[asterisk-users] Security
Dear I am using [EMAIL PROTECTED] V 2.6.I used to sniff the server for 24 hours ,the result was that my server is talking to another servers through port 80 and 22 and 1000, tcp Ips are 4.254.167.67 65.119.192.235 83.133.127.60 Is there any backdoor or a Trojan ,and how can I fix or debug it to be able to stop it, Remark: My hard drive and my network is completely cleaned from viruses,I used to burn a new image and I get these results . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: what is the real use of AEL?
On Thu, 17 Aug 2006 04:02:15 +0400 , Jean-Michel Hiver wrote:
On Wed, 2006-08-16 at 18:30 -0700, asterisk-users-
[EMAIL PROTECTED] wrote:
Steve Murphy joined our development team at the beginning of June. Steve
(murf on IRC/Mantis) had rewritten Asterisk's expression parser and the
AEL language parser as a volunteer community member, along with various
other bug fixes and improvements.
Which makes me think, what is the real use of AEL. I have taken a look
at it, and it makes asterisk's config file almost as unreadable as SER.
What exactly does AEL do that a well written AGI / FastAGI app doesn't?
I would think (but I'm surely wrong) that it would be better to do work
on having well defined APIs that allow us to script Asterisk (such as
AGI and the Manager interface) rather than invent Yet Another Pseudo
Programming Language - which is going to be an endless task... Don't you
think?
That being said, just like the rest of the community, I'm very happy
with Kevin's exciting announcement!
Cheers,
Jean-Michel.
In the above, Jean-Michel puts it right on the table: of what
possible use is AEL? Why am I bothering to waste my time with it? It's a
valid question! It deserves some discussion!
(For those of you who don't have any idea about what this discussion is
about, find out all about AEL via the web page,
http://www.voip-info.org/wiki/view/Asterisk+AEL2 )
Many thanks to rushowr for his reply in my defense as pertaining to
speed and readability. Actually, it might be interesting to get a
handle on the real speed of the AEL extension engine vs. the AGI program
at the other end of the pipe. Here is a little AEL script:
context sppedtest {
771 = { // Get the PIPS (priority instructions per second)
// for your current asterisk box.
// perform this on quiescent asterisk box --
// all your CPU cycles are belong to us
iterations=10; // Choose iterations wisely--
// around 10 sec. of execution time
// is good, gives good granularity.
Set(time1=${EPOCH});
for(i=1; ${i} ${iterations}; i = ${i} + 1)
NoOp(Hello);
Set(time2=${EPOCH});
Verbose(The time diff is $[ ${time2} - ${time1} ] seconds);
Verbose(Which means that the priorities/sec = $[ 4*${iterations} /
(${time2} - ${time1}) ]);
SayNumber($[ 4 * ${iterations} / (${time2} - ${time1}) ]);
}
}
The above compiles into this extensions format:
exten = 771,1,Set(iterations=$[10])
exten = 771,2,Set(time1=${EPOCH})
exten = 771,3,Set(i=$[1])
exten = 771,4,GotoIf($[${i} ${iterations}]?5:8)
exten = 771,5,NoOp(Hello)
exten = 771,6,Set(i=$[${i} + 1])
exten = 771,7,Goto(4)
exten = 771,8,NoOp(Finish for-workext-48)
exten = 771,9,Set(time2=${EPOCH})
exten = 771,10,Verbose(The time diff is $[${time2} - ${time1} ]
seconds)
exten = 771,11,Verbose(Which means that the priorities/sec = $[4*
${iterations} / (${time2} - ${time1}) ])
exten = 771,12,SayNumber($[4 * ${iterations} / (${time2} - ${time1}) ])
(Please remember that email packages break lines where they shouldn't be
broken for extensions.conf format to work!)
Stick the 771 extension into a context where you can dial it. Play with
the iterations number until it runs roughly 10 seconds or so. Do it on
a quiet, unloaded asterisk box. The goal is to get the biggest number
you can in the 10 or so seconds. Set your Verbose level to 0. You don't
want a hundred-thousand log entries.
Now, rewrite the above into an AGI script. You'll definitely be able
to run the loop math faster, I'd bet. But... well, just see. My little
1.8Ghz machine runs at 83k PIPS. A 64-bit dual processor monster ran at
roughly 200K PIPS. I'd be very interested to see what kind of
performance an (exactly) equivalent perl AGI, or PHP AGI, or even a
straight compiled C program would get in this case. They should all be
much, much, much faster, right? Orders of magnitude faster? My guess, is
maybe 3 or 4 times faster at the most, but... well, try it and see. Just
make sure that the NoOp(Hello) gets executed in asterisk, OK?
Sorry for the diversion. My answer to Jean-Michel's straightforward
question goes along some different lines than rushowr. I never really
cared how fast/efficient the extension engine was-- it's obvious I'm
not writing stuff for thousands of concurrent users like rushowr. But in
the majority of cases, it's the apps that are run from AEL that take up
all the execution time. As long as AEL execution time is pretty minimal
between priorities, it's probably going to be OK. (users of dialplans
for intensively loaded sites may HIGHLY disagree!)
My first reason for getting excited about AEL, enough so, to rewrite the
parser to make it more user-friendly, and add a few bells and whistles,
was that it provided an opportunity to code dialplans with higher
level constructs than gotos. Truly, AEL
RE: [asterisk-users] Security
I make an namp search and the result was its was talking to irc server ,is it normal [EMAIL PROTECTED] ~]# nmap 65.119.192.235 Starting nmap 3.70 ( http://www.insecure.org/nmap/ ) at 2006-08-17 02:34 E Interesting ports on ftp2.c3icare.com (65.119.192.235): (The 1650 ports scanned but not shown below are in state: closed) PORT STATE SERVICE 21/tcp open ftp 135/tcp filtered msrpc 139/tcp filtered netbios-ssn 445/tcp filtered microsoft-ds 990/tcp open ftps 1000/tcp open cadlock 1025/tcp open NFS-or-IIS 3389/tcp open ms-term-serv 4899/tcp open radmin 1/tcp open snet-sensor-mgmt Nmap run completed -- 1 IP address (1 host up) scanned in 352.570 seconds [EMAIL PROTECTED] ~]# nmap 83.133.127.60 Starting nmap 3.70 ( http://www.insecure.org/nmap/ ) at 2006-08-17 02:42 E Interesting ports on webdiscounter.ch (83.133.127.60): (The 1651 ports scanned but not shown below are in state: closed) PORT STATE SERVICE 22/tcp open ssh 135/tcp filtered msrpc 139/tcp filtered netbios-ssn 445/tcp filtered microsoft-ds 1422/tcp open autodesk-lm /tcp open irc-serv 6667/tcp open irc 6668/tcp open irc 7000/tcp open afs3-fileserver Nmap run completed -- 1 IP address (1 host up) scanned in 496.802 seconds From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Thursday, August 17, 2006 9:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: [asterisk-users] Security Dear I am using [EMAIL PROTECTED] V 2.6.I used to sniff the server for 24 hours ,the result was that my server is talking to another servers through port 80 and 22 and 1000, tcp Ips are 4.254.167.67 65.119.192.235 83.133.127.60 Is there any backdoor or a Trojan ,and how can I fix or debug it to be able to stop it, Remark: My hard drive and my network is completely cleaned from viruses,I used to burn a new image and I get these results . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SIP 183 Session Progressing
16 aug 2006 kl. 07.26 skrev Dinesh Nair: On 08/15/06 23:30 Michael J. Tubby B.Sc (Hons) G8TIC said the following: I suspect your problem is with the softphone implementation... definitely, the SIP spec iianm says that UACs should play a ringing tone when the 180 is received. Occasionally calls which go from 100 - 180 without going via the 183 result in the Cisco ringing and combined rining genrated by the telephone exchange which is weird but ok. the supplementary question then is, since i can't change the softphone would i break anything if i forced the sending of the 183 packet anyways from within chan_sip ? Don't do it within chan_sip, do it within the dialplan by using playback with the no answer option before you dial out... You can check the user agent with a dialplan function. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXMODEM
maybe you need to clean the iaxmodem sources including the lib to be able to recompilemake cleanthen try agin ./build staticOn 8/17/06, Giedrius Augys [EMAIL PROTECTED] wrote:Hi , I removed from /usr/local all spandsp and iax libraries and ./build static . But I get this: creating libspandsp.la(cd .libs rm -f libspandsp.la ln -s ../libspandsp.la libspandsp.la)make[2]: Leaving directory `/home/eryx/hylafax/iaxmodem- 0.1.14/lib/spandsp/src'make[1]: Leaving directory `/home/eryx/hylafax/iaxmodem- 0.1.14/lib/spandsp/src'make[1]: Entering directory `/home/eryx/hylafax/iaxmodem-0.1.14/lib/spandsp'make[1]: Nothing to be done for `all-am'.make[1]: Leaving directory `/home/eryx/hylafax/iaxmodem-0.1.14/lib/spandsp' And that's all , it didn't compile iaxmodem. Maybe something is missing 2006/8/17, Lee Howard [EMAIL PROTECTED]: Giedrius Augys wrote: And when I make ./build , I get this: /tmp/cckNRgf3.o(.text+0xfc): In function `cleanup': /home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:165: undefined reference to `iax_destroy'You probably have a conflicting version of libiax2.try doing: ./build static instead. Lee.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help: RTP Stream not set up correct only when calling out
Dear astreisk-users mailing list subscribers, with my asterisk under debian (Version: 1:1.0.9.dfsg-5) i have the problem, that the RTP stream is not set up correct with an outgoing call. Incoming calls are working with no problems. The problem is, that the RTP stream is initiated from IP A and my asterisk or my router/with firewall (ISA 2004) replies to the port the RTP stream comes from, but to the ip-adress it is talking SIP, not the IP A, from where the RTP stream is set up. For a better understaning please see http://rapidshare.de/files/29699435/graph_outgoing-call.JPG; or http://rapidshare.de/files/29699317/graph_outgoing-call.txt;. More infos can be found in the trace i took at my router/firewall at the external interface (myIP): http://rapidshare.de/files/29699460/outgoing-call.trace.sip.txt;. Please have a look at my problem and give me a feedback. Thanks in advance, ontae - End forwarded message - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone for Windows Mobile 5?
Christian wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. We are using PPCIax -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
On Wed, Aug 16, 2006 at 03:43:26PM -0600, Douglas Garstang wrote:
[ Unnecessary flames snipped ]
I did mention the code. I mentioned the two python API's, and the perl API.
http://sourceforge.net/projects/pyst
http://py-asterisk.berlios.de/py-asterisk.php
http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager/Manager.pm
I haven't tried the python interface. I have experimented a bit with the
perl Asterisk POE component.
perldoc POE::Component::Client::Asterisk::Manager
gives you a nice little program. However you must realise that it is
part of a bigger framework (POE). Try
perldoc POE
for starters.
Also note that those modules don't attempt to give you a full
documentation of the Asterisk manager interface, as the manager
interface's specific functions depend on the availbility of specific
modules in the Asterisk installation and of the version of Asterisk. The
manager interface has basic documentation in Asterisk ('show manager
command foo' or 'show manager commands').
IIRC, however, something I found lacking at that wrapper to the manager
interface is that its main loop does not give a separate ID to each
command I send. And thus it can be confused when you get a fast flow of
events.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
___
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Re: [asterisk-users] Force immediate re-registration on sip reload
16 aug 2006 kl. 22.02 skrev kjcsb: Is there any way to force Asterisk to re-register after a sip reload is issued? At the moment, after a sip reload is issued, sip show registry reports all sip UA entries as Unregistered. How can I get Asterisk to immediately send out a registration request to the proxy? After a reload, my Asterisk re-registers to everything in the registry. Check what's going on by starting SIP debug and see if there's any REGISTER messages going out. Similarly all SIP peers lose their registration status with Asterisk. So when the device is used to make a call immediately after the SIP reload the call is not processed by Asterisk. It takes about 2 minutes 20 seconds before Asterisk starts processing SIP register requests from UAs and before it sends out the registration request to the proxy. How can I reduce this time? Asterisk saves SIP peer registrations in the AST db (run database show) so that we can restore the status after a reload. This does not apply to subscriptions. If it takes 2 minutes and 20 seconds to reload fully - do you have a very large installation? Check before reload if there's anything in the AST db. Also, please always indicate your version of Asterisk when you ask questions. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk Bootcamp Boston next week - open seats available! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
16 aug 2006 kl. 23.01 skrev Michael Collins: Doug, I'm sure that you are not the only one who considers an API w/o docs to be of limited or no value. I just doubt that many people have use for a management API because they don't use the Asterisk manager interface very much. Manager actions are well documented in the CLI from Asterisk version 1.2. There's also a readme file. The only part of manager that is poorly documented today is all the manager events - any help to document those are appreciated. If you don't think the manager action documentation is correct or enough, I look forward to your input and patches/suggestions. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk Beachcamp - Malaga, Spain! September 25-29 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP-NAT failure on dynamic IP
16 aug 2006 kl. 23.54 skrev Technical Support: We have a client running Asterisk using a dynamic IP. When the IP lease is renewed to a different address, their SIP connections to external clients fail (one way audio). A simple asterisk restart fixes the problem, but they can't count on an admin always available. Is there way in sip.conf or other to eliminate the dependence on a particular IP? (for clients behind NAT)? Look at the externhost= setting in sip.conf. That's where you set a dyndns hostname that will be refreshed regurlarly by Asterisk as you change the IP. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * [EMAIL PROTECTED] VON Fall, Boston, Sept 11-14 - http://www.pulver.com/ asterisk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5
Below is a good example of misusing init.d scripts with the obvious results of the need for ugly workarounds. Why do you fight against the design of your system? See specific comments below, On Wed, Aug 16, 2006 at 08:20:55AM -0400, Steven wrote: rc.local: touch /var/lock/subsys/local setpci -v -s 00:1f.1 LATENCY_TIMER=4 setpci -v -s 02:0e.0 LATENCY_TIMER=4 setpci -v -s 0b:07.0 LATENCY_TIMER=4 setpci -v -s 0c:08.0 LATENCY_TIMER=4 setpci -v -s 10:0d.0 LATENCY_TIMER=0 setpci -v -s 06:02.0 LATENCY_TIMER=ff sleep 5 Why a sleep here? What exactly do you wait for? sleep is an indication of a bad workaround for a race condition. It is a bad workaround, as you can never know if you sleep enough. echo UnLoading wct4xxp rmmod -v wct4xxp rmmod -v zaptel sleep 3 Huh? Why is that? BTW: 'genzaptelconf -u' will unload all zaptel modules. If 'modprobe -r' works on your system: that's even better. echo Loading wct4xxp /sbin/modprobe -v zaptel sleep 5 /sbin/modprobe -v wct4xxp sleep 5 # ztcfg - #sleep 5 The 'sleep'-s here are because of the following bugs: 1. running ztcfg automatically on modules load 2. not properly waiting for /dev/zap/ctl to be generated by udev echo 1 /proc/irq/201/smp_affinity echo 1 /proc/irq/217/smp_affinity echo 0 /proc/irq/209/smp_affinity echo 1 /proc/irq/14/smp_affinity /usr/sbin/amportal start Now, that's an init.d script. Most of the code above should be merged into the zaptel init.d script (which is run before asterisk/amportal). Then the init system will run amportal later. If actually needed. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security
Khaled Chehab wrote: Dear I am using [EMAIL PROTECTED] V 2.6.I used to sniff the server for 24 hours ,the result was that my server is talking to another servers through port 80 and 22 and 1000,… tcp This isn't the [EMAIL PROTECTED] mailing list - Take your issues to the appropriate forum. Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security
On Thu, Aug 17, 2006 at 09:41:02AM +0300, Khaled Chehab wrote: Dear I am using [EMAIL PROTECTED] V 2.6.I used to sniff the server for 24 hours ,the result was that my server is talking to another servers through port 80 and 22 and 1000,. tcp netstat -lntp , for starters. Ips are 4.254.167.67 65.119.192.235 83.133.127.60 Is there any backdoor or a Trojan ,and how can I fix or debug it to be able to stop it, Remark: My hard drive and my network is completely cleaned from viruses,I used to burn a new image and I get these results . The above two remarks indicate a very simplistic approach to the issue of security. Please do some basic reading before providing services from your system to the internet. You don't add a security component to your system. Trojans and viruses don't just land in a system. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Festival through AGI can't handle strings longer than 15 chars
I'm having a tough problem when using Festival with Asterisk through AGI: it seems that when I pass more than 15 chars to the Festival command, when from inside an AGI, no sounds (speech) at all is generated. The following (from inside the dialplan) correctly works: exten = 333,1,Answer() exten = 333,2,FESTIVAL(Telefono spento uno) exten = 333,3,Hangup But, when moved from within an AGI, the same Festival command doesn't work: EXEC FESTIVAL Telefono spento uno the symptom is that no text is played, although the return code from command is zero. One important note: if I shorten the text to Telefono spento (i.e. at most 15-chars wide) everything works as expected. I really can't figure out the reason of this weird behavior. What I can do is to exclude some possible reasons: 1. It is not a festival-related problem since when called from the Dialplan everything works as expected. 2. It is not a language-related issue, since I tried this both with English and Italian 3. It is not a missing call to flush()... yes, I added a flush() at the end of my Python-based AGI call 4. It is not a problem related to Python, since I use Python extensively with AGI Does anyone have a hint on what I can do to investigate or solve this problem? Does enyone know if this is a known bug? Thanks in advance, Mario ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF issues on voicemail on Zap
On Thu, Aug 17, 2006 at 01:19:28AM -0400, Justin Tunney wrote: First, you want to set verbose to AT LEAST 666. hmmm... Any reference in the code to something that requires that specific debug level? I've seen too many such magic numbers thrown here. You may also want to turn on debug messages in logger.conf This is actually the more important factor. Chan_zap lists the dtmf digits it detects and/or sends in debug messages. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time
I guess my problem might be that, because I pretend Asterisk to use my sip.conf static configuration file and also MySQL tables referenced in extconfig.conf like this: [settings] sipusers = mysql,asterisk,sip sippeers = mysql,asterisk,sip voicemail = mysql,asterisk,voicemail While I'm using one thing I can't use the other right??? Based on my limited knowledge you *can* use both at the same time. Any Sip details created in the sip table in your asterisk database will be available immediately to Asterisk. They will not be reported in the command line if you enter sip show peers. This is sometimes called realtime dynamic. Any Sip entries in sip.conf will *also* be available to Asterisk but only on reload e.g. sip reload. These will be reported in the command line if you enter sip show peers. However to complicate matters further there are two additional things to be aware of: realtime caching realtime static Realtime caching loads the sip details from the database in a similar way to how the details from sip.conf are loaded i.e. both the details from sip.conf and from the database will be reported if you enter sip show peers. However changes made in the database are not immediately available - you need to reload just like if you made a change in sip.conf. To enabled this you must set rtcachefriends=yes in sip.conf Realtime static is totally different to the realtime discussed above. It uses a different database structure and is intended to replace the Asterisk static files. Beyond that I'm unsure. Personally I think realtime is a very misleading name. Extconfig would be a better term. Extconfig allows Asterisk to read its configuration files from any external source. Asterisk can be configured to source certain configuration files (e.g. sip.conf) internally (the default which will read from a text file) or (these are mutually exclusive) from an external source such as a database (so-called realtime static). *In addition*, Extconfig can read configuration information from an external source on-the-fly (realtime dynamic) or cached (realtime cached). If anything I've said above is incorrect I'd sure appreciate an expert correcting me. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
2000+ lines:D Film Script? Madhawa Jeremy McNamara wrote: Douglas Garstang wrote: We have a 2000+ line python script that handles all call routing logic. You expect that to scale? I do call routing in 3 contexts with ~maybe~ a dozen extension each - and we have many thousands of customers and more than hundreds of companies using our Asterisk systems as a hosted solution. I really think you need to totally re-think your operation - and no, I'm not going to explain it to you, so don't even ask. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXMODEM
I did as you say and the result is the same. Maybe it is need to create a link in /usr/src/linux into kernel2006/8/17, From PH [EMAIL PROTECTED] :maybe you need to clean the iaxmodem sources including the lib to be able to recompile make cleanthen try agin ./build staticOn 8/17/06, Giedrius Augys [EMAIL PROTECTED] wrote: Hi , I removed from /usr/local all spandsp and iax libraries and ./build static . But I get this: creating libspandsp.la(cd .libs rm -f libspandsp.la ln -s ../libspandsp.la libspandsp.la)make[2]: Leaving directory `/home/eryx/hylafax/iaxmodem- 0.1.14/lib/spandsp/src'make[1]: Leaving directory `/home/eryx/hylafax/iaxmodem- 0.1.14/lib/spandsp/src'make[1]: Entering directory `/home/eryx/hylafax/iaxmodem-0.1.14/lib/spandsp'make[1]: Nothing to be done for `all-am'.make[1]: Leaving directory `/home/eryx/hylafax/iaxmodem-0.1.14/lib/spandsp' And that's all , it didn't compile iaxmodem. Maybe something is missing 2006/8/17, Lee Howard [EMAIL PROTECTED]: Giedrius Augys wrote: And when I make ./build , I get this: /tmp/cckNRgf3.o(.text+0xfc): In function `cleanup': /home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:165: undefined reference to `iax_destroy'You probably have a conflicting version of libiax2.try doing: ./build static instead. Lee.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Pagarbiai,Giedrius AugysSiauliu Universitetas, ISTIP telefonijos inzinieriusTel. 8 41 590408 Mob. Tel. 8 678 05790el. pastas [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wanpipe1:w1g1: Rx Error: 'Retry' exceeds maximum (64k): pci fatal error! (0x0007C03C)
Hi, I'm trying to install Asterisk 1.2.9.1 on a Debian Sarge distro with a Sangoma A102 using zaptel 1.2.6 and libpri 1.2.3. Asterisk seems to work correctly but when I reboot the server I get the following error inside /var/log/messages: localhost kernel: wanpipe1:w1g1: Rx Error: 'Retry' exceeds maximum (64k): pci fatal error! (0x0007C03C) What does it mean exactly? Notice that I haven't connected any PRI line to the card. Is there a way to know if the card is working correctly? TIA Giorgio Incantalupo My *zaptel.conf* is: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,1,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 and my *zapata.conf *is: [channels] signalling=pri_cpe context=blah channel = 1-15 signalling=pri_cpe context=blah channel = 17-31 signalling=pri_cpe context=blah channel = 32-46 signalling=pri_cpe context=blah channel = 48-62 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExtensionState always returns 1
Hello, I have one SIP extension that can make calls, but not receive. If I use ExtensionState in the manager, I always see 1 (In Use), with the softphone not in use. Is there a way to change the extension status? Thanks. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR inaccuracies
Hi list, i am currently having problems with CDR accuracy on my asterisk PBX MY SETUP = I have two Asterisk systems, the first one (tagged primary pbx) has E1 lines connected to it and this processes calls on behalf of the secondary pbx. Now the CDR on the primary PBX are very accurate but i cannot use this since it collects calls for IAX trunks and there is no way of identifying who made what calls. The primary PBX only provides E1 connectivity to the secondary PBX. Users do not connect to it. The secondary pbx connects all the usersand it handls outbount call authentications. it also saves a separate cdr for all calls and flags each call with the call code of the caller. This data is however inaccurate because when a call is trunked through the promary pbx, it is always awnsered by the promary, thus the duration and billsec are always the same. the dispositon colums cannot be relied upon also because the right reasons are not always sent back. For instance, if i decided to drop a ringing call, the primary will record 'no awnser' in my disposition but the secondary will record 'awnsered' can anyone help me out? i know that it has to do with call presentations, etc. but it seems that SIP does not handle 'call in progress' signals correctly ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
On Thu, 2006-08-17 at 14:22 +0600, Dualcall.com wrote: 2000+ lines:D Film Script? No, SCO lines. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conflict between S L option in Dial?
Hello, I have a question about whether the L option in the dial command conflicts with the S option? For example, I have the following in my Dial command: SIP/[EMAIL PROTECTED]|60|HL(:3:1)S(120) I see in my CDRs that there are calls lating more than 120 seconds. By reading the description of the dial application, I assumed that since I want the user to talk for 120 seconds I should use: S(120) instead of L(12:3:1) Does anyone know if using L(:3:1) and S(120) conflict each other? Thank you, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing out using SIP terminal
Hi all,
I have a VoIP GSM (SIP) terminal that I have
successfully configured and registered in asterisk,
would like to:
a)Answer calls via asterisk coming from this terminal.
b)Route outbound calls to this terminal.
What Dial command do I use so as to have the sip
terminal dial an outside line, for example when using
Zap, I use the following commands successfully.
extension.conf
[from-sip]
exten=20,1,Dial(Zap/4/0w10136)
exten=_22.,1,Dial(Zap/4/0w${EXTEN:2},5,r)
If I use the command below in an attempt to dial a
number 0729932165 on the SIP/2006 channel, the call is
answered by the terminal but doesn't dial out.
exten=21,1,Dial(SIP/2006/0729932165,5,r)
My current configuration looks like this
sip.conf
[2006]
type=friend
username=2006
secret=2006
context=voip_gsm; Where to start in the dialplan
when this phone calls
callerid=Allan 2006 ; Full caller ID, to override
the phones config
host=dynamic
defaultip=192.168.0.100
extension.conf
[voip_gsm]
exten=s,1,NoOp(${EXTEN})
exten=s,2,NoOp(${CALLERID})
exten=s,3,Dial(SIP/2001,5,Ttm) ;dial a sip hardphone
configured in my Asterisk installation.
exten=s,4,Voicemail([EMAIL PROTECTED])
exten=s,5,Hangup
exten=s,104,Voicemail([EMAIL PROTECTED])
exten=s,105,Hangup
[from-sip]
exten=21,1,Dial(SIP/2006/0729932165,5,r)
Allan.
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Re: [asterisk-users] Festival through AGI can't handle strings longer than 15 chars
Hi Mario. Have you tried to enable AGI debug? CLI agi debug That will show what Asterisk is receiving from your script. Also enable all the debug messages in the logger.conf file for the console Go and try that and post what you see here, and we may be able to help you On 8/17/06, Mario [EMAIL PROTECTED] wrote: I'm having a tough problem when using Festival with Asterisk through AGI: it seems that when I pass more than 15 chars to the Festival command, when from inside an AGI, no sounds (speech) at all is generated. The following (from inside the dialplan) correctly works: exten = 333,1,Answer() exten = 333,2,FESTIVAL(Telefono spento uno) exten = 333,3,Hangup But, when moved from within an AGI, the same Festival command doesn't work: EXEC FESTIVAL Telefono spento uno the symptom is that no text is played, although the return code from command is zero. One important note: if I shorten the text to Telefono spento (i.e. at most 15-chars wide) everything works as expected. I really can't figure out the reason of this weird behavior. What I can do is to exclude some possible reasons: 1. It is not a festival-related problem since when called from the Dialplan everything works as expected. 2. It is not a language-related issue, since I tried this both with English and Italian 3. It is not a missing call to flush()... yes, I added a flush() at the end of my Python-based AGI call 4. It is not a problem related to Python, since I use Python extensively with AGI Does anyone have a hint on what I can do to investigate or solve this problem? Does enyone know if this is a known bug? Thanks in advance, Mario ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SendDTMF - how to use
I would like to send dtmf digits to automaticaly processing ivr (i.e.
without user intervention from phone),
here some really simple example, what I would like to do:
- when enter context test automaticaly dial extension 8297 (using DTMF
code), but this doesn't working, or this is completely wrong use this app?
thank you!
PJ
[test]
exten = s,1,Playback(beep)
exten = s,n,SendDTMF(8297)
exten = _ZXXX,1,Dial(IAX2/astpeer/${EXTEN})
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Re: [asterisk-users] Asterisk 'Hosting'
Jeremy McNamara wrote: as most people here know, I yell at stupid people. Be honest, Jeremy, you yell at everyone! -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial out based on SIP invite
Assume that I receive an Invite from a SIP device that Asterisk has
registered with. How do I get Asterisk to dial out using the Invite details
as if the Invite had been received from a UA registered with Asterisk? i.e.
UA - SIP Proxy - Asterisk - PSTN gateway.
For example
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:147.202.nn.nnn;ftag=bf7eced18eb7271b;lr=on
Via: SIP/2.0/UDP 147.202.nn.nnn;branch=z9hG4bKe49c.21b320a3.0
Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK76bf3dec8d45b972
From: User sip:[EMAIL PROTECTED];tag=bf7eced18eb7271b
To: sip:[EMAIL PROTECTED]
etc
If the Invite was received from a SIP device registered with Asterisk (in
the [from-internal] context) then the call would be routed to
[outrt-003-test] and dial out correctly.
I want to do the same thing with the Invite received from the SIP proxy. Can
anyone advise how I can achieve this (in Asterisk 1.2.9)? Cut-down versions
of conf files are below.
sip.conf
register=1122334455:[EMAIL PROTECTED]/66554433
[1122334455]
type=peer
host=proxy.domain.com
fromuser=1122334455
context=from-internal
extensions.conf
[from-internal]
include = from-internal-additional
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
exten = 66554433, 1, ?
[from-internal-additional]
include = outbound-allroutes
[outbound-allroutes]
include = outrt-003-test
exten = foo,1,Noop(bar)
[outrt-003-test]
exten = _90[2-79]XX.,1,Macro(dialout-trunk,1,${EXTEN:1},,)
exten = _90[2-79]XX.,n,Macro(dialout-trunk,5,${EXTEN:1},,)
exten = _90[2-79]XX.,n,Macro(dialout-trunk,3,${EXTEN:1},,)
exten = _90[2-79]XX.,n,Macro(dialout-trunk,2,${EXTEN:1},,)
exten = _90[2-79]XX.,n,Macro(outisbusy,)
[macro-dialout-trunk]
exten = s,1,GotoIf($[${ARG3} = ]?3:2) ; arg3 is pattern password
exten = s,2,Authenticate(${ARG3})
exten = s,3,Macro(user-callerid)
exten = s,4,Macro(record-enable,${CALLERID(number)},OUT)
exten = s,5,Macro(outbound-callerid,${ARG1})
exten = s,6,Set(GROUP()=OUT_${ARG1})
exten = s,7,GotoIf($[ ${GROUP_COUNT()} ${OUTMAXCHANS_${ARG1}} ]?108)
; if we've used up the max channels, continue at (n+101)
exten = s,8,Set(DIAL_NUMBER=${ARG2})
exten = s,9,Set(DIAL_TRUNK=${ARG1})
exten = s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial
string for this trunk
exten = s,11,Set(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ; OUTNUM is
the final dial number
exten = s,12,Set(custom=${CUT(OUT_${ARG1},:,1)}) ; Custom trunks are
prefixed with AMP:
exten = s,13,GotoIf($[${custom} = AMP]?16)
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},120,${TRUNK_OPTIONS}) ; Regular
Trunk Dial
exten = s,15,Goto(s-${DIALSTATUS},1)
; This is a custom trunk. Substitute $OUTNUM$ with the actual number and
rebuild the dialstring
; example trunks: AMP:CAPI/:b$OUTNUM$,30,r,
AMP:OH323/[EMAIL PROTECTED]:
exten = s,16,Set(pre_num=${CUT(OUT_${ARG1},$,1)})
exten = s,17,Set(the_num=${CUT(OUT_${ARG1},$,2)}) ; this is where we
expect to find string OUTNUM
exten = s,18,Set(post_num=${CUT(OUT_${ARG1},$,3)})
exten = s,19,GotoIf($[${the_num} = OUTNUM]?20:21) ; if we didn't find
OUTNUM, then skip to Dial
exten = s,20,Set(the_num=${OUTNUM}) ; replace OUTNUM with the actual
number to dial
exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS})
exten = s,22,Goto(s-${DIALSTATUS},1)
exten = s,108,Noop(max channels used up)
exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten = s-BUSY,2,Busy()
exten = s-BUSY,3,Wait(60)
exten = s-BUSY,4,NoOp()
exten = _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})
Please note that Asterisk also receives Invites from the same proxy (same IP
and port) that need to be treated differently i.e. as if they were external
incoming calls. If this were not the case then the following sip.conf
achieves the desired result (I've tested this successfully). The call gets
into the from-internal context and the outbound call to the PSTN is made:
sip.conf
register=1122334455:[EMAIL PROTECTED]
[1122334455]
type=peer
context=from-internal
However when I create another SIP peer, even though the Invite from the
Proxy has different From details, and I specify fromuser and host in
sip.conf under [1122334455], the call is treated as an external call.
Any advice appreciated.
Cameron
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[asterisk-users] Trying to build a hybrid Asterisk system w/ one POTS and 2 Speakeasy VOIPs - anyone done something similar w/ Speakeasy?
Hola! The Speakeasy guy says they don't support Astericks but as that's what he always says about everything I though I'd try it anyway. Has anyone done this? Iy uses a Motorala VT1005. If this is totally impossible I guess I'll go to Vonage, unless someone has a better notion. I currently have a 1.5/768 from Speakeasy and am happy with it. Thanks for any help, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip suppression
Hi, When I make a call using asterisk (SIP - provider - PSTN), it is cutting out all the background noise, making it a bit like a walkie talkie when they stop talking. Is this something asterisk is doing?!? I have a 2MB leased line just for calls, so bandwidth is fine, I have tried with G711a/u and GSM codec but they all do the same. I'm using a Polycom IP501 to connect to asterisk, but have the VAD set to 0 in the config file to disable Voice Activity Detection on the phones. Could it be my provider trying to do this? As I have noticed on the CLI 'NOTICE[2530]: rtp.c:593 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible.' Thanks, Dean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Hi,Here I am posting my problem. I am getting this problem since 8 days. I have studied documentation and looked previous posts in forums. But, I am unable to solve this problem. Please show me a solution. I am from India. We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN-Eg. Mobile), I am not getting the callerid number of the caller and getting callerid as "Asterisk" in my softphones (XLite).Here I am giving my config files and error message. Please see it.zaptel.conf contents:loadzone = usdefaultzone=usfxsks=1-4zapata.conf contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nomusiconhold=defaultringtimeout=8000cidsignalling=dtmfcidstart=ringgroup=1callgroup=1pickupgroup=1channel = 1sip.conf contents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.conf contents:[incoming]exten = s,1,Wait(4)exten = s,n,Answerexten =
s,n,SetMusicOnHold(default)exten = s,n,Set(TIMEOUT(digit)=5)exten = s,n,Set(TIMEOUT(response)=10)exten = s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten = s,n,Hangup()include = leader[leader]exten = 105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten = 105,3,Voicemail(b105)exten = 105,4,Hangupexten = _9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten = _5,1,Dial(Zap/1/${EXTEN:1}) ; Local Landlineinclude = internal[internal]exten = 105, 1, Dial(SIP/105,15)When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console:Error Message:Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8)Aug 17 19:45:41 WARNING[10449]:
chan_zap.c:6087 ss_thread: CallerID feed failed: SuccessAug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1'Please tell me the solution. Looking forward to your kind response. Thank you.Regards,Chandra.
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[asterisk-users] IAX2 Turnks
I have two asterisk (trixbox) connected by IAX2 Trunk. Both of them have interfaces TE205P configured and working fine. I can places calls to PSTN on both of them. I can place calls from SIP phones connected on asterisk one, using the IAX2 Trunk, to SIP phones connected on the asterisk two. I can not place calls from SIP phones connect on asterisk one to ZAP Trunk connected on the other. Is it possible?? Please help me, I am getting crazy. Hernany Oliveira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] Recent additions to the Digium Asterisk development team
Very good news. Really good to know about the success of companies(like Digium) and developers(like all mentioned by Kevin) that are working with and for the Asterisk community. I just have one thing to complain: When will Digium invite a developer to put the MFCR2 stack(channel) on Asterisk official core? Keep in mind that South American/Asian markets is growing UP pretty faster on VoIP, and of course Asterisk is one of the tools that have been used to get this grow. MFCR2 is almost on 90% of all telephony carriers in Brazil. I'm the founder of AsteriskBrasil.org(born on 2004), we have 5000 users and 2000 members on the email discussion list/IRC. All of them are using MFCR2, implemented by Steve Underwood that deserves all of AsteriskBrasil.org community's respect. The VERY GOOD work done by Steve on the chan_unicall, spandsp and libmfcr2 turn on the possibility to work with Asterisk in Brazil, but is a pain to apply a patch every time a new Asterisk version is announced, is pain to maintain two software trees. AsteriskBrasil.org has its own developers that is doing a very good work on translating, coding and recoding things to work in Brazil (some of limfr2 stuff, voicemail, grammar, etc -I'll prepare a full list-) that should help the Asterisk dev team to put some of our needs on the core. I'll not write more lines here, I just wanna know: Is Digium interested to keep/grow business in South America/Asia? Thanks for all of you specially for Steve(coppice). Denis Galvão AsteriskBrasil.org On 16 de ago de 2006, at 19:12, Kevin P. Fleming wrote: Some of you may have noticed some new people with '@digium.com' email addresses lately... yes, we have been hiring to expand our Asterisk development team and I should have made an official announcement some time ago :-) Joshua Colp joined our development team a few months ago. Josh (file on IRC/Mantis) has been working on Asterisk development for quite some time and had contributed many features and bug fixes as a volunteer community member, along with being very active on the IRC channels and issue tracker. Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements. Jason Parker joined our development team at the beginning of this week. Jason (qwell on IRC/Mantis) has been maintaining the chan_skinny driver for Cisco SCCP phones as well acting as a bug marshal and fixing various bugs in Asterisk for the past year or more. Russell Bryant has been a Digium part-time employee and an active Asterisk maintainer since before I got involved with Asterisk :-) His contributions are innumerable, and he has worked far more than the 'ten to twenty hours per week' he claims to have available outside of his school work! Russell (russellb on IRC/Mantis) will be joining us full time in Huntsville after the winter semester is complete, when he expects to graduate. Please join me in welcoming all these new members of our development team; they are helping to make Asterisk (and our other software products) better every day and will enable us to accelerate our products into the future. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SIP 183 Session Progressing
On 08/17/06 14:56 Olle E Johansson said the following: Don't do it within chan_sip, do it within the dialplan by using playback with the no answer option before you dial out... yes, that will force early media and cause sip_write() to force send a 183. thanx, this should work. i'll test it out and report back. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI transfer question
I'm looking for some guidance on setting up an extension (associated with a physical port) properly and being able to transfer calls in an AGI script. I started out my home brew system with just an fxo port attached to the PSTN. Now, I've added a fxs port and I'd like to hang a phone off of it (TDM411). My current extensions.conf looks like this: exten = s,1,Wait,1 exten = s,2,Answer (and so on) and it works just fine picking up an incoming call and launching into my AGI script. But I'd like to expand the system to be able to transfer calls between the fxo and fxs ports in an AGI script. Basically, a very simply PBX. So, an incoming call on the fxo could be sent to the fxs. And an outbound call when the phone on the fxs port is picked up, would be able to dial 9 to get out. Thanks in advance, Roy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] valgrind + Asterisk
Hi, has anybody got valgring to work with asterisk i do a -- valgrind --tool=memcheck -v asterisk -c then Asterisk just dies. The problem I have is that on the box I have Asterisk running, the memory is reported as being used up, then when there is liitle ram left, the box just hangs. So Asterisk might have a memory leak, and I am trying to find it. Can anybody help? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] valgrind + Asterisk
On Thu, Aug 17, 2006 at 02:37:52PM +0200, yusuf wrote: Hi, has anybody got valgring to work with asterisk Yes i do a -- valgrind --tool=memcheck -v asterisk -c then Asterisk just dies. What version of asterisk? Did you use any special build options to build it? The problem I have is that on the box I have Asterisk running, the memory is reported as being used up, then when there is liitle ram left, the box just hangs. What do you mean? [EMAIL PROTECTED]:~$ free total used free sharedbuffers cached Mem:237124 231096 6028 0 35700 36992 -/+ buffers/cache: 158404 78720 Swap: 976744 69024 907720 This box aparantly has only 6028kb availble. However if you ignore memory that the kernel temporarily uses for its own optimizations (buffering and such) it actually has almost 78720 kb free. So Asterisk might have a memory leak, and I am trying to find it. To debug memory allocations, build asterisk with memory debugging. Probably a lot less overhead than valgrind. Look for astmm. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI problems - no D channel
Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] spanDSP + rxfax
Hi, I trying to recieve fax with spanDSP and rxfax. scheme: asterisk---sipUA(audiocodes mp-112)--fax(hp5610). I'm allways getting by from UA at the beginning of transmission. Hi does that because sees that rtp connection is broken. I suppose my multifunction hp box is not working correctly but I don't have other to try ;). Maybe somebody from debug may say thats wrong? Aug 17 09:53:58 DEBUG[704] chan_sip.c: Setting NAT on RTP to 0 Aug 17 09:53:58 DEBUG[704] chan_sip.c: Checking SIP call limits for device 123 Aug 17 09:53:58 DEBUG[704] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED] Aug 17 09:53:58 VERBOSE[704] logger.c: -- Executing Answer(SIP/123-df9b, ) in new stack Aug 17 09:53:58 VERBOSE[704] logger.c: -- Executing RxFAX(SIP/123-df9b, /home/minkpr/test.tif|debug) in new stack Aug 17 09:53:58 DEBUG[704] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Match Found Aug 17 09:53:59 DEBUG[704] app_rxfax.c: FLOW HDLC carrier up Aug 17 09:53:59 DEBUG[704] app_rxfax.c: FLOW HDLC carrier down Aug 17 09:53:59 DEBUG[704] app_rxfax.c: FLOW HDLC carrier up Aug 17 09:53:59 DEBUG[704] app_rxfax.c: FLOW HDLC carrier down Aug 17 09:54:01 DEBUG[704] app_rxfax.c: FLOW Changed from phase 1 to 4 Aug 17 09:54:01 DEBUG[704] app_rxfax.c: FLOW DIS:Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 00Aug 17 09:54:01 DEBUG[704] app_rxfax.c: ceAug 17 09:54:01 DEBUG[704] app_rxfax.c: f4Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 81Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:01 DEBUG[704] app_rxfax.c: 18Aug 17 09:54:01 DEBUG[704] app_rxfax.c: Aug 17 09:54:03 DEBUG[704] app_rxfax.c: FLOW HDLC underflow in state 9 Aug 17 09:54:03 DEBUG[704] app_rxfax.c: FLOW Changed from phase 4 to 3 Aug 17 09:54:06 DEBUG[704] app_rxfax.c: FLOW T4 timeout in state 9 Aug 17 09:54:06 DEBUG[704] app_rxfax.c: FLOW Changed from phase 3 to 4 Aug 17 09:54:06 DEBUG[704] app_rxfax.c: FLOW DIS:Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 00Aug 17 09:54:06 DEBUG[704] app_rxfax.c: ceAug 17 09:54:06 DEBUG[704] app_rxfax.c: f4Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 81Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:06 DEBUG[704] app_rxfax.c: 18Aug 17 09:54:06 DEBUG[704] app_rxfax.c: Aug 17 09:54:07 DEBUG[704] app_rxfax.c: FLOW T2 timeout Aug 17 09:54:07 DEBUG[704] app_rxfax.c: FLOW Start receiving document Aug 17 09:54:07 DEBUG[704] app_rxfax.c: FLOW DIS:Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 00Aug 17 09:54:07 DEBUG[704] app_rxfax.c: ceAug 17 09:54:07 DEBUG[704] app_rxfax.c: f4Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 81Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:07 DEBUG[704] app_rxfax.c: 18Aug 17 09:54:07 DEBUG[704] app_rxfax.c: Aug 17 09:54:08 DEBUG[704] app_rxfax.c: FLOW HDLC underflow in state 9 Aug 17 09:54:08 DEBUG[704] app_rxfax.c: FLOW Changed from phase 4 to 3 Aug 17 09:54:08 DEBUG[704] app_rxfax.c: FLOW HDLC carrier up Aug 17 09:54:08 DEBUG[704] app_rxfax.c: FLOW HDLC framing OK Aug 17 09:54:10 DEBUG[704] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 3: Match Found Aug 17 09:54:10 VERBOSE[704] logger.c: == Forcing Marker bit, because SSRC has changed Aug 17 09:54:10 DEBUG[704] app_rxfax.c: FLOW DCS:Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 83Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 00Aug 17 09:54:10 DEBUG[704] app_rxfax.c: c6Aug 17 09:54:10 DEBUG[704] app_rxfax.c: f4Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 80Aug 17 09:54:10 DEBUG[704] app_rxfax.c: 00Aug 17 09:54:10 DEBUG[704] app_rxfax.c: Aug 17 09:54:10 DEBUG[704] app_rxfax.c: FLOW DCS with final frame tag Aug 17 09:54:10 DEBUG[704] app_rxfax.c: FLOW In state 9 Aug 17 09:54:10 DEBUG[704] app_rxfax.c: FLOW Get at 9600bps, modem 1 Aug 17 09:54:10 DEBUG[704] app_rxfax.c: FLOW Changed from phase 3 to 5 Aug 17 09:54:10 DEBUG[704] app_rxfax.c: FLOW Non-ECM carrier up Aug 17 09:54:11 DEBUG[704] app_rxfax.c: FLOW Non-ECM carrier down Aug 17 09:54:11 DEBUG[704] app_rxfax.c: FLOW Non-ECM carrier up Aug 17 09:54:11 DEBUG[704] app_rxfax.c: FLOW Non-ECM carrier trained Aug 17 09:54:12 DEBUG[704] app_rxfax.c: FLOW
Re: [asterisk-users] valgrind + Asterisk
Hi, Tzafrir Cohen wrote: On Thu, Aug 17, 2006 at 02:37:52PM +0200, yusuf wrote: Hi, has anybody got valgring to work with asterisk Yes i do a -- valgrind --tool=memcheck -v asterisk -c then Asterisk just dies. What version of asterisk? Did you use any special build options to build it? asterisk 1.2.1 I went into asterisk source and did a 'make valgrind' The problem I have is that on the box I have Asterisk running, the memory is reported as being used up, then when there is liitle ram left, the box just hangs. What do you mean? Asterisk has just hung now. When I go asterisk -rvvv i only get Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = even a killall asterisk does not help. total used free sharedbuffers cached Mem: 1035276 320380 714896 0 61272 190624 -/+ buffers/cache: 68484 966792 Swap: 2096472 02096472 [EMAIL PROTECTED]:~$ free total used free sharedbuffers cached Mem:237124 231096 6028 0 35700 36992 -/+ buffers/cache: 158404 78720 Swap: 976744 69024 907720 This box aparantly has only 6028kb availble. However if you ignore memory that the kernel temporarily uses for its own optimizations (buffering and such) it actually has almost 78720 kb free. So Asterisk might have a memory leak, and I am trying to find it. To debug memory allocations, build asterisk with memory debugging. Probably a lot less overhead than valgrind. Look for astmm. thanks, I will try this :) -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BRI-PRI switching and synchronization (data/fax calls)
Hello, I found some mentions in archives, but nothing specific - maybe you can point me somewhere if this subject was already discussed... How do you deal with switching of sensitive traffic (analog data calls, faxes, ISDN data) from BRI via Asterisk to PRI? Till now I have experiences only with BRI and I know, that in BRI only environment there is this problem solved (normal TDM switching between ports of junghanns or beronet cards or via PCM bus with multiple cards). But what are the plans with BRI - PRI? I know there is scheduled introduction of 4-port BRI from Digium. Will be possible to interconnect B410P and for example TE110P to create PCM bus and/or to keep synchronization between cards OK for sensitive traffic? And what are your experiences with BRI-(Asterisk)-PRI in current situation without this means. Is it usable for data/fax calls? Thank you, Michal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange behaviour of a zaptel device
Hi! I am working hard on getting a useful attented transfer. (The built-in atxfer feature isnt useful - because of calls getting lost - has been discussed a few months ago) I have all my analog phones on sipura boxes. With the flash hook i can do such attended transfers without problems now. But, the asterisk box is connected to a POTS line via a digium card. And here I have a strange behaviour: .) a call comes in (via the digium card) .) Person A takes the call (on a sipura pap or spa device) .) Person A presses the flash button and dials an other extension for Person B .) Person B hooks the phone off (sipura pap or spa), talks a few words with Person A .) Person A hooks the phone on. .) the incoming call is transferred to Person B, BUT the caller on the incoming call cant hear Person B, while Person B can hear the incoming caller! Does anyone have any hints for me ? Asterisk at log level 10 doesnt show anything... :-( thx in advance, Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 11:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: Oh, and I see nufone caters to residential. We only cater to business customers, who's needs are a lot more demanding. Apparently you haven't actually gone to our website which, since you brought it up, will be re-launched on September 5th, 2006 with new support for ENTERPRISE AND CARRIER SOLUTIONS - All Powered by Asterisk. Apparently you can't read, because this is all I see. On Tuesday, September 5th, 2006 at 8 AM Eastern, we will launch our exciting new website. Join our announcement list to learn about many of the new services and features you can expect from NuFone. We will also launch an official support team, which I am very glad to say that I WILL NOT be a part of, because as most people here know, I yell at stupid people. Good luck with supporting enterprise and carrier solutions with 3 contexts. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Assigning specific RTP ports to SIP clients
Dear All, Is there a way of getting asterisk to assign specific UDP ports for RTP communication between SIP clients. We have a dodgy SIP device that will only send it's RTP audio out to port 8000. I have tried fiddling with rtp.conf, but this is global, restricting it to just one RTP port stops other SIP devices from working (duh). If the above isn't possible, is there a way of finding which port has been assigned? A small iptables script could then divert the RTP stream to the correct port. Thanks, Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pri rdnis found as Facility but not set
Hi, I'm running asterisk with a PRI. But I can't get hold of the rdnis number. When running pri debug I can see the true rdnis number as Facility, the number 703289840 as shown below. Is it possible to get hold of this value in some way from extensions.conf? Or is it necessary to modify the source for asterisk, in that case does someone know where and how? Thanks in advance, Henrik Protocol Discriminator: Q.931 (8) len=79 Call Ref: len= 2 (reference 40/0x28) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] [1c 26 91 a1 23 02 02 00 80 02 01 0f 30 1a 02 01 01 0a 01 02 a1 12 a0 10 a1 0e 0a 01 02 12 09 37 30 33 32 38 39 38 34 30] Facility (len=40, codeset=0) [ 0x91, 0xa1, 0x23, 0x02, 0x02, 0x00, 0x80, 0x02, 0x01, 0x0f, '0', 0x1a, 0x02, 0x01, 0x01, 0x0a, 0x01, 0x02, 0xa1, 0x12, 0xa0, 0x10, 0xa1, 0x0e, 0x0a, 0x01, 0x02, 0x12, 0x09, '703289840' ] [1e 02 84 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0a 21 83 31 38 31 33 34 32 35 35] Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '18134255' ] [70 05 c1 38 35 35 36] Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8556' ] -- Making new call for cr 40 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Aug 17 16:36:39 WARNING[31243]: chan_zap.c:8379 pri_dchannel: PRI_EVENT_RING Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 40/0x28) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] -- Accepting call from '18134255' to '8556' on channel 0/10, span 1 -- Executing Answer(Zap/10-1, ) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 40/0x28) (Terminator) Message type: CONNECT (7) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing NoOp(Zap/10-1, name: ) in new stack -- Executing NoOp(Zap/10-1, number: 18134255) in new stack -- Executing NoOp(Zap/10-1, ani: 18134255) in new stack -- Executing NoOp(Zap/10-1, dnid: 8556) in new stack -- Executing NoOp(Zap/10-1, rdnis: ) in new stack -- Executing Goto(Zap/10-1, test|1) in new stack -- Goto (default,test,1) -- Executing Answer(Zap/10-1, ) in new stack -- Executing Wait(Zap/10-1, 1) in new stack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sangoma a102: Rx Error: 'Retry' exceeds maximum (64k): pci fatal error
Hi, after booting my server I get the following error: Aug 17 16:40:57 localhost kernel: wanpipe1: Disable E1 CAS signalling mode! Aug 17 16:40:57 localhost kernel: wanpipe2: Disable E1 CAS signalling mode! Aug 17 16:40:57 localhost kernel: Registered tone zone 11 (Italy) Aug 17 16:40:58 localhost kernel: wanpipe1:w1g1: Rx Error: 'Retry' exceeds maximum (64k): pci fatal error! (0x0007C03C) Aug 17 16:40:58 localhost kernel: wanpipe1:w1g1: Rx Error: 'Retry' exceeds maximum (64k): pci fatal error! (0x0007C03C) This happens with wanpipe-beta7-2.3.4 and wanpipe 2.3.3-3 and only after boot. If I make a wanrouter stop and then wanrouter start, I get no message. Is there anybody who knows what this means? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 480 Temporarily Unavailable message
Hi everybody! I have a SIP peer correctly registered on my asterisk server (Status: OK (2ms)). I can call the peer normally from another peers, os th DND is no set. But sometimes I got -- Got SIP response 480 Temporarily Unavailable back from 172.16.34.17 -- SIP/XXX-d910 is circuit-busy The peer never loses its registry and there are no packet losses between it and the server. Can someone help me debug and resolve this problem? Thanks a lot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Large Asterisk system
Hi, what hardware (and distro) would you recommend to run an * box with 2500 SIP peers each doing 100 minutes of calls a month? No transcoding at all - all calls will be g729. According to my calcs if the calls are mainly spread over an 8 hour period each day that is (2500*100)/(8*60*30) = 17.36 Thus an average of 17-18 simultaneous calls at any given time but obviously it will peak much higher than that. Tx Marnus van Niekerk -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
OT: Re: [asterisk-users] Asterisk 'Hosting'
Jeremy McNamara wrote: Douglas Garstang wrote: Oh, and I see nufone caters to residential. We only cater to business customers, who's needs are a lot more demanding. Apparently you haven't actually gone to our website which, since you brought it up, will be re-launched on September 5th, 2006 with new support for ENTERPRISE AND CARRIER SOLUTIONS - All Powered by Asterisk. We will also launch an official support team, which I am very glad to say that I WILL NOT be a part of, because as most people here know, I yell at stupid people. *snipped eh? what was that? i couldn't hear you. giggles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Parking initiator cannot retrieve parked calls
Greets all. Having an issue with parking calls. Here is the scenario Calls comes in -- Answered -- Placed on Park -- Go to retrieve parked call -- Can't This only happens for the person who answered the call (Initiator). They can't retrieve the calls they parked. Anyone else can, and if they repark it, then and only then can the initiator pick it back up but I'm looking for a way to not place a call on hold, then park it... I would like to be able to do the following: Call arrives -- Answered -- Place on Park -- if need be retrieve it Calls are parked to page over an intercom so if someone is not available, I need to be able to pick that call back up. Now parking is semi working because I can pick up the call on the parked extension from any other phone except the one that parked it. Eventually it will ring back if the person doesn't respond but I'd like to be able to get it back at will... J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 Fingerprint: 7B02 28CF 24D3 ACA7 9907 789A 8772 7736 1383 A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
VPS is a resource hogger. Not worth the cost. I would stick to multiple contexts. - Original Message - From: Pablo L. Arturi [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 16, 2006 3:16 PM Subject: Re: [asterisk-users] Asterisk 'Hosting' and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Use VPSs, like www.openvz.org Pablo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
No thats CF's job. But to give him credit he yells at stupidity ;) - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 17, 2006 6:15 AM Subject: Re: [asterisk-users] Asterisk 'Hosting' Jeremy McNamara wrote: as most people here know, I yell at stupid people. Be honest, Jeremy, you yell at everyone! -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing CID
I have my phonecompany and the traditional SetCallerID(7325551212) does not work for changing caller ID. I got the following in an email from them but I dont understand it. Can anyone shed light ? Thanks. Dovid I think they could achieve what they want by doing the following: Change the From: to be the CID number that they want Add a CC-Diversion: field with the number that they have an account with us. All they have to do is add a “CC-Diversion” header to their INVITE with the number they want to authenticate off of. We will use this first, before the “Remote-Party-ID” or “From” headers. Example: CC-Diversion: sip:[EMAIL PROTECTED]:5060. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] strange behaviour of a zaptel device SOLVED
Thomas Artner wrote: Hi! I am working hard on getting a useful attented transfer. (The built-in atxfer feature isnt useful - because of calls getting lost - has been discussed a few months ago) I have all my analog phones on sipura boxes. With the flash hook i can do such attended transfers without problems now. But, the asterisk box is connected to a POTS line via a digium card. And here I have a strange behaviour: .) a call comes in (via the digium card) .) Person A takes the call (on a sipura pap or spa device) .) Person A presses the flash button and dials an other extension for Person B .) Person B hooks the phone off (sipura pap or spa), talks a few words with Person A .) Person A hooks the phone on. .) the incoming call is transferred to Person B, BUT the caller on the incoming call cant hear Person B, while Person B can hear the incoming caller! in sip.conf: canreinvite = no solved this issue Does anyone have any hints for me ? Asterisk at log level 10 doesnt show anything... :-( thx in advance, Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to add prefix 0 (zero) when coming from ISDNtrunk
For an incoming call ?
exten = s,1,SetCallerID(${CALLERIDNAME} 0${CALLERIDNUM}
- Original Message -
From: kitti jaisong [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, August 17, 2006 12:46 AM
Subject: [asterisk-users] how to add prefix 0 (zero) when coming from
ISDNtrunk
Dear all
operator network send caller id to asterisk.it doesn't have prefix (0)
how can add prefix to caller id
ti
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[asterisk-users] Registration Error
G'Day List; I hoping for some direction here: The following messageisscrolling without end on my asterisk box, continuously: (NOTE: date and time changes accordingly and IP addresses are not real) Aug 17 11:49:53 NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' Failed for '64.64.64.12' Aug 17 11:49:53 NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' Failed for '64.64.12.12' Aug 17 11:49:53 NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' Failed for '64.64.12.12' Aug 17 11:49:53 NOTICE[1034]: chan_sip.c:8038 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' Failed for '64.64.12.12' Just so you know, the asterisk box sits on a public IP (64.64.64.64) that's on the same subnet as my firewall(64.64.64.12), behind which, my 7960 sits. Any thoughts? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recent additions to the Digium Asterisk development team
Keep up the Excellent work - bravo Digium!Rob LithConnection Telecom - South AfricaOn 17/08/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:Some of you may have noticed some new people with '@ digium.com' email addresses lately... yes, we have been hiring to expand our Asterisk development team and I should have made an official announcement some time ago :-)Joshua Colp joined our development team a few months ago. Josh (file on IRC/Mantis) has been working on Asterisk development for quite some time and had contributed many features and bug fixes as a volunteer community member, along with being very active on the IRC channels and issue tracker. Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's _expression_ parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements. Jason Parker joined our development team at the beginning of this week. Jason (qwell on IRC/Mantis) has been maintaining the chan_skinny driver for Cisco SCCP phones as well acting as a bug marshal and fixing various bugs in Asterisk for the past year or more. Russell Bryant has been a Digium part-time employee and an active Asterisk maintainer since before I got involved with Asterisk :-) His contributions are innumerable, and he has worked far more than the 'ten to twenty hours per week' he claims to have available outside of his school work! Russell (russellb on IRC/Mantis) will be joining us full time in Huntsville after the winter semester is complete, when he expects to graduate. Please join me in welcoming all these new members of our development team; they are helping to make Asterisk (and our other software products) better every day and will enable us to accelerate our products into the future. --Kevin P. FlemingSenior Software EngineerDigium, Inc.___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I can´t set to work two tdm2400p and one TE205p on same machine, please help
Hi, I´m trying to set up this three boards, two tdm240p (with 32 FXS ports) and one TE205p, I appreciate if anyone could take a look at my config files to see what´s wrong. If I set only the TDM240p or only the TE205p they work fine separately, just can´t make them work together, this is my zaptel, I do ztcfg and shows no errors span=1,1,0,ccs,hdb3 span=2,1,0,ccs,hdb3 bchan=1-15 bchan=17-31 dchan=16 bchan=32-46 bchan=48-62 dchan=47 fxols=63-94 this is my zapata.conf [channels] language=en usecallerid=yes hidecallerid=no callerid=asreceived restrictcid=no usecallingpres=yes switchtype=euroisdn resetinterval=240400 signalling=pri_cpe immediate=no pridialplan=unknown prilocaldialplan=unknown priindication=outofband chocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=-3.0 txgain=-6.0 context=pri_public group=0 channel=1-15 channel=17-31 channel=32-46 channel=48-62 signalling=fxo_ls context=default immediate=no group=1 overlapdial=no channel=63-94 this is the output in messages: Apr 12 10:44:04 NOTICE[6982] cdr.c: CDR simple logging enabled. Apr 12 10:44:04 WARNING[6987] res_musiconhold.c: Found no files in '/usr/share/asterisk/mohmp3' Apr 12 10:44:04 WARNING[6987] res_musiconhold.c: Unable to spawn mp3player Apr 12 10:44:04 NOTICE[6982] config.c: Registered Config Engine odbc Apr 12 10:44:04 NOTICE[6982] res_odbc.c: Adding ENV var: INFORMIXSERVER=my_special_database Apr 12 10:44:04 NOTICE[6982] res_odbc.c: Adding ENV var: INFORMIXDIR=/opt/informix Apr 12 10:44:04 NOTICE[6982] res_odbc.c: registered database handle 'asterisk' dsn-[asterisk] Apr 12 10:44:04 NOTICE[6982] res_odbc.c: Connecting asterisk Apr 12 10:44:04 WARNING[6982] res_odbc.c: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified Apr 12 10:44:04 NOTICE[6982] res_odbc.c: res_odbc loaded. Apr 12 10:44:04 WARNING[6982] chan_iax2.c: Error opening firmware directory '/usr/share/asterisk/firmware/iax': No such file or directory Apr 12 10:44:04 WARNING[6982] chan_zap.c: Unable to specify channel 63: No such device Apr 12 10:44:04 ERROR[6982] chan_zap.c: Unable to open channel 63: No such device here = 0, tmp-channel = 63, channel = 63 Apr 12 10:44:04 ERROR[6982] chan_zap.c: Unable to register channel '63-94' Apr 12 10:44:04 WARNING[6982] loader.c: chan_zap.so: load_module failed, returning -1 Apr 12 10:44:04 WARNING[6982] loader.c: Loading module chan_zap.so failed! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial statement problem
Need a little assist by someone else's eyes; mine have gone blurry.
Running v1.2.10 checked out from svn as of today.
Problem: When dial statement is executed with a timeout value and no one
answers the call, the next priority (#4) is not being executed as expected.
When an incoming pstn call arrives, the zap/4 channel properly handles
the call and sends it to the [inbound-bus-line] context. The CLI for a
sample call appears just below the following extensions.conf paste.
When the 20 second Dial() timeout occurs, step #4 is not executed.
Rather, the next dialplan entry executed is from the next context that
immediately follows. Why?
Portion of extensions.conf:
[inbound-bus-line]
exten = s,1,NoOp,${CALLERID(all)}
exten = s,2,NoOp,bus-line-step2
exten = s,3,Dial(${PHONE1}${PHONE2}|20)
exten = s,4,NoOp,bus-line-step3
exten = s,5,Goto(bus-ivr-main|s|1)
exten = s,104,NoOp,bus-line-step103
exten = s,105,Goto(bus-ivr-main|s|1)
[inbound-bus-dialin] ; goes directly to IVR and allows most dialplan dialing
include = local-extns
include = misc-extns
include = outgoing-calls
include = parkedcalls
exten = s,1,NoOp,${CALLERID(all)}
exten = s,2,NoOp,bus-dialin-step
exten = s,3,Answer
exten = s,4,Goto(bus-ivr-main|s|1)
snip
[bus-ivr-main]
exten = s,1,Wait,1
exten = s,2,NoOp,step 2
exten = s,3,Answer
exten = s,4,Set(TIMEOUT(digit)=5)
exten = s,5,Set(TIMEOUT(response)=10)
exten = s,6,NoOp,${CALLERID(all)}
exten = s,7,Background(npi-greeting) ; Thanks for calling press 1 for
snip
phoenix*CLI
-- Starting simple switch on 'Zap/4-1'
Aug 17 11:44:52 NOTICE[15342]: chan_zap.c:6073 ss_thread: Got event 2
(Ring/Answered)...
-- Executing NoOp(Zap/4-1, Adamson Richard 402432) in
new stack
-- Executing NoOp(Zap/4-1, bus-line-step2) in new stack
-- Executing Dial(Zap/4-1, SIP/3000SIP/3001|20) in new stack
-- Called 3000
-- Called 3001
-- SIP/3000-09eed5e0 is ringing
-- SIP/3001-09ef2b20 is ringing
Note: problem starts here. The GoTo in [inbound-bus-line] step #5 is
not executed. Rather, dialplan processing continues in the next context.
-- Starting simple switch on 'Zap/2-1'
-- Executing NoOp(Zap/2-1, ) in new stack
-- Executing NoOp(Zap/2-1, bus-dialin-step) in new stack
-- Executing Answer(Zap/2-1, ) in new stack
-- Executing Goto(Zap/2-1, bus-ivr-main|s|1) in new stack
-- Goto (bus-ivr-main,s,1)
-- Executing Wait(Zap/2-1, 1) in new stack
-- Executing NoOp(Zap/2-1, step 2) in new stack
Any help would be greatly appreciated.
Rich
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Re: [asterisk-users] PRI problems - no D channel
What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI problems - no D channel
The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI problems - no D channel
My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] astbill white screen!!
Hi all, I've installed asterisk and astbill according with all recommendation (mysql5, drupal included with astbill, php, apache2...). When I write http://server_adress/astbill, I get a white screen page. Browser doesn´t give me an error page, it just a white screen page. However asterisk doesnt have any problem, and works well with mysql. I also have installed Drupal 4.7.3 linked to other database with other user and password working well. And I have phpMyAdmin too. All working very good at the same server. I tried changing index.php to phpinfo.php in the same directory and it works well too. Can anybody help me with that please? Any suggestion will be very appreciated. Thanks, very much in advance Sebastian On 7/14/06, varun [EMAIL PROTECTED] wrote: Hello, Our asterisk server is on Centos 4.2 We want to use Astbill. Astbill requires Drupal and mysql 5. I could not find rpms mysql5 for centos. We are getting mysql extensions issues because of php-mysql. How do we solve this ? Any other billing software that similar to Astbill ? Thanks Varun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] em wink, TE110P, * answers too soon
Well you say too much and not enough about the problem or configuration So, I assume the DID's are on Ports 1 - 24 T1?. If asterisk is missing the first digit, then I'll bet the DID T1 from Telco is set to immediate on their side, not wink - Because dialing should NOT start until after the wink from asterisk - Try changing Telco T1 to immediate start and test. Bart Steve Linabery wrote: Hi, I've been googling all over the place and have read the relevant articles in the Digium knowledge base. I have tried all the suggestions I found in the K.B. Spent some time on the asterisk irc, tweaking some parameters as people thereon thought would be helpful, but to no avail. I am trying to set up * on an em wink trunk currently attached to an Avaya Merlin Magix system. The provider of the T1 is McLeodUSA; our location is St Paul MN USA. I am in the process of getting more specific timing information from their tech support, but it takes days. I can call into the * PBX from my cell phone just fine. I can call between the two grandstream phones I bought for testing just fine. Here's the problem. When a call comes into *, * attempts to route it to an extension prematurely. For example, if the DTMF digits coming from upstream are '538', * tries to send the call to extn '53'. I still receive the '8', but too late. Here's a snip from /var/log/asterisk/messages where the incoming DID digits are '535': Aug 7 22:30:00 DEBUG[31492] chan_zap.c: Monitor doohicky got event Ring/Answered on channel 1 Aug 7 22:30:00 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 2 (In use) Aug 7 22:30:00 VERBOSE[31493] logger.c: Asterisk Ready. -- Starting simple switch on 'Zap/1-1' Aug 7 22:30:00 DEBUG[31494] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Aug 7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1 Aug 7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 3 on Zap/1-1 Aug 7 22:30:01 DEBUG[31493] chan_zap.c: Enabled echo cancellation on channel 1 Aug 7 22:30:01 VERBOSE[31493] logger.c: == Unknown extension '53' in context 'demo' requested Aug 7 22:30:04 DEBUG[31493] channel.c: Set channel Zap/1-1 to write format gsm Aug 7 22:30:04 DEBUG[31493] channel.c: Scheduling timer at 160 sample intervals Aug 7 22:30:04 VERBOSE[31493] logger.c: -- Playing 'ss-noservice' (language 'en') Aug 7 22:30:04 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Exception on 20, channel 1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Got event On hook(1) on channel 1 (index 0) Aug 7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on channel 1 Aug 7 22:30:07 DEBUG[31493] channel.c: Scheduling timer at 0 sample intervals Aug 7 22:30:07 DEBUG[31493] channel.c: Hanging up channel 'Zap/1-1' Aug 7 22:30:07 DEBUG[31493] chan_zap.c: zt_hangup(Zap/1-1) Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Hangup: channel: 1 index = 0, normal = 20, callwait = -1, thirdcall = -1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on channel 1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Aug 7 22:30:07 DEBUG[31493] chan_zap.c: Updated conferencing on 1, with 0 conference users Aug 7 22:30:07 VERBOSE[31493] logger.c: -- Hungup 'Zap/1-1' Aug 7 22:30:07 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 0 (Unknown) Aug 7 22:30:07 DEBUG[31495] app_queue.c: Device 'Zap/1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Here are some settings from /etc/asterisk/zapata.conf: [trunkgroups] [channels] wink=300 rxwink=300 start=3000 context=default switchtype=national toneduration=100 usecallerid=no cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no switchtype = national context = demo signalling = em_w group = 1 channel = 1-20 It has occurred to me that I could just set immediate=yes, read the incoming DTMF digits into a variable, and route to the appropriate extension. That seems more fragile to me since we could someday (when I'm not here) start getting more than 3 digits (caller id, for example). Plus I'd like to make it work the way it's *supposed* to. Any help/suggestions are appreciated! Cheers, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astbill white screen!!
I've installed asterisk and astbill according with all recommendation (mysql5, drupal included with astbill, php, apache2...). When I write http://server_adress/astbill, I get a white screen page. Browser doesn´t give me an error page, it just a white screen page. you have to enable it in php settings. Go in /etc/php.ini - change setting error_reporting to E_ALL - change setting display_errors to On - restart apache now, at least, it will tell you what goes wrong N.B.: display_errors should not be enabled on a production server hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: modprobe wctdm fails in /etc/rc.local on FC5
Inline: -- -- Steven http://www.glimasoutheast.org Tzafrir Cohen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Below is a good example of misusing init.d scripts with the obvious results of the need for ugly workarounds. Why do you fight against the design of your system? See specific comments below, On Wed, Aug 16, 2006 at 08:20:55AM -0400, Steven wrote: rc.local: touch /var/lock/subsys/local setpci -v -s 00:1f.1 LATENCY_TIMER=4 setpci -v -s 02:0e.0 LATENCY_TIMER=4 setpci -v -s 0b:07.0 LATENCY_TIMER=4 setpci -v -s 0c:08.0 LATENCY_TIMER=4 setpci -v -s 10:0d.0 LATENCY_TIMER=0 setpci -v -s 06:02.0 LATENCY_TIMER=ff sleep 5 Why a sleep here? What exactly do you wait for? This was just leftover from previous testing. sleep is an indication of a bad workaround for a race condition. It is a bad workaround, as you can never know if you sleep enough. echo UnLoading wct4xxp rmmod -v wct4xxp rmmod -v zaptel sleep 3 Huh? Why is that? When I restart asterisk I: 1. stop now on the asterisl CLI. 2. execute rc.local which unloads and reloads zaptel. It is not used but does no harm when booting the machine. BTW: 'genzaptelconf -u' will unload all zaptel modules. If 'modprobe -r' works on your system: that's even better. I will have to look into that. Why is 'modprobe -r' better? I do think that I have genzaptelconf on my machine. Isn't that for [EMAIL PROTECTED] echo Loading wct4xxp /sbin/modprobe -v zaptel sleep 5 /sbin/modprobe -v wct4xxp sleep 5 # ztcfg - #sleep 5 The 'sleep'-s here are because of the following bugs: 1. running ztcfg automatically on modules load 2. not properly waiting for /dev/zap/ctl to be generated by udev These sleeps were required for issue number 2 above. the ztcfg is not used. was there for previous testing. echo 1 /proc/irq/201/smp_affinity echo 1 /proc/irq/217/smp_affinity echo 0 /proc/irq/209/smp_affinity echo 1 /proc/irq/14/smp_affinity /usr/sbin/amportal start Now, that's an init.d script. Most of the code above should be merged into the zaptel init.d script (which is run before asterisk/amportal). Then the init system will run amportal later. If actually needed. I can read your tone here. This is an rc.local, not init.d script. That is why amportal start is chained to the end of the file. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Chris Mason (Lists) wrote: Be honest, Jeremy, you yell at everyone! I wouldn't say absolutely everyone - I don't think I've yelled at kram or kpfleming, yet :P Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5
I just reread the post I replied to. I thought that they were asking for a ref. rc.local. I didn't notice that it said modprobe.conf. alias eth0 e1000 alias eth1 e1000 alias scsi_hostadapter megaraid_mbox alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd options torisa base=0xd install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp /sbin/ztcfg install pciradio /sbin/modprobe --ignore-install pciradio /sbin/ztcfg install ztd-loc /sbin/modprobe --ignore-install ztd-loc /sbin/ztcfg install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg alias wcfxs wctdm alias wct2xxp wct4xxp -- -- Steven http://www.glimasoutheast.org Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] rc.local: touch /var/lock/subsys/local setpci -v -s 00:1f.1 LATENCY_TIMER=4 setpci -v -s 02:0e.0 LATENCY_TIMER=4 setpci -v -s 0b:07.0 LATENCY_TIMER=4 setpci -v -s 0c:08.0 LATENCY_TIMER=4 setpci -v -s 10:0d.0 LATENCY_TIMER=0 setpci -v -s 06:02.0 LATENCY_TIMER=ff sleep 5 echo UnLoading wct4xxp rmmod -v wct4xxp rmmod -v zaptel sleep 3 echo Loading wct4xxp /sbin/modprobe -v zaptel sleep 5 /sbin/modprobe -v wct4xxp sleep 5 # ztcfg - #sleep 5 echo 1 /proc/irq/201/smp_affinity echo 1 /proc/irq/217/smp_affinity echo 0 /proc/irq/209/smp_affinity echo 1 /proc/irq/14/smp_affinity /usr/sbin/amportal start -- -- Steven http://www.glimasoutheast.org Robert La Ferla [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Can someone send me their modprobe.conf file? I think that may be the problem. A zaptel file is created during install in /etc/ modprobe.d but modprobe.conf must need to reference it... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote: Good luck with supporting enterprise and carrier solutions with 3 contexts. Mr. Troll, I don't need luck, because I am doing it already. Perhaps you can't comprehend the fact that NuFone is not the only operation I am involved with. Plus, don't forget about the thousands of hours or more that I have billed out consulting others on their system design. Go away. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail and Fax on same extension
Hi, I'm trying to accomplish having a single extension that always answers with an automated voicemail prompt and record a user message, but can recognize if the call is fax and handle it accordingly. Anyone here has any experience with this kind of configuration? Thanks, Adi. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI problems - no D channel
Quoting C F [EMAIL PROTECTED]: My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. That was the first thing I tried: first trying to unload/reload the wct1xxp and zaptel modules, then reload them, then tried rebooting the computer (multiple times), then this morning, I gave it a 30 minute time out. No effect - still getting the D-chan errors. Unfortunately, the system is some 90 miles north of here so I can't verify if anything on the Adtran has changed or not (or reseat cables). My remote hands aren't available right now either so I can't verify anything regarding the circuit at this time. The fact that I am not getting any error reports at all about the transport (HDLC type errors) tends to make me think that the circuit is fine. I would have to imagine that if the channel switched from PRI to T1 it would throw all kinds of errors. Same thing if the signaling or buildout or what not was incorrect. The only errors I am seeing are the ones about the D channel not being there, and one (probably quite related) about head of queue has not been transmitted yet. Ron On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: Good luck with supporting enterprise and carrier solutions with 3 contexts. Mr. Troll, I don't need luck, because I am doing it already. Perhaps you can't comprehend the fact that NuFone is not the only operation I am involved with. Plus, don't forget about the thousands of hours or more that I have billed out consulting others on their system design. Go away. That's funny. I remember asking a question, and I remember you immediately attacking my intelligence, and now I'm suddenly a troll. I can comprehend that Nufone is not the operation you are involved with. However, it's the first time you've stated that, so if you think I should have known that already, then you've either a) lost touch with reality or b) think your some big shot and I should know who you are by name. I see you haven't addressed the specifics of my reply to you. Did you eat too much cheese yesterday? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip suppression
snip Hi, When I make a call using asterisk (SIP - provider - PSTN), it is cutting out all the background noise, making it a bit like a walkie talkie when they stop talking. Is this something asterisk is doing?!? I have a 2MB leased line just for calls, so bandwidth is fine, I have tried with G711a/u and GSM codec but they all do the same. I'm using a Polycom IP501 to connect to asterisk, but have the VAD set to 0 in the config file to disable Voice Activity Detection on the phones. Could it be my provider trying to do this? As I have noticed on the CLI 'NOTICE[2530]: rtp.c:593 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible.' /snip Do what it says. Turn it off in the client (i.e. the phone). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI problems - no D channel
Heh...everytime I have lost the D channel it has always been on the provider end. - Original Message - From: Ron Gage [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, August 17, 2006 2:32 PM Subject: Re: [asterisk-users] PRI problems - no D channel Quoting C F [EMAIL PROTECTED]: My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. That was the first thing I tried: first trying to unload/reload the wct1xxp and zaptel modules, then reload them, then tried rebooting the computer (multiple times), then this morning, I gave it a 30 minute time out. No effect - still getting the D-chan errors. Unfortunately, the system is some 90 miles north of here so I can't verify if anything on the Adtran has changed or not (or reseat cables). My remote hands aren't available right now either so I can't verify anything regarding the circuit at this time. The fact that I am not getting any error reports at all about the transport (HDLC type errors) tends to make me think that the circuit is fine. I would have to imagine that if the channel switched from PRI to T1 it would throw all kinds of errors. Same thing if the signaling or buildout or what not was incorrect. The only errors I am seeing are the ones about the D channel not being there, and one (probably quite related) about head of queue has not been transmitted yet. Ron On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.1/421 - Release Date: 8/16/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astbill white screen!!
Great help Time, thanks very much. That's the error I have Fatal error: SELECT command denied to user 'astbill_user'@'localhost' for table 'pbx_users' query: SELECT u.*, s.* FROM pbx_users u INNER JOIN pbx_sessions s ON u.uid = s.uid WHERE s.sid = '2e57e35eeb3e8464a4cc1bd10c1997a9' AND u.status 3 LIMIT 0, 1 in /home/astbill/wwwroot/includes/database.mysql.inc on line 66 I understand I have to GRANT all privileges to astbill_user over the asterisk database. Sebastian On 8/17/06, Time Bandit [EMAIL PROTECTED] wrote: I've installed asterisk and astbill according with all recommendation (mysql5, drupal included with astbill, php, apache2...). When I write http://server_adress/astbill, I get a white screen page. Browser doesn´t give me an error page, it just a white screen page. you have to enable it in php settings. Go in /etc/php.ini - change setting error_reporting to E_ALL - change setting display_errors to On - restart apache now, at least, it will tell you what goes wrong N.B.: display_errors should not be enabled on a production server hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI problems - no D channel
What does the telco say when they test the circuit? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI problems - no D channel Quoting C F [EMAIL PROTECTED]: My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. That was the first thing I tried: first trying to unload/reload the wct1xxp and zaptel modules, then reload them, then tried rebooting the computer (multiple times), then this morning, I gave it a 30 minute time out. No effect - still getting the D-chan errors. Unfortunately, the system is some 90 miles north of here so I can't verify if anything on the Adtran has changed or not (or reseat cables). My remote hands aren't available right now either so I can't verify anything regarding the circuit at this time. The fact that I am not getting any error reports at all about the transport (HDLC type errors) tends to make me think that the circuit is fine. I would have to imagine that if the channel switched from PRI to T1 it would throw all kinds of errors. Same thing if the signaling or buildout or what not was incorrect. The only errors I am seeing are the ones about the D channel not being there, and one (probably quite related) about head of queue has not been transmitted yet. Ron On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote: That's funny. I remember asking a question, and I remember you immediately attacking my intelligence, and now I'm suddenly a troll. I can comprehend that Nufone is not the operation you are involved with. However, it's the first time you've stated that, so if you think I should have known that already, then you've either a) lost touch with reality or b) think your some big shot and I should know who you are by name. You made an obvious assumption by looking at my email address, without bothering to consider any other operations I may be involved in or have assisted in development and deployment. I could care less who you think I am, really and who knows, I may have no clue what reality is - But who really does? I see you haven't addressed the specifics of my reply to you. I don't see any specific questions, only statements that prove you haven't fully comprehended what you have gotten yourself into. I think you are the one that will need some very good luck. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI problems - no D channel
Quoting Bill Gibbs [EMAIL PROTECTED]: What does the telco say when they test the circuit? Bill Bill: I am having my remote hands check first on the Adtran that is feeding the Asterisk box, then then go upstream from there. Thanks for helping me see the obvious path to follow! :) Ron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI problems - no D channel Quoting C F [EMAIL PROTECTED]: My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. That was the first thing I tried: first trying to unload/reload the wct1xxp and zaptel modules, then reload them, then tried rebooting the computer (multiple times), then this morning, I gave it a 30 minute time out. No effect - still getting the D-chan errors. Unfortunately, the system is some 90 miles north of here so I can't verify if anything on the Adtran has changed or not (or reseat cables). My remote hands aren't available right now either so I can't verify anything regarding the circuit at this time. The fact that I am not getting any error reports at all about the transport (HDLC type errors) tends to make me think that the circuit is fine. I would have to imagine that if the channel switched from PRI to T1 it would throw all kinds of errors. Same thing if the signaling or buildout or what not was incorrect. The only errors I am seeing are the ones about the D channel not being there, and one (probably quite related) about head of queue has not been transmitted yet. Ron On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED]
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: That's funny. I remember asking a question, and I remember you immediately attacking my intelligence, and now I'm suddenly a troll. I can comprehend that Nufone is not the operation you are involved with. However, it's the first time you've stated that, so if you think I should have known that already, then you've either a) lost touch with reality or b) think your some big shot and I should know who you are by name. You made an obvious assumption by looking at my email address, without bothering to consider any other operations I may be involved in or have assisted in development and deployment. I could care less who you think I am, really and who knows, I may have no clue what reality is - But who really does? I see you haven't addressed the specifics of my reply to you. I don't see any specific questions, only statements that prove you haven't fully comprehended what you have gotten yourself into. What's not specific about this...? handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster and so on while ALSO letting customers maintain all this via a web interface? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime include
Does realtime support include = yet? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI problems - no D channel
I know but you could save some time and have it tested while waiting...they might find a problem and save you a lot of headache. I can tell you are one of the rare people who actually checks their stuff before calling anyone but like another posted said, D Channels tend to be provider related for some reason! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 3:29 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] PRI problems - no D channel Quoting Bill Gibbs [EMAIL PROTECTED]: What does the telco say when they test the circuit? Bill Bill: I am having my remote hands check first on the Adtran that is feeding the Asterisk box, then then go upstream from there. Thanks for helping me see the obvious path to follow! :) Ron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI problems - no D channel Quoting C F [EMAIL PROTECTED]: My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. That was the first thing I tried: first trying to unload/reload the wct1xxp and zaptel modules, then reload them, then tried rebooting the computer (multiple times), then this morning, I gave it a 30 minute time out. No effect - still getting the D-chan errors. Unfortunately, the system is some 90 miles north of here so I can't verify if anything on the Adtran has changed or not (or reseat cables). My remote hands aren't available right now either so I can't verify anything regarding the circuit at this time. The fact that I am not getting any error reports at all about the transport (HDLC type errors) tends to make me think that the circuit is fine. I would have to imagine that if the channel switched from PRI to T1 it would throw all kinds of errors. Same thing if the signaling or buildout or what not was incorrect. The only errors I am seeing are the ones about the D channel not being there, and one (probably quite related) about head of queue has not been transmitted yet. Ron On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED]
Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote: What's not specific about this...? handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster Again, it tells me you have not fully thought out exactly how each of those functions fit within the realm of Asterisk. Before you even bothered to learn basic asterisk fundamentals, you have went and reinvented the wheel for absolutely no reason. and so on while ALSO letting customers maintain all this via a web interface? And the problem is? Our current members portal already does 90% of everything you listed. When the new version comes out we will pick up the remaining points and perhaps more. Yet once again you have proven beyond any shadow of a doubt, you didn't bother to comprehend the power of Asterisk before bastardizing it with your ungodly huge python script. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI problems - no D channel
Quoting Bill Gibbs [EMAIL PROTECTED]: I know but you could save some time and have it tested while waiting...they might find a problem and save you a lot of headache. I can tell you are one of the rare people who actually checks their stuff before calling anyone but like another posted said, D Channels tend to be provider related for some reason! Very true that is - I just like to be 1000% certain that I am not causing my own headaches! Yes, it's a bit time consuming but it tends to save heavily on the crow when it comes time to figure out where the problem actually lies. Plus it really helps to have all your ducks in a row when dealing with an ILEC - they just don't seem to have much of a sense of humor about these things. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 3:29 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] PRI problems - no D channel Quoting Bill Gibbs [EMAIL PROTECTED]: What does the telco say when they test the circuit? Bill Bill: I am having my remote hands check first on the Adtran that is feeding the Asterisk box, then then go upstream from there. Thanks for helping me see the obvious path to follow! :) Ron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage Sent: Thursday, August 17, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI problems - no D channel Quoting C F [EMAIL PROTECTED]: My guess is it's some Intel mobo. Did you restart the system since? If you did that might be the problem, try restarting and unplug the power for at least 60 seconds before powering it back up. That was the first thing I tried: first trying to unload/reload the wct1xxp and zaptel modules, then reload them, then tried rebooting the computer (multiple times), then this morning, I gave it a 30 minute time out. No effect - still getting the D-chan errors. Unfortunately, the system is some 90 miles north of here so I can't verify if anything on the Adtran has changed or not (or reseat cables). My remote hands aren't available right now either so I can't verify anything regarding the circuit at this time. The fact that I am not getting any error reports at all about the transport (HDLC type errors) tends to make me think that the circuit is fine. I would have to imagine that if the channel switched from PRI to T1 it would throw all kinds of errors. Same thing if the signaling or buildout or what not was incorrect. The only errors I am seeing are the ones about the D channel not being there, and one (probably quite related) about head of queue has not been transmitted yet. Ron On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: The PRI is connected at one end to an Adtran Atlas, I believe a 600. The other end of the PRI is connected to a Digium T100. The two are seperated by roughly 10 foot of cat-5. Motherboard is whatever Dell put into their Precision 530MT line of workstations. Like I said, it worked just fine yesterday and today I am getting D-Channel errors. Thanks for your assistance! Ron Quoting C F [EMAIL PROTECTED]: What is the PRI connected to: What hardware for the T1? What Motherboard? On 8/17/06, Ron Gage [EMAIL PROTECTED] wrote: Hey guys: I am having a bit of a problem with our PRI under Asterisk. I am seeing the following error every 10 seconds... Aug 17 08:54:55 WARNING[2458]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Of course, I have the d-chan defined in /etc/zaptel.conf... loadzone = us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 Neat thing is: this worked yesterday and isn't working today (Asterisk isn't answering the PRI on any inbound DID). zttool shows no problems with the T100 and no alarm conditions. The PRI is being drove by an Adtran Atlas. HELP! Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ron Gage - Westland MI [EMAIL PROTECTED] This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Accessing SIP URI (not ${SIPURI})
How to I access the URI from an Invite:
INVITE sip:[EMAIL PROTECTED]
I want to set a variable to equal 5556678. The variable ${SIPURI} returns
the From URI.
Regards
Cameron
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RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: What's not specific about this...? handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster Again, it tells me you have not fully thought out exactly how each of those functions fit within the realm of Asterisk. I spent 8+ hours a day, 5+ days per week for over 6 months thinking how these functions fit within the realm of Asterisk. At every single turn, after going down every single path, there where limitations that forced us to backtrack and evaluate a different approach. A script that could handle call routing, in conjection with MySQL and stored procedures was the only way to implement our requirements. The MySQL command had limitations, realtime was way too resource intensive, unreliable and undocumented and so on. Yep... i definitely haven't thought about this at all. Before you even bothered to learn basic asterisk fundamentals, you have went and reinvented the wheel for absolutely no reason. and so on while ALSO letting customers maintain all this via a web interface? And the problem is? Our current members portal already does 90% of everything you listed. When the new version comes out we will pick up the remaining points and perhaps more. Yet once again you have proven beyond any shadow of a doubt, you didn't bother to comprehend the power of Asterisk before bastardizing it with your ungodly huge python script. Yet again you have not addressed my statements that showed that the MySQL Dial plan command was not capable of nesting SQL queries, and therefore not capable of implementing findme/followme. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
guys can we take the flame fest off list please? kthx Douglas Garstang wrote: -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: That's funny. I remember asking a question, and I remember you immediately attacking my intelligence, and now I'm suddenly a troll. I can comprehend that Nufone is not the operation you are involved with. However, it's the first time you've stated that, so if you think I should have known that already, then you've either a) lost touch with reality or b) think your some big shot and I should know who you are by name. You made an obvious assumption by looking at my email address, without bothering to consider any other operations I may be involved in or have assisted in development and deployment. I could care less who you think I am, really and who knows, I may have no clue what reality is - But who really does? I see you haven't addressed the specifics of my reply to you. I don't see any specific questions, only statements that prove you haven't fully comprehended what you have gotten yourself into. What's not specific about this...? handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster and so on while ALSO letting customers maintain all this via a web interface? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime Extension Lookups
I'm curious, have always been curious... why does realtime repeat the same query multiple times when lookup up extensions? This is one of the reasons we chose not to use it, but I'm still wondering why it does it? [14:[EMAIL PROTECTED](pbx1):asterisk]# ngrep -d eth0 port 3306 | grep SELECT ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '1' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '1' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '1' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '1' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '1' (FIVE TIMES Why 5 times?) ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '2' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '2' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '2' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '3' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '3' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '3' ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '4' nSELECT * FROM extensions_table WHERE exten LIKE '\_%' AND context = 'test1' AND priority = '4' ORDER BY exten ]SELECT * FROM extensions_table WHERE exten = '1000' AND context = 'test1' AND priority = '1' ZSELECT * FROM extensions_table WHERE exten = 'h' AND context = 'test1' AND priority = '1' nSELECT * FROM extensions_table WHERE exten LIKE '\_%' AND context = 'test1' AND priority = '1' ORDER BY exten Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Festival through AGI can't handle strings longer than 15 chars
Moises, follow on and you'll find the exact output that I got from Asterisk once I raised the more detailed debug level as you suggested. I'm sorry, that's quite a long text, but at least I'm sure you have all the info available. However, I now have SOME IMPORTANT notes to add: 1. I'm running Asterisk 1.2.9.1 2. Once I restarted the server, everything worked fine. Thus, it seems that the problem could be solved by just restarting the server. However... read on. 3. Once restarted the server I changed the text inside my Python AGI and the error appeared again. My CONCLUSION: the error doesn't probably depend on Festival or Exec command. To make the error occur, I simply do this: a) I restart server b) I run the AGI script with whatever text (as wide as I wish) and it will work c) I shorten the text... it still will work d) I then widen again the text: now it won't work! It will work only as long as I shorten the text (or leave it the same length), but not if I widen it. I suspect that there is some malloc()ed area (I can't imagine in which C module) that gets successfully narrowed based on the AGI passed text, but never gets enlarged unless the server restarts... Does it seems reasonable? Hope it helps. Note: this is my 2nd reply. Since I didn't see my 1st reply in the newsgroup, I'm now omitting the console log since it is probably useless once I understood the cause of the problem (what I'm missing is how to fix it). I suspect that because there was too much text, my whole reply has been discarded. Moises Silva wrote: One step more, enable the following in logger.conf console = notice,warning,error,debug,verbose Application app_festival has some interesting debug messages like: ast_log(LOG_DEBUG, Text passed to festival server : %s\n,(char *)data); and that shows in the console the exact test is passed to the festival server. I keep looking into the code trying to find the reason of the behaviour you describe but I havent succed so far. Please report any feedback. Regards On 8/17/06, Mario [EMAIL PROTECTED] wrote: Thanks for your help, Moises. I did activate the AGI DEBUG as you suggested (thanks for that!). However, I'm now only a little bit more sure that I'm passing the right stuff to the Festival command. Following you'll see what I'm passing for the short text (shorter than 15 chars) and for the wider text. As you can see, both the calls seem to work, but for the 2nd I do not hear any sound. At this point, any idea is really welcome. Thanks for your help. *** Short text *** AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx EXEC FESTIVAL Telefono spento -- AGI Script Executing Application: (FESTIVAL) Options: (Telefono spento) == Parsing '/etc/asterisk/festival.conf': Found AGI Tx 200 result=0 -- AGI Script test_command.py completed, returning 0 == Auto fallthrough, channel 'SIP/1-9803' status is 'UNKNOWN' *** Longer text *** AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx EXEC FESTIVAL Telefono utente spento -- AGI Script Executing Application: (FESTIVAL) Options: (Telefono utente spento) == Parsing '/etc/asterisk/festival.conf': Found AGI Tx 200 result=0 -- AGI Script test_command.py completed, returning 0 == Auto fallthrough, channel 'SIP/1-67c2' status is 'UNKNOWN' Moises Silva wrote: Hi Mario. Have you tried to enable AGI debug? CLI agi debug That will show what Asterisk is receiving from your script. Also enable all the debug messages in the logger.conf file for the console Go and try that and post what you see here, and we may be able to help you On 8/17/06, Mario [EMAIL PROTECTED] wrote: I'm having a tough problem when using Festival with Asterisk through AGI: it seems that when I pass more than 15 chars to the Festival command, when from inside an AGI, no sounds (speech) at all is generated. The following (from inside the dialplan) correctly works: exten = 333,1,Answer() exten = 333,2,FESTIVAL(Telefono spento uno) exten = 333,3,Hangup But, when moved from within an AGI, the same Festival command doesn't work: EXEC FESTIVAL Telefono spento uno the symptom is that no text is played, although the return code from command is zero. One important note: if I shorten the text to Telefono spento (i.e. at most 15-chars wide) everything works as expected. I really can't figure out the reason of this weird behavior. What I can do is to exclude some possible reasons: 1. It is not a festival-related problem since when called from the Dialplan everything works as expected. 2. It is not a language-related issue, since I tried this both with English and Italian 3. It is not a missing call to flush()... yes, I added a flush() at the end of my Python-based AGI call 4. It is not a problem related to Python, since I use Python extensively with AGI Does anyone have a hint on what I can do to investigate or solve this problem? Does enyone know if this is a known bug? Thanks in
Re: [asterisk-users] Asterisk 'Hosting'
On Thursday 17 August 2006 16:04, Douglas Garstang wrote: Yet again you have not addressed my statements that showed that the MySQL Dial plan command was not capable of nesting SQL queries, and therefore not capable of implementing findme/followme. Off the top of my head I am fairly certain that I have come up with an AGI which would implement findme/followme *without* nested SQL queries, or even stored procedures. With some more thought I am fairly confident I could do it without an AGI at all, although the dialplan functions to split up the destinations would be ugly as hell. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Crazy Boy wrote: Hi, Here I am posting my problem. I am getting this problem since 8 days. I have studied documentation and looked previous posts in forums. But, I am unable to solve this problem. Please show me a solution. I am from India. We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN-Eg. Mobile), I am not getting the callerid number of the caller and getting callerid as Asterisk in my softphones (XLite). SNIP When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console: Error Message: Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8) Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread: CallerID feed failed: Success Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1' Please tell me the solution. Looking forward to your kind response. Do you actually _HAVE_ caller ID on that PSTN line? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Douglas Garstang Sent: Thursday, August 17, 2006 2:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 'Hosting' -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: What's not specific about this...? handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster Again, it tells me you have not fully thought out exactly how each of those functions fit within the realm of Asterisk. I spent 8+ hours a day, 5+ days per week for over 6 months thinking how these functions fit within the realm of Asterisk. At every single turn, after going down every single path, there where limitations that forced us to backtrack and evaluate a different approach. A script that could handle call routing, in conjection with MySQL and stored procedures was the only way to implement our requirements. The MySQL command had limitations, realtime was way too resource intensive, unreliable and undocumented and so on. Yep... i definitely haven't thought about this at all. Oops. I almost forgot intra-organisational 4 digit extension dialling. Not just company, but organisational, where a company may have multiple organisational units. It might be possible to hack together a flat intra-business 4 digit extension dial lookup in the native dialplan, but trying to make it a multi-level organisation lookup would be pure hell... unless you farm the task out to a more advanced scripting langauge like python, perl whatever. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] astbill white screen!!
Sounds like a sessions error -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Milioto Sent: Thursday, August 17, 2006 2:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] astbill white screen!! Hi all, I've installed asterisk and astbill according with all recommendation (mysql5, drupal included with astbill, php, apache2...). When I write http://server_adress/astbill, I get a white screen page. Browser doesn´t give me an error page, it just a white screen page. However asterisk doesnt have any problem, and works well with mysql. I also have installed Drupal 4.7.3 linked to other database with other user and password working well. And I have phpMyAdmin too. All working very good at the same server. I tried changing index.php to phpinfo.php in the same directory and it works well too. Can anybody help me with that please? Any suggestion will be very appreciated. Thanks, very much in advance Sebastian On 7/14/06, varun [EMAIL PROTECTED] wrote: Hello, Our asterisk server is on Centos 4.2 We want to use Astbill. Astbill requires Drupal and mysql 5. I could not find rpms mysql5 for centos. We are getting mysql extensions issues because of php-mysql. How do we solve this ? Any other billing software that similar to Astbill ? Thanks Varun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 2:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 'Hosting' On Thursday 17 August 2006 16:04, Douglas Garstang wrote: Yet again you have not addressed my statements that showed that the MySQL Dial plan command was not capable of nesting SQL queries, and therefore not capable of implementing findme/followme. Off the top of my head I am fairly certain that I have come up with an AGI which would implement findme/followme *without* nested SQL queries, or even stored procedures. Yes, that's what we have done. I assume your referring to a small, function specific agi script. We have it as one large script instead. Obviously if you make a dozen of more agi calls like that during the course of processing a single call, it's going to get slow, and the caller may notice the effect. We didn't/don't have the resources available to be writing a multi-threaded fast agi server right now to fix that. Yep... as Jeremy said, I sure haven't thought about this at all. Nope... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Douglas Garstang Sent: Thursday, August 17, 2006 2:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 'Hosting' -Original Message- From: Douglas Garstang Sent: Thursday, August 17, 2006 2:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 'Hosting' -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: What's not specific about this...? handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster Again, it tells me you have not fully thought out exactly how each of those functions fit within the realm of Asterisk. I spent 8+ hours a day, 5+ days per week for over 6 months thinking how these functions fit within the realm of Asterisk. At every single turn, after going down every single path, there where limitations that forced us to backtrack and evaluate a different approach. A script that could handle call routing, in conjection with MySQL and stored procedures was the only way to implement our requirements. The MySQL command had limitations, realtime was way too resource intensive, unreliable and undocumented and so on. Yep... i definitely haven't thought about this at all. Oops. I almost forgot intra-organisational 4 digit extension dialling. Not just company, but organisational, where a company may have multiple organisational units. It might be possible to hack together a flat intra-business 4 digit extension dial lookup in the native dialplan, but trying to make it a multi-level organisation lookup would be pure hell... unless you farm the task out to a more advanced scripting langauge like python, perl whatever. I see the MySQL dial plan command still doesn't support stored procedures either, unless you hack around with the source. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, August 17, 2006 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk 'Hosting' -Original Message- From: Douglas Garstang Sent: Thursday, August 17, 2006 2:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 'Hosting' **snip** I spent 8+ hours a day, 5+ days per week for over 6 months thinking how these functions fit within the realm of Asterisk. At every single turn, after going down every single path, there where limitations that forced us to backtrack and evaluate a different approach. A script that could handle call routing, in conjection with MySQL and stored procedures was the only way to implement our requirements. The MySQL command had limitations, realtime was way too resource intensive, unreliable and undocumented and so on. Yep... i definitely haven't thought about this at all. Oops. I almost forgot intra-organisational 4 digit extension dialling. Not just company, but organisational, where a company may have multiple organisational units. It might be possible to hack together a flat intra-business 4 digit extension dial lookup in the native dialplan, but trying to make it a multi-level organisation lookup would be pure hell... unless you farm the task out to a more advanced scripting langauge like python, perl whatever. I see the MySQL dial plan command still doesn't support stored procedures either, unless you hack around with the source. I've just recently come up against this limitation. Care to share info/code concerning making stored procs work with the addon? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Rushowr [mailto:[EMAIL PROTECTED] Sent: Thursday, August 17, 2006 2:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 'Hosting' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, August 17, 2006 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk 'Hosting' -Original Message- From: Douglas Garstang Sent: Thursday, August 17, 2006 2:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 'Hosting' **snip** I spent 8+ hours a day, 5+ days per week for over 6 months thinking how these functions fit within the realm of Asterisk. At every single turn, after going down every single path, there where limitations that forced us to backtrack and evaluate a different approach. A script that could handle call routing, in conjection with MySQL and stored procedures was the only way to implement our requirements. The MySQL command had limitations, realtime was way too resource intensive, unreliable and undocumented and so on. Yep... i definitely haven't thought about this at all. Oops. I almost forgot intra-organisational 4 digit extension dialling. Not just company, but organisational, where a company may have multiple organisational units. It might be possible to hack together a flat intra-business 4 digit extension dial lookup in the native dialplan, but trying to make it a multi-level organisation lookup would be pure hell... unless you farm the task out to a more advanced scripting langauge like python, perl whatever. I see the MySQL dial plan command still doesn't support stored procedures either, unless you hack around with the source. I've just recently come up against this limitation. Care to share info/code concerning making stored procs work with the addon? Hi. I only just stumled across it myself. I was trying to prove a point to Jeremy. On the voip wiki: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MYSQL under a comment titled 'Calling MySQL 5 stored procedures from app_mysql', it looks like someone has managed to modify the source to get it to work. I haven't tried it yet... Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Turn Off chan_sip Debug Messages
Afternoon everyone, A while back I had a Sangoma tech poking around in one of my systems. He turned on a higher level of debugging, and I can't seem to figure out how to turn it off. I'm seeing messages like the following: Aug 17 15:19:49 DEBUG[2692]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' Aug 17 15:19:52 DEBUG[2692]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' Aug 17 15:19:57 DEBUG[2692]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Aug 17 15:20:09 DEBUG[2692]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' How do I turn off chan_sip debugging? I thought "sip no debug" or "no debug channel sip" would do it, but that doesn't seem to have stopped the logging. Can someone point me in the right direction? Cheers,Alex-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Return data from Fast AGI
Ok, maybe I'm having a brain fart, or maybe I've never gotten quite this far, but, if you call a fast AGI script, how do you RETURN data from the fast AGI back to the dialplan??? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and T1 Extensions.
Hi folks, Just a small question, i always use a X101 Card where in my office with asterisk, works great etc. Now I receive a T1/E1 30 Lines and buy a Digium Card for it. Very well, how my extensions change ?? Did i still use ZAP/01, ZAP/02 etc for the 30 lines ?? How can i know what zap is used for line ? So If i hold make a direct link like make a extension. ie: exten = 345 equal to phone - to receive and make call how i do this ? Any help please. Point a document will help alot too ;) Thanks ! Carlos ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
