my landline without picking up the calls. I just let it ring once and hung up on the calling phone. A look at the call records on MyNikotel reveals that I was charged six seconds for every of these calls. I have raised a support request with Nikotel to ask for this to be corrected. However, Kelly said he had done so before and never got any reply. Therefore, I would like to ask everybody on the list who is using Nikotel to verify this for themselves and raise a support request on Nikotel's website. Together we may be able to get this fixed. thanks regards benjamin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat doesn't work
thanks On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that "ignorepat" only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the "ignorepat" in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk
[Asterisk-Users] callerid= being ignored
Hi
I have defined my SIP phones like this ...
[Sip1]
username=gs1
callerid= "Full name" <1001>
etc etc
Now, when I do this in a given extension
exten => ,1,NoOp(${CALLERIDNUM})
then I get "" as callerid and not "<1001>" as defined with callerid=
Sure, I could set the usernames to their respective extensions, but I
don't want to do that. I'd like to keep login names independent of
extensions.
Is there any fix for this so that the real callerid shows up?
thanks in advance
regards
bk
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[Asterisk-Users] incoming callerid on FXO
Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the callerid and so I guess that it is just a matter of configuring the FXO card so that it uses the right method for detection. I seem to remember that I read somewhere that this can be changed but I can't seem to find any reference to that now. Does anybody know how to change the method for detecting callerid? thanks in advance regards bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid= being ignored
On Thursday, July 10, 2003, at 01:31 AM, Martin Pycko wrote:
At the moment asterisk can get the callerid from the "From: " field.
Thanks,
I tried fromuser=1001 in the [Sip1] section and CALLERIDNUM still
returns ""
What is the "callerid=" directive good for if not setting the CALLERID,
it doesn't seem to make any sense.
Does anybody know how to fix this?
thanks in advance
regards
bk
On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:
I have defined my SIP phones like this ...
[Sip1]
username=gs1
callerid= "Full name" <1001>
etc etc
Now, when I do this in a given extension
exten => ,1,NoOp(${CALLERIDNUM})
then I get "" as callerid and not "<1001>" as defined with
callerid=
Sure, I could set the usernames to their respective extensions, but I
don't want to do that. I'd like to keep login names independent of
extensions.
Is there any fix for this so that the real callerid shows up?
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Re: [Asterisk-Users] It's true - Nikotel charge for not-completed calls
On Thursday, July 10, 2003, at 02:59 AM, Jon Pounder wrote: Guys, unless their site states something to the contrary you don't have a hope in hell with this. you are paying them for a voip circuit, which you are using to attempt a call. you have taken up several seconds of voip bandwidth which they are charging you for, the same way you would pay if there is 6 seconds of silence during your phone conversation. (This takes roughly the same amount of transfer as an unsuccessful call setup.) By the same reasoning, if you have some strange audio in your conversation which does not compress well, they don't charge you extra since it costs more to deliver your call than you are paying. The price is based on averages. Do you complain at an all you can eat buffet if you get full after a small plateful ? Same concept. First of all, nobody complained. As far as your argument is concerned, it can be dismissed very easily because of a) common sense; and b) about 120 years of telephony history during which it has been customary to charge for payload not for effort VoIP is not going to change this well established business practise If Nikotel gets enough requests, they are likely to rectify their billing, if not, sooner or later some competitor is trying to get a competitive edge by making it their sales pitch that they don't charge for unconnected time. Just like by-six-second billing or even by-the-second billing has eventually won over by-30-seconds or by-the-minute billing. It is that simple. Your reasoning notwithstanding. rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to modify dialed number?
On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote:
Is there simple way how to add prefix to dialed number?
I need change 0X. to 0X.
how about this
exten => 0X.,1,Dial(0{EXTEN:1})
rgds
bk
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Re: [Asterisk-Users] How to modify dialed number?
On Thursday, July 10, 2003, at 08:13 AM, Martin Pycko wrote:
You forgot about "_" in front of 0X
Indeed, and I also forgot to put a channel (Zap, SIP, IAX ...) in the
dial string.
Oh dear, I guess I have to go to bed a bit earlier ;-)
rgds
bk
On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:
On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote:
Is there simple way how to add prefix to dialed number?
I need change 0X. to 0X.
how about this
exten => 0X.,1,Dial(0{EXTEN:1})
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[Asterisk-Users] What does "callerid=" in sip.conf do?
Hi since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? thanks regards bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird experience with MOH
Hi I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there was only silence when putting a caller on hold. Somebody told me I may have the wrong mp3 app (321 vs 123) while I was getting busy with something else and so I put this aside. Although I found that I did have mpg123 installed. Yesterday, I copied some music files into /var/lib/asterisk/mohmp3 in anticipation that I would get this to work eventually and to my surprise, putting a caller on hold now plays the music. I have no idea why it didn't work at first, but it would seem that for some unknown reason, Asterisk didn't like the sole sample-hold.mp3 file. rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird experience with MOH
On Saturday, July 12, 2003, at 01:39 PM, Matthew Hardeman wrote: If you're on a RedHat system, mpg321 is installed by default, and is symlinked to as mpg123... So, it can easily look like you have mpg123, but you really have mpg321... Interesting thought, but according to the man pages I really do have mpg123. Also the package manager shows me that mpg123 is installed and mpg321 is installable. BTW, I am running Mandrake 9.1 not RH. Besides it's now working without installing anything since when I had the problem that it didn't work. Anyway, I found that neither a reload nor a shutdown and restart of Asterisk will actually cause Asterisk to recognise any changes in the directories where the mp3 files are. I have added in more and more mp3s now but Asterisk continued to loop through the initial three I put in there before I noticed that MOH is actually working. It seems you have to change the class in musiconhold.conf to another class with a different path, do a reload and then change it back, do another reload in order for Asterisk to take notice of additional files. This behaviour offers - though not a full explanation - but a hint as to why my MOH didn't work initially. There may have been something going on - such as some cache timing out - that made it eventually kick in. rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does "callerid=" in sip.conf do?
Hi thanks everybody for your replies on this thread On Saturday, July 12, 2003, at 05:01 PM, The Traveller wrote: since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? Works perfectly over here. My guess is that the calls you make with that phone aren't coming in over the user / peer-entry you're expecting, for some reason. Everybody seems to say its working for them, but have you actually set your [phone] entries in sip.conf such that the username is different from the caller ID? I have come across various example configs which suggest that you define the extension number as the username, but I don't want the username to be tied to a particular extension. One of the merits of IP telephony is that you get away from the static assignment of account identifier and trunk number. Thus, if your sip.conf looks like this ... [phone1] username=1001 callerid="Fred Flintstone" <1001> then of course you'd see 1001 as caller ID. But I did this ... [flintstone1] username=fredf callerid="Fred Flintstone" <1001> and I get "" as callerid. When I asked about this here earlier, someone said that Asterisk can take the callerid from the "from" field which led me to believe that I could fix this by doing ... fromuser=1001 but this didn't have any effect at all. I even tried variants such as fromuser= <1001> and "1001" and "Fred" <1001> but no luck. any ideas how I can fix this without having to resort to setting the usernames equal to the extension numbers? thanks rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIP 30 phone
On Sunday, July 13, 2003, at 12:01 AM, Darren Poulson wrote: I'm just learning about VoIP and Asterisk. I've got a developers kit on its way and I've managed to get hold of a couple of cheap Cisco VIP 30 phones. I've trawled the web and found a few snippets of information on these phones but I still can't get them to work. Does anyone have any config files or any idea on how (if I can) to get them to work? Many Cisco phones can be loaded with different firmware to support different protocols. Have you been able to establish with certainty which protocol those phones of yours speak? Also, it may be helpful if you post the relevant snippets of your configuration files. Further, some clues from the debug info may be helpful to get an idea what's going on. You know that you can watch the packets on the console by turning debug on (ie "sip debug" for SIP) don't you?! rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
