[Asterisk-Users] red alarm on wildcard
Alarms Span RED wildcard X101P Board1 OK wcusb/0 0 ive got my asterisk server up and running and working correctly, the first time after a reinstall and reboot everything was fine - i had both alarms OK and i could get the USB extension ringing when i ran the house number from my mobile. as soon as i tried again i got a red alarm on the wildcard board. now im using the sample configuration files for the devkit lite (downloaded from the website) and i know that the loadzone settings will be different here (serbia) how do i find out what settings i should do for lines here in serbia. the telco here is worse than most ive used, and arent exactly helpful. my local exchange is very old and hasnt been upgraded for a very long time, services such as call waiting are not offered, but even so - everytime im on the internet and someone tries to call me i get disconnected. so what could be causing the red alarm and how (and where) do i change the loadzone settings? thanks in advance duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] red alarm on wildcard
Red Alarm on an X100P means that it does not detect a phone line connected to the device. I would double check your cabling. The loadzone only has to do with the sound of dialtone, etc, and i wouldn't mess with it until you have the basic functionality working. ok now i when i plug the phoneline into the "phone" socket of the X100P there are no alarms, but when i dial the phone line its connected to i get "number busy" (from my phone) and when i reboot the machine it kernel panics (scroll lock and caps lock flashing). i havent installed anything else, except a default "everything" redhat 7.2 and webmin. so i decided to unload all the modules[0] before i reboot - and i get this message "rmmod: module wcfxo is not loaded" and then rebooting doesnt cause a kernel panic. after i do this and reboot the machine i get a red alarm on the X100P card again, even thought the line is still connected to the "phone" socket of the card. any ideas? is there any way i can get these modules to autounload before a reboot, the kernel panic problem was the reason i did a machine rebuild in the first place (didnt think through the problem hard enough that time). duncan [0] /sbin/rmmod audio /sbin/rmmod wcfxo /sbin/rmmod wcusb /sbin/rmmod zaptel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
REPOST: [Asterisk-Users] red alarm on wildcard
im posting this again because i didnt get a reply to it earlier and i think its a fairly major problem. Red Alarm on an X100P means that it does not detect a phone line connected to the device. I would double check your cabling. The loadzone only has to do with the sound of dialtone, etc, and i wouldn't mess with it until you have the basic functionality working. ok now i when i plug the phoneline into the "phone" socket of the X100P there are no alarms, but when i dial the phone line its connected to i get "number busy" (from my phone) and when i reboot the machine it kernel panics (scroll lock and caps lock flashing). i havent installed anything else, except a default "everything" redhat 7.2 and webmin. so i decided to unload all the modules[0] before i reboot - and i get this message "rmmod: module wcfxo is not loaded" and then rebooting doesnt cause a kernel panic. after i do this and reboot the machine i get a red alarm on the X100P card again, even thought the line is still connected to the "phone" socket of the card. any ideas? is there any way i can get these modules to autounload before a reboot, the kernel panic problem was the reason i did a machine rebuild in the first place (didnt think through the problem hard enough that time). duncan [0] /sbin/rmmod audio /sbin/rmmod wcfxo /sbin/rmmod wcusb /sbin/rmmod zaptel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?
Hi,I want to get the latest asterisk code from CVS. But the computer OS I used for travelling internet is Windows. I don't know how to I deal with the CVS. Thanks. i use wincvs available from http://www.wincvs.org/ this gives me a copy of the source, but i have had problems copying that version onto a linux machine to install. the make install specifically hooks into CVS to get the version information. so make sure you have a .version file in your linux install directory the contents of the .version file should be something like: CVS-01/08/03-17:27:36 and windows will not let you create a file called ".version" - to do this i had to install the unixutils from http://unxutils.sourceforge.net/ and used "touch" to initially create the filename. good luck duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 100 vs SNOM 200??
what is WMI ? > WMI didn´t work with SNOM200. i think WMI refers to Windows Management Instrumentation http://msdn.microsoft.com/library/default.asp?url=/library/en-us/dnclinic/html/scripting06112002.asp some hardware uses it for configuration (others use things like SNMP or COM). dont take my word on it though. just the only thing i could think of thats the most likely abbreviation. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] low-cost * (newbie question)
gnophone on your * server will require X on your * server, and that would not be recomended. On my 1ghz AMD chip the screensaver could cause a severe degradation on my VoIP channels. Put gnophone on your other asterisk boxes and call each other. agreed, this is good advice, but you can run asterisk on a machine also running X without _too_ much of a performance hit, just make sure you have a blank screensaver set - rather than one that activally involves processing to reder itself. likewise its a bad idea to install seti or other programs like it on a server. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk vs. system user accounts
ODBC == BAD ODBC is the lowest common denominator for database connections. In this case a full blown database is overkill for lookups mainly. You don't need the whole SQL stuff. This is why LDAP would be okay, it essentially is a flat file that can be searched fast and remotely. i think ODBC support would be very useful, exactly because it is the lowest common denominator. fairly recently i had to work with an IVR company who wouldnt insert values into a database hosted at my location because their application did not support ODBC - instead they offered to FTP individual text files to an FTP server where I had to process them and insert them into a database myself. needless to say that company isnt getting any business from me anytime in the near future. ODBC is widely used and supported because its simple to use/support and very widely supported. ok technically its not a very elegant solution - but as a business solution it is very powerful. just my 2 cents... duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI accountcode.
I've setup cdr_mysql and am using AGI to authenticate users based on the
called-from # (callerid), use the AGI perl module. Looking at the info
stored in the caller detail, I see a field called "accountcode", is it
possible for me to set this field in AGI? I'd like to tie it to a
username, that I pull during my SQL authentication, so I can search the
cdr table based on a username, rather then a source phone number.
using the (excellent) AGI perl modules you would do this:
$AGI->exec('SetAccount', $username);
hope this helps
duncan
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Re: [Asterisk-Users] limiting out going calls to a maximum duration
I want to limit my sons phone useage, by setting a 30min limit on out going calls from his room is there a simple way of doing this with asterisk? id make him have a specific context, and set absolute timeout in it as 30 minutes (1800 seconds) [son-outbound] exten => _X.,1,AbsoluteTimeout,1800 exten => _X.,2,Dial,however you are dialing out... i think this should end the call after 1800 seconds... duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip and pix
does anyone have a sip working through a cisco pix firewall? i can get the phone to register and the call to be negotiated, but as soon as the call is answered there is no sound and the call ends immediately. im sure this is due to the RTP negotiation being rejected by the pix. any helpful ways around this? right now my only solution is to put a small box outside the firewall and IAX the connection to the main asterisk server, but its not a long term solution and id rather not have anything outside the firewall if i can help it. thanks in advance duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Starting Development Perl or Python
>From the drift of the mailing list most people seem to use perl for their AGI scripts. I personally have more experience with Python. Could you please advise why or if Perl is more suitable? Is it faster? Better supported? More documentation? i use perl because its a language im more familiar with, there are lots of CPAN modules available to aid with the type of applications i make and most of all its more supported thanks to the wonderful asterisk-perl module that makes agi development a lot easier. asterisk-perl can be found here: http://asterisk.gnuinter.net/ and theres a specific mailing list setup (available on the above url) to help answer any asterisk-perl specific problems. so yes, it is better supported, and there are more examples in perl than with any other language. speed im not so sure about, and nothing has a ton of documentation yet. personally id suggest you stick with what you know initially - if you prefer writing in python then thats probably the best language for you to develop in. good luck duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Core dumps from Asterisk
cvs update Got 3 core dumps from asterisk in a very short period of time, not sure what was going on, but I am posting these in case anybody wants to see them, if you want to see more info I can provide it on request. I am using the asterisk 0.5.0 download. he mentioned he was using the asterisk 0.5.0 download though. surely this means we should update the 0.5.0 release to solve these problems? can you confirm that it was the asterisk-0.5.0.tar.gz file install that caused these segfaults? duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Core dumps from Asterisk
> he mentioned he was using the asterisk 0.5.0 download though. surely this > means we should update the 0.5.0 release to solve these problems? > > can you confirm that it was the asterisk-0.5.0.tar.gz file install that > caused these segfaults? It was, I just downloaded the tar file, I haven't touched CVS. I assume the tar file is intended to be the latest stable version, unlike a CVS checkout which would give me a development version? yes thats what i thought was the idea as well. looks like there should be an asterisk-0.5.1.tar.gz release sometime soon then. anyone know of a "good" checkout day with not much missing or broken? duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] eBay Sip Phone Scam.
> Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as > the 102D. And to make matters worse he starts the bid at $90.00 Beware. There's no need to beware -- anyone who doesn't shop around deserves to get suckered. right, but this isnt about shopping around. its more of a warning that someone is selling grandstream 101's as 102D's. i think its a good thing that people in this community are looking out for each other, admittedly this would be the ideal candidate for a seperate list, but thanks for the warning. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR Questions?
I failed to mention in my original post that I've looked at perl through
AGI, but haven't yet found a function that allows me to capture digits to
a variable that I can then manipulate. I probably should also mention
that I'm not a programmer-type, although I can usually muddle through
simple scripts for smaller uses like this one. Can you suggest which
function I would use?
using the asterisk-perl module from http://asterisk.gnuinter.net/
you would use the get_data function:
my $captured_dtmfs =
$AGI->get_data('/var/lib/asterisk/sounds/whatever','5','4');
from the "show agi get data" help
show agi get data
Usage: GET DATA [timeout] [max digits]
Stream the given file, and recieve DTMF data. Returns the digits
recieved from the channel at the other end.
hope this helps
duncan
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[Asterisk-Users] iax2 trunk
Im having problems setting up a trunk between two locations. Heres the
setup I have:
Server A is connected to the PSTN at my datacenter
Server B is connected to a clients e1 line at his datacenter
I only want to route calls from Server B to Server A and out through the
PSTN. Server A has a lot of other things connecting to it, so I need a
very specific context for all calls to go through. Because of the volume
of calls between the two servers I wish to setup a trunk.
Server A has this entry in iax.conf
[serverb]
type=friend
host=serverbipaddress
trunk=yes
auth=md5,plaintext,rsa
secret=s3rv3rb
username=serverb
context=serverb
qualify=yes
Server B doesnt have much in iax.conf - only codec and port information
under [general]
Server B is using this in his extensions.conf though:
exten => _X.,1,Dial,IAX2/serverb:[EMAIL PROTECTED]/${EXTEN}|180
now i know some things are wrong, i know i can use type=peer because its
only a one way connection (im not making calls to serverb, only recieving
calls from it)
but when i do an iax2 trunk debug i get this:
IAX2 Trunk Debug Requested
Beginning trunk processing
Processed trunk peer 'serverb' (0.0.0.0:0) with 0 call(s)
Ending trunk processing with 1 peers and 0 calls processed
even though there are calls going between the servers - so obviously they
arent using the trunking facility. so whats the deal. what do i have to
do in iax.conf on both sides and in extensions.conf on the side of ServerB
thanks in advance
duncan
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Re: [Asterisk-Users] iax2 trunk
try trunking=yes i already have trunk=yes (which is correct according to the version of the handbook i have) and it doesnt seem to be using it duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunk
i already have trunk=yes (which is correct according to the version of the handbook i have) and it doesnt seem to be using it You have to have a matching type=peer on each end to use IAX2 Trunking? How about getting it working without trunking first? it is working without trunking - calls are going from serverb to servera. now if i have matching entries on both servers the calls are rejected, and im not sure why. if i have matching entries what information do i put in each one. does someone have an example of this i can see, everytime i try and set it up it fails? duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunk
Look at the iax.conf.sample in the Asterisk source directory. Maybe the Handbook is wrong. If so, report it to bugs.digium.com. from my iax.conf.sample ; ;[biggateway] ;type=peer ;host=192.168.0.1 ;secret=myscret ;trunk=yes ; Use IAX2 trunking with this host ; trunk=yes, not trunking=yes can anyone verify this? maybe its an update that has changed over time. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunk
Im having problems setting up a trunk between two locations. Heres the
setup I have:
Server A is connected to the PSTN at my datacenter
Server B is connected to a clients e1 line at his datacenter
I only want to route calls from Server B to Server A and out through the
PSTN. Server A has a lot of other things connecting to it, so I need a
very specific context for all calls to go through. Because of the volume
of calls between the two servers I wish to setup a trunk.
Server A has this entry in iax.conf
[serverb]
type=friend
host=serverbipaddress
trunk=yes
auth=md5,plaintext,rsa
secret=s3rv3rb
username=serverb
context=serverb
qualify=yes
Server B doesnt have much in iax.conf - only codec and port information
under [general]
Server B is using this in his extensions.conf though:
exten => _X.,1,Dial,IAX2/serverb:[EMAIL PROTECTED]/${EXTEN}|180
now i know some things are wrong, i know i can use type=peer because its
only a one way connection (im not making calls to serverb, only recieving
calls from it)
but when i do an iax2 trunk debug i get this:
IAX2 Trunk Debug Requested
Beginning trunk processing
Processed trunk peer 'serverb' (0.0.0.0:0) with 0 call(s)
Ending trunk processing with 1 peers and 0 calls processed
even though there are calls going between the servers - so obviously they
arent using the trunking facility. so whats the deal. what do i have to
do in iax.conf on both sides and in extensions.conf on the side of ServerB
ok, bad form to reply to my own posting, but in case anyone else has this
problem in future it was a very silly setting. on server b i didnt need
the entry in iax.conf - all i needed was a register statement.
duncan
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[Asterisk-Users] concurrent calls
So whats the best way to find the maximum number of concurrent calls in this setup: IAX2 Trunk using GSM over a 512k internet line. thanks duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] concurrent calls
> So whats the best way to find the maximum number of concurrent calls in > this setup: > > IAX2 Trunk using GSM over a 512k internet line. show channels? actually i meant how to find out how many i could push down the 512k line - with regards to codec bandwidth and signalling etc... duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one-way audio
Im experiencing a problem with a current setup and I've run out of ways to debug it and come to a resolution. I have two E100P's in a machine which is routing traffic over the internet to a machine that has 1 E400P connected to the PSTN. Clients are able to make calls successfully but when the call is connected experience one-way audio. They cannot hear anything said by the person who is being called. This appears to be intermittent, with occasional times where conversations can be held, but what sounds like silence suppression on the side of the caller. They hear audible clicks and then silence, and assume the person they have called has hung up. On the side of the person being called there are no audible problems at all (except for the fact that most of the time there is nothing being heard on their side). so the link between the two machines is using a 512k connection at one end, and a 2meg connection at the other. an IAX2 trunk between them. the only thing i can think of as a potential cause of this problem is that one is using a cvs thats more recent than the other. im worried about updating the older machine without being sure that its not going to introduce problems as there are many other applications and services running on that server. any help would be much appreciated duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one-way audio
Im experiencing a problem with a current setup and I've run out of ways to debug it and come to a resolution. I have two E100P's in a machine which is routing traffic over the internet to a machine that has 1 E400P connected to the PSTN. Clients are able to make calls successfully but when the call is connected experience one-way audio. They cannot hear anything said by the person who is being called. This appears to be intermittent, with occasional times where conversations can be held, but what sounds like silence suppression on the side of the caller. They hear audible clicks and then silence, and assume the person they have called has hung up. On the side of the person being called there are no audible problems at all (except for the fact that most of the time there is nothing being heard on their side). so the link between the two machines is using a 512k connection at one end, and a 2meg connection at the other. an IAX2 trunk between them. the only thing i can think of as a potential cause of this problem is that one is using a cvs thats more recent than the other. im worried about updating the older machine without being sure that its not going to introduce problems as there are many other applications and services running on that server. ooh replying to my own posts again. bad duncan. some more information on the problems being encountered. if i have an IAX2 Trunk - the person making the call cant be heard but the person recieving the call can hear them if i have a standard call over IAX2 the person recieving the call cant hear anything but can be heard by the person making the call if i have a standard IAX call - both sides can hear each other fine. any ideas where the problem lies based on this new information? duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No sound with SIP Phones on the Internet
This is bull... I can't believe that... Must be a solution... sip is very tricky to get working behind firewalls. sip clients work quite well with nat, just make sure nat=yes is in the sip profile in sip.conf my solution has always been to put an asterisk box behind the firewall and make all the sip clients connect to that, then IAX out of the firewall to the other machines. i spent a few days trying unsuccessfully to find a decent sip proxy that worked the way i wanted and decided that the asterisk solution was much better. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No sound with SIP Phones on the Internet
Dunca: I am not sure I understand your statemnet. SIP devices (UA) on the other side of the Internet behnid a NAT communicate to * on the public Internet. Then this Asterisk connects to other Asterisks (via IAX) that can be behind Firwalls (or NATS). am I understanding correctly? this is possible yes. or wherever you have a firewall - make sure theres an asterisk server behind each one. then open up IAX or IAX2 through your firewall and let the SIP clients connect "locally" to the asterisk server. remember you can run asterisk on very little processing power and it doesnt need any specific hardware to just do VoIP stuff. my setups are something like this: branch office sip phones -> asterisk server -> firewall -> internet datacenter internet -> firewall -> asterisk server -> pstn so i dont have any problems with trying to get SIP working through firewalls - i just put a local asterisk server wherever theres going to be sip phones. because IAX is so flexible with regard to firewalls it doesnt matter if the main asterisk server is on the public internet or behind a firewall with the IAX/2 ports opened up. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] qualify=yes (new subject)
I've added info on qualify=yes and how this can help the NAT dilemma to the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+qualify As Wipeout stated, this is extremely useful and an important part of the "undocumented Asterisk features" we're trying to document on the Wiki ;-) I can't find a setting for how often Asterisk sends the Qualify/Options request, but vaguely remember there being one. Any help? qualify=yes i think means the request gets sent every 60 seconds. you can of course make qualify=600 to make it send every 10 minutes. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] latest cvs update
ok, ive just updated a server and now im getting these messages a lot on the console: -- Called g1/X WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to forward voice WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to forward voice WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to forward voice WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to forward voice any ideas what the problem is? i havent change the configurations or setup of anything just upgraded from the cvs duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax over wireless
ok, ive just got a wireless internet connection installed at my home. its pretty much the only way to get a stable connection here in belgrade - so im happy with it. now the next trick is to get asterisk working bearably with it. unfortuantely its using a modified version of the 802.11b spec - called turbocell. it appears that to solve some of the problems with 802.11b in an outdoor environment and with multiple clients connecting over various distances, they have aggregated packets. it looks like packets are grouped together into 16 or 32k sizes before being sent. i have a local asterisk server here and i can establish an IAX connection to a remote asterisk server - it just sounds incredibly crackly, i suspect because of this packet aggregation going on. can anyone recommend any solutions to try and get the quality a bit better. its actually bearable for me, but people complain about the crackling. would having a jitterbuffer help out? what would be everyones recommended settings? thanks in advance duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
Is 'top' suggesting that * is actually consuming 98%? If it is, take a look at the * logs for signs of what it might be. We've seen this happen on a lab RH9 system, but its usually while we been doing other unusual things. (In our case, two extra instances of mpg consuming the ~98%; copying *.conf files to a second system that didn't actually have any x100p cards in it, etc.) FWIW, I'm running yesterday's cvs on two RH9 systems just fine. i had a problem with asterisk consuming all the resources available on redhat 9. it would occur roughly every 24 hours or so - and would cause all sorts of problems. when a new channel opened up it fought for resources for a few seconds - so no speech could be heard, then when it could grab enough resources to process the channel it would... but the quality would be terrible. it can be solved with this: export LD_ASSUME_KERNEL=2.4.1 so now thats in my asterisk init script before actually starting asterisk. since doing this i havent had a problem (3 weeks ago). duncna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Question
> >I recall in the archives somewhere, and through someone's post earlier > >today, that there is some sort of problem with RH9 with Zaptel (hardware) > >drivers and that RH8 is preferred. > > Do you recall what kind of problem? The only problem I have is an annoying > echo that I haven't yet gotten rid of. the only problems ive had with redhat 9 is the new thread model. it can be solved using: export LD_ASSUME_KERNEL=2.4.2 (i think) before you start asterisk, i do this in the asterisk init script so i dont forget. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
> Finaly, someone has started the IAX soft phone ball :) > > Thanks, Dan... actually theres been an opensource multiplatform iax soft phone on sourceforge for a while now: http://iaxclient.sourceforge.net/ duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Crashed Asterisk
> Stop using RH9 since its majorly broken and that wont happen Since not all of us are so enlightened, please complete the sentence so that at least some of us have a clue as to the non-partisan issues? No offense intended, but there really are folks here that would like to learn from other's experience however boring that might be. my experience with redhat 9 resulted in this nugget of information to solve the problems i was having. asterisk threads sometimes cause the processor to max out all resources - and when a new thread is initiated it takes a while to fight back enough processing power. theres a bit of a config that needs to be done to make sure it doesnt happen. in /etc/init.d/asterisk (installed by running "make config") make sure: case "$1" in start) # Start daemons. echo -n "Starting asterisk: " daemon safe_asterisk should be case "$1" in start) # Start daemons. export LD_ASSUME_KERNEL=2.4.1 echo -n "Starting asterisk: " daemon safe_asterisk this makes redhat 9 use the old thread model rather than the new one, and stops asterisk from getting all messed up. this is the only change ive had to make to stop asterisk behaving differently from my 7.3 and 8 machines. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 128 kbs satelite link
Senad Jordanovic said: > Hi all, > > Anyone has experience using * through > 128 kbs (or bigger) satelite link? > > In particular I am interested to hear how many calls could be put > through 128Kbs satelite link simultaneously? i havent tried a satellite link with * yet, but i do have it running on a 128k wifi link. due to the technology used for the wireless link (turbocell) the smaller packets are grouped together into a "superpacket" - this means the lower bandwidth codecs dont work very well. i found that using a higher quality codec actually gave better results. so i use ulaw all the time now - resulting in well, one concurrent call. so this really has nothing to do with answering your question, but maybe it might help someone else. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] offtopic: possible asterisk meeting saturday amsterdam
hello everyone, theres a bi-monthly computer fair in amsterdam on saturday and it looks like a few asterisk users will be attending, and hopefully some more might be able to turn up. admittedly this probably is a bad idea to advertise because the more asterisk people the less likely i am to find cheap AVM Fritz ISDN cards - but what the hell. if you feel like a chat, and want to avoid the family on saturday. myself, fuzzycat and a few others will be at the RAI from around 10am till 1pm. http://www.pcdumpdag.nl/ email me if you think you might make it - and i'll give you my mobile number so we can try and arrange where to meet. sorry for the completely offtopic message. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P on Redhat 9
Does anyone have any ideas what might be causing this, and how to diagnose without taking our main phone switch down? I have access to a PRI, but it has no phone numbers that ring into it, so I can connect to real network equipment and make calls, but cannot receive any. it might be something different, but previously ive had lots of problems with redhats new thread model and asterisk. it was causing processes to hog resources and generally made my asterisk servers incredibly unstable and often unusable. the problem was solved by running this before starting asterisk: export LD_ASSUME_KERNEL=2.4.1 i now have this in /etc/init.d/asterisk before asterisk starts - so its always run before asterisk starts. it may not be the cause of your problems, but even so i found it was useful and so i hope this helps duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deal with error messages passed as Early Media
On 04/02/16 06:00, Scott Griepentrog wrote: For calls that fail, even where early media is played, the call should terminate with a 4xx or 5xx SIP response which to a certain degree correlates to the nature of the actual failure. The SIP error code is delayed until the media playback completes, but should be no different whether or not early media is used (for the same actual failure). Early media is simply an audio stream for human consumption to explain the failure. There should be no need to attempt to recognize it, unless your ITSP is not terminating the call correctly. I recently ran some analysis of early media messages found in cellular networks by recording the calls and running it through CMU-Sphinx. There are only a few types of early media messages per network, to cover a raft of failures. But I always got a SIP error code back as well, the early media I found tended to play for upto 20 secs then drop the call, then you get the error code. It might not be a very descriptive error but its still an error code, and the early media audio message is not always very distinct either. In the GSM networks the GSM failure code is more useful but still seems somewhat randomly assigned by the provider, even including the odd temporary failures. You are not really worried about what the failure reason is, its the caller who needs to decide - did I misdial, is the number really disconnected, is their phone out of coverage etc, you just need to try your next available network, if the caller hasn't already hung up after hearing the message You could possibly examine the audio before you get an answer but then you might get caught by some other system or PBX playing early media before answer that isn't actually a failure. If your ITSP is not giving you an error code then you have an issue. Cheers Duncan On Wed, Feb 3, 2016 at 8:41 AM, Olivier <mailto:oza.4...@gmail.com>> wrote: Hello, I'm trunking with an ITSP that, when treating an outbound to an unknown destination, either: - send a SIP error code (I can't be more explicit, at the moment), - or cast a pre-recorded audio message using Early Media. At the same time, I'm also trunking with Contact Center solution which doesn't support Early Media. Beside asking my ITSP to treat calls consistently or ask Contact Centerto support Early Media, is there a way to configure Asterisk to unify both above error treaments into a single one ? How can I best deal with error messages passed as Early Media. Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Digium logo Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice recognition IVR Is it possible?
CMU Sphinx is really good if you know what sentences you want to recognise I am not sure how well it works with random stuff but if you have a list of common phrasings then you can do really well (having used it recently) - although I would say its much better at recognising North American speech than NZ. I think there are different language options too. Cheers Duncan On 23/02/16 08:56, John Kiniston wrote: I think I saw an old page on the voip-info wiki on how to use CMU Sphinx with asterisk. http://www.voip-info.org/wiki/view/Sphinx IMHO It's not going to be anywhere as good as a commercial product without a lot of work. On Mon, Feb 22, 2016 at 11:34 AM, Daniel Chavez <mailto:topdog2...@gmail.com>> wrote: Thanks for the link. Are there no free alternatives for speech recognition? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rasterisk freeze on 4G link
> With this new link, whenever I launch a vim, a nano or a rasterisk session, my terminal freezes (rasterisk) or remains empty (nano, vim). > > When a session is frozon, I can open a new one at the same so it excludes a basic connectivity loss. > Usually incorrect MTU gives you this effect. Use ping with MTU size set to test and find what works. I think its value: it's 1272, which amazes me. You will probably also break it with any large text dump eg cat /var/log/syslog will also do it Yes "cat /var/log/syslog" also broke my console. Why would my console break because of inadequate MTU and other PC on the same location, seem unaffected ? Because, they most probably mostly use SMTP and HTTP ? Is possible to simulate a given MTU on a LAN to reproduce such freezes ? (the remote location is at the other side of the country and I would like to prepare things as much as possible). I think you need to go through a router or some device that can constrain the MTU. But live changing your server MTU should be straight forward as openvpn should try and reconnect, and you can change the server back. I haven't lost connectivity before with this Also the session is probably timed out rather than gone, in 10-15 mins maybe less it will come back (or does for me) Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lost outgoing SIP packets
One issue that can catch you is a packet MTU limit in your path to your SIP box lower than your standard MTU. You can check that with ping -s 1500 option Cheers Duncan On 01/04/16 17:12, Pete Mundy wrote: Roel, Just another thought bouncing around... Your ifconfig output was specific to eth1. Is there an eth0 too? Is there a chance packets are heading to that other interface when they shouldn't be? Running a second tcpdump on eth0 at the same time should at least disprove the theory quickly. Pete On 1/04/2016, at 2:59 am, Roel van Meer <mailto:r...@1afa.com>> wrote: Thanks for the heads up, and thanks for thinking with me everyone! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best timing source?
On 07/04/16 09:00, Carlos Chavez wrote: On 4/6/16 2:39 PM, Duncan Turnbull wrote: I am starting to think that the problem may be in the server hardware itself. We are using a Dell R220 with 8gb of ram and 2 hard disks in a Raid 1 configuration (Linux Raid). We are using CentOS 7. We had to remove the raid card from the server to install an E1 card (the raid card was Windows only so no loss there really). Internally everything sounds good (from E1 to a conference or music) but once you hit a network interface we start getting pops and drops. Anyone with this server and Asterisk ever had issues like these? Just checking you have your E1 timing set to slave off the upstream. If not you are going to have E1 sync errors which will give you the voice problems you describe Dahdi_test gives me a 99.97% average. The problem is present on all calls (except calling into the E1 to a conference or to MoH). I am preparing a new server to see if it is a hardware issue. This is the bit I mean, but if you have calls going over the E1 that are okay then its probably not this. http://www.voip-info.org/wiki/view/Asterisk+PRI |# span=,,(LBO)>,,[,yellow] # # All T1/E1 spans generate a clock signal on their transmit side. The # parameter determines whether the clock signal from the far # end of the T1/E1 is used as the master source of clock timing. If it is, our # own clock will synchronise to it. T1/E1's connected directly or indirectly to # a PSTN provider (telco) should generally be the first choice to sync to. The # PSTN will never be a slave to you. You must be a slave to it. # # Chose 1 to make the equipment at the far end of the E1/T1 link the preferred # source of the master clock. Chose 2 to make it the second choice for the master # clock, if the first choice port fails (the far end dies, a cable breaks, or # whatever). Chose 3 to make a port the third choice, and so on. If you have, say, # 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each # port should be different. # # If you choose 0, the port will never be used as a source of timing. This is # appropriate when you know the far end should always be a slave to you. If the # port is connected to a channel bank, for example, you should always be its # master. Any number of ports can be marked as 0. # # Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed # faxes, unreliable modem operation, and is a general all round bad thing. | -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble getting Asterisk Running with FreePBX 11
On 22/04/16 01:52, Daniel Chavez wrote: Hi, I recently had to reinstall Asterisk and FreePBX. asteirsk 11.20 and FreePBX 12. This is running on Centos 6.7 32 bit. When I use amportal start It comes up with the errors below Error in argument 1, char 2: option not found r /usr/local/sbin/amportal: line 49: Usage:: command not found WARNING: ERROR IN CONFIGURATION My guess is you might not have run amportal with enough permissions to allow it to create /var/run/asterisk If you create the directory and give asterisk permissions you should get past this but since it can be wiped its better to make sure your permissions are right. astrundir in '/etc/asterisk' is set to but the directory does not exists. Attempting to create it with: 'mkdir -p ' mkdir: missing operand Try `mkdir --help' for more information. ERROR: COULD NOT CREATE Attempt to execute 'mkdir -p ' failed with an exit code of 1 You must create this directory and the try again. I'm not sure how to solve this? Asterisk already has the run directory in /var/run/asterisk and the config file notes this. Perhaps it's PHP? Before I ran PHP 5.4 and now am running PHP 5.5. Good luck -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to decrypt encrypted SIP user's secret
On 29/06/16 16:37, Nathan Anderson wrote: You must mean that engineer before you used "md5secret" instead of "secret" for each user in sip.conf? If so, why can't you just copy the md5secret line from the old server to the new server for each user? -- Nathan *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ikka Tirtawidjaja *Sent:* Tuesday, June 28, 2016 6:20 PM *To:* asterisk-users *Subject:* [asterisk-users] how to decrypt encrypted SIP user's secret Dear all, My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP User's secret. But the voip engineer before me didn't save / documented those password. Now the server's hardware is begin to broke, it hangs a lot, and have a lot of call problem. We already have a new asterisk PBX to replace it, but we have difficulty to retrieve the encrypted password. Are you talking about sip registrations, as above they can be found in the sip config files, assuming your PBX is still at least working and you can look at the config files. about a hundred of our customer use an old IPPhone that doesn't have a reset button to hard reset the admin password (back to factory default). The previous engineer also change the IPPhone's admin password without any documentation. So, we can not move / change those IPPhone to the new PBX. Is there a way for us to retrieve / decrypt those SIP secret ? Or anyone has any experience how to reset this IP Phone (Dayou Ddip-100) Most phones don't have a reset button, rather a keypress combination to reset them. I can't read the manual unfortunately, can you get hold of the manufacturers? Any suggestion are appreciate, because I'm really desperate. You can figure out the SIP password from the files, and if you shift the IP address to the new server you should be fine. Thanks in advance, Regards, Ikka Jakarta - Indonesia -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iptables for SIP talk to other port
Don't you want udp rather than tcp? Have a look at the iptables stats to see if any packets are hitting your rule. Also I think the source port from your host will be 5068 so your replies will be to the right port but you can double check tcpdump is also very useful here sudo tcpdump -i eth0 -n udp and host 192.168.1.3 should show you packets between your machine and your odd host Cheers Duncan On 17/10/16 11:55, Mike wrote: I'm by no means an iptables guru... Not sure if it's necessary to enable forwarding via: echo "1" > /proc/sys/net/ipv4/ip_forward Also have you tried without the "POSTROUTING" rule? I seem to recall that "iptables" is smart enough to correctly route packets back out without that rule. On Sat, 15 Oct 2016, Jerry Geis wrote: I have a host 192.168.1.3 that wants to run SIP on 5068 (long story).My host is 192.168.10.201. My host needs to stay on 5060 because of all the other devices I have connected. I tried putting port=5068 in my SIP extension definition but that did not work. So I thought about using iptables to accomplish this: iptables -t nat -A PREROUTING -p tcp --dport 5068 -j REDIRECT --to-port 5060 iptables -t nat -A POSTROUTING -p tcp --dport 5060 -d 192.168.1.3 -j REDIRECT --to-port 5068 Do I not have the right format of the command? Anything incoming destined for 5068 redirect to 5060... Anything going out to 192.168.1.3 and port 5060 redirect to 5068. Seems like that should have worked? Thoughts? sip show peers still says unreachable. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuration management and update deployment - what do you use?
Hi All We have about 15 different asterisk boxes around the place and on my list has been automate deployment updates and keep a revision history. They are mostly not publicly accessible, and external SIP access is closely firewalled , so updates happen straight away when its something like heartbleed, but take a while to trust/test new releases. Our boxes are Ubuntu LTS - mostly 14.04 at the moment. We use Freebpx as the configuration front end and so that tends to be a more manual update, although there is an API we could use to keep things in step. We run backups from freepbx and archive those as well as any specific asterisk settings missed. At the moment our scale means manual is okay, but automation would make it easier if the learning curve and new issues aren't too high. We compile asterisk from source as the packages aren't usually quite what we want. I was just curious how people deploy asterisk across multiple platforms and keep them all up to date? What tools are good for this sort of thing? Thanks very much Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuration management and update deployment - what do you use?
On 19/10/16 05:57, Ludovic Gasc wrote: +10 for Ansible. We use that on our production. Okay, I will investigate Ansible Thanks very much Cheers Duncan -- Ludovic Gasc (GMLudo) http://www.gmludo.eu/ 2016-10-18 14:00 GMT+02:00 marek cervenka <mailto:cerva...@gmail.com>>: ansible.com <http://ansible.com> Dne 18/10/2016 v 11:46 Duncan napsal(a): Hi All We have about 15 different asterisk boxes around the place and on my list has been automate deployment updates and keep a revision history. They are mostly not publicly accessible, and external SIP access is closely firewalled , so updates happen straight away when its something like heartbleed, but take a while to trust/test new releases. Our boxes are Ubuntu LTS - mostly 14.04 at the moment. We use Freebpx as the configuration front end and so that tends to be a more manual update, although there is an API we could use to keep things in step. We run backups from freepbx and archive those as well as any specific asterisk settings missed. At the moment our scale means manual is okay, but automation would make it easier if the learning curve and new issues aren't too high. We compile asterisk from source as the packages aren't usually quite what we want. I was just curious how people deploy asterisk across multiple platforms and keep them all up to date? What tools are good for this sort of thing? Thanks very much Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference <http://www.asterisk.org/community/astricon-user-conference> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue command to force SIP client to re-register
On 22/11/16 11:33, Telium Technical Support wrote: Is there a way to force a SIP client to re-register using a SIP command (or an AMI command)? SIP Notify can usually force a lot of phones to reset. The notify type depends on the phone, but it copes with most major brands Try it out Cheers Duncan If not, is there some other standard way to do so – or would I have to post/get to a web GUI of the phone (unique to each model) to force a reset, etc. -Raj- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Touch tone stutter
On 23/11/16 13:49, Pete Mundy wrote:
One direction that may be worth exploring further is his ATA's config (or
perhaps swapping it for a different model). Eg adjusting echo cancellation or
line impedance settings.
Is the ATA he is using the same as the ATA you use?
Failure to correctly recognise and decode DTMF is just one of many reasons why
I never use them (ATAs). Like faxing over VoIP, they're just too much trouble :(
Genuine IP phones are pretty good value these days. Could you drop one of those
on-site as a temporary measure to prove that it's phone and/or ATA related?
Pete
Ps, you might also want to consider joining VoiceOps (if you're not already
subscribed) and posting there. https://puck.nether.net/mailman/listinfo/voiceops
On 23/11/2016, at 12:16 pm, D'Arcy Cain wrote:
I am hoping someone else has seen this and can offer a solution or at least a direction to investigate. I am
running 11.23. Most of my clients are fine but one has a strange behaviour. He has a Grandstream HT701 like
most of my clients who use an ATA. He can make call and they are crystal clear. However, when he tries to
use phone menus ("dial 234 for John Doe" for example) it doesn't work. At first I thought that the
tones were not being delivered but I had him play them to me and the issue is that each tone stutters. As a
result, entering "234" becomes "223344" which is not understood as a valid input.
What are the problems with DTMF and VoIP?
http://www.voipmechanic.com/dtmf-issues.htm
In some VoIP routes a switch may be configured to detect in-band DTMF
which is sent by the VoIP ATA, but then switches to an out of band
RFC2833 DTMF required for an upstream provider. This upstream carrier
then terminates the call to the PSTN, possibly to a voice mail system,
which will require regeneration of the audible inband DTMF tones. The
switch has to detect and remove the tone sent by the ATA from the audio
stream because the upstream provider specified RFC2833 DTMF. At times
the switch can't always completely remove the in-band DTMF tone which is
a problem, because by the time it has detected the DTMF tone, it has
already passed a short amount of it. This small amount of in-band tone
along with the RFC2833 tone sent are both received by the far end voice
mail system which will then register an error (problem), possibly an
invalid mailbox or invalid password.
If this is happening You can set asterisk to use receive the tones
inband which if this is occurring might help
Trial and error probably, good luck
He has a recent phone and, in fact, is almost the same model I have at home.
His is a Panasonix TX-TGD220 and mine is a TX-TGD-212. The difference is
mainly that his has a built in answering machine.
As I said, no one else is having the issue. One person has a horrible
connection with voice drops all the time but the touch tones still work.
I have made two files available. http://darcy.vex.net/Bishop.ogg is an OGG file of the
sound that it makes at the receiving end and the other at http://darcy.vex.net/Bishop.png
is a picture of the wave form. I had the user think "one Mississippi" etc. and
alternately press and release three different buttons. I recorded off my SIP phone which
is going through the same Asterisk server as the user.
The only thing I can see in my configs that might apply is in sip.conf "dtmfmode=rfc2833"
which I don't want to change unless I am absolutely sure. No one else is having the problem so I
don't want to risk it. Would I be safe if I set it to "auto"? Is there any chance that
it would help? Is there some place else I should be looking?
Thanks in advance for any help.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
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Re: [asterisk-users] Various extensions ring once and go to voicemail
On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters wrote:
We have an old Asterisk 1.8.7.0 system desperately need to keep alive
for another 6 months or so. We had all kinds of hardware problems, so
we virtualized it.
Thats a while back, I think it tended to use zaptel or dahdi hardware
as a timer, you may need to find a timing source as perhaps the clock
in the VM is just going too fast
Now, random extensions ring once and go straight to voicemail.
I went to one of the affected extensions and changed the ring time
from the default (20) to 26. Still one ring. I changed it to 30. Now
I get two rings. Other extensions ring once or twice. After some
time has gone by since this was first reported, all phones in my
random sample ring only twice.
As I trace a call to that extension through the log, I notice it
setting the ring timer properly (I think) and then I see
app_dial.c – SIP/1234- is ringing
Then eventually
app_dial.c: -- Nobody picked up in 3 ms
But according to the timestamps, the time from the one event to the
other is ten seconds!
Here’s another example- call starts:
[2019-01-14 08:17:33] VERBOSE[13311] pbx.c: -- Executing
[3327@cc-long-distance:1] ExecIf("SIP/4704-1265",
"0?Set(__RINGTIMER=0)") in new stack
. . .
[2019-01-14 08:17:33] VERBOSE[13311] app_dial.c: --
SIP/3327-1266 is ringing
. . .
[2019-01-14 08:17:41] VERBOSE[13311] app_dial.c: -- Nobody picked
up in 2 ms
So again, the elapsed time is nowhere near 20 seconds.
Another: This time the ring time has been set to 30 seconds (and I
still get only 2 rings)
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing
[3327@cc-long-distance:1] ExecIf("SIP/4704-1304",
"1?Set(__RINGTIMER=30)") in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: --
Executing [s@macro-exten-vm:5] Set("SIP/4704-1304", "RT=30") in
new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: --
Executing [s@macro-dial-one:30] Set("SIP/4704-1304",
"D_OPTIONS=trWw") in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] app_dial.c:
-- SIP/3327-1305 is ringing
. . .
[2019-01-14 08:42:05] VERBOSE[16008] app_dial.c:
-- Nobody picked up in 3 ms
So, after 9 seconds, it says “Nobody picked up after 3 ms”???
Is this some weirdness of Oracle VMs? Anybody have any advice for me?
Additional information:
FreePBX version 2.9.0.7
PBX in a Flash Version 1.2 Daemon Status
* Asterisk * ONLINE * Dahdi * ONLINE * MySQL * ONLINE *
* SSH * ONLINE * Apache* ONLINE * Iptables * OFFLINE *
* Fail2ban * OFFLINE * IP Connect* ONLINE * Ip6tables * OFFLINE *
* BlueTooth * ONLINE * Hidd * ONLINE * NTPD * ONLINE *
* Sendmail * ONLINE * Samba * OFFLINE * Webmin * LOADING *
* Ethernet0 * ONLINE * Ethernet1 * ONLINE * Wlan0 * N/A *
* Running Asterisk Version : Asterisk 1.8.7.0
* Asterisk Source Version : 1.8.7.0
* Dahdi Source Version : 2.5.0.1+2.5.0.1
* Libpri Source Version: 1.4.12
* Addons Source Version: 1.4.7
Voipserver on 10.10.141.251 - eth0
Red Hat Enterprise Linux Server release 4.5 (Tikanga) :32 Bit Kernel:
2.6.18-92.1.6.el5
Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org
Milwaukee County Transit System
1942 N 17th Street | Milwaukee, WI 53205
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[asterisk-users] Asterisk and SIP Proxy on same host = media problem
Hi there I have a pbx (v16.10) on AWS (Ubuntu 18.04) with Freepbx (14) that I am trying to set up the proxy reSIProcate on the same host as pbx. I can make it all work when the proxy is on a different host but when the proxy is on the same host asterisk sends the media address as 127.0.0.1 which the end user then happily sends media to 127.0.0.1 but it doesn’t get anywhere. Asterisk then disconnects after 30 seconds for lack of RTP activity |==AWS==| ( Asterisk <=> Proxy )<=> Local Firewall <=> End user Because Asterisk and Proxy are on the same host then the interface addresses are either localhost, local AWS IP, or external AWS IP. But when using either Chan_sip or PJSIP the media address asterisk presents is always 127.0.0.1 and not the externip I am looking for any advice on whether this is possible, recommended or completely foolish and if it is possible what I should look for to make it go. Asterisk is on internal ports while the proxy is on 5060. The SIPs peer in chan_sip and pjsip are using context from-internal. The call sets up but then fails once media flows. This works well if the proxy is on another host. I also realise reSIProcate is old but its relatively straight forward compared to Kamailio (for me), although I eventually plan to figure out Kamailio when I have more time. I don’t know whether I could affect this differently with Kamailio. Thanks very much Cheers Duncan p.s. apologies if this is sent twice -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best practice security for internet access to Asterisk
Hi All For the scenario of a single asterisk server that needs to serve clients on the net, as well as local office clients, I would be very interested in people's views of the best method to handle security to prevent net based attacks while still allowing the client access. Some of the challenges I see are: - preventing brute force and bot type attacks - monitoring for unusual events and notifying and acting appropriately - limiting damage if someone does get in - avoiding a Denial or degradation of service on your asterisk platform - making it easy for staff to use Some of this can be done with - firewall control - but its hard to limit where your clients will come from, besides restricting ports - scripts monitoring logs, I saw a recipe for checking password failures then blocking that ip after x failures, I imagine this could get quite sophisticated - using separate restrictions for offnet users but this kind of makes it harder for the staff members. - using a proxy in front of asterisk for SIP, to limit the available extensions and minimise the scanning impact on the asterisk box. I am hoping this could detect and prevent illegitimate or poorly formed requests or unknown user agents. Staff should be using a standard set. - using iax softclients to shift the attack requirements - I don't know much about how well these work - running all clients over a vpn e.g open vpn, but this is not so good for wireless handsets or other devices that can't do a vpn I am interested in all views and recommendations Thanks very much Cheers Duncan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco to Asterisk migration
Hi Femi We have about 50 Cisco 7960s on one site off Asterisk 1.4.18 Its all SIP and it doesn't stress a P3 system much at all. I am not sure what phones you are using - the 7960s are not hard to configure, a bit of process to convert from the Cisco Skinny to SIP (using SIP v8.6) but everything seems to work well. The 7961s or 7971s use an XML config which is probably Everything loads off the TFTP server. We are using the Linksys POE Switches SFE2000P which seem okay but don't always like to be fully loaded Things I would work on are automating or simplifying the provisioning (doesn't change that much once its done), firmware upgrades, and getting to know the config files well. Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Femi Sent: Friday, 25 April 2008 21:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Cisco to Asterisk migration Hi Guys, I have client with a Cisco 2690 call manager solution that wants to upgrade but cannot stomach the costs of continuing with Cisco The installation will go up to 100 users The client currently has about 40 Cisco phones and would like to continue with these phones with the odd Polycom I'm looking at plugging in an Asterisk box and using the existing Cisco box as a PSTN gateway only Has anyone on the list done this? Any pitfalls or tips you would like to share? Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio after call transfer
I had a similar issue in 1.2 after transfer and we were using SIP only but an upgrade cured it We are now on 1.4.18 still without issues Cheers Duncan Rilawich Ango wrote: >Hi all, > > Recently, I experienced one way audio after call transfer. > >incalling call (PSTN) A --> GXP2000 thro' zap --blind transfer--> destination >B >Finally A and B reach each others, but there is only one way audio. >Anyone get the same experience before? How to solve the problem? > >Asterisk vesion: >Asterisk 1.4.15 >zaptel 1.4.7 >asteriks-addon 1.4.5 > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell PowerEdge 860, Sangoma A108
Hi Helen Sounds good, I think Troy will need me to setup the notification list to the winners though so it might pay to send me those details directly Should be better rugby this weekend for one of us ;-) Cheers Duncan on 10/09/07 14:20 Paul Hales said the following: > We have used a quite a few dell 860's in our installs with Digium cards > (Te120's) without any issues. > > PaulH > > > On Mon, 2007-10-08 at 15:28 -0700, Girts Graudins wrote: > >> Hello everyone, >> >> I'm considering getting me a quad-core Dell PowerEdge 860 to run >> Asterisk. Anyone else using this model? Any tales of woe and sorrow >> I should know about? >> >> Then, in a couple of weeks, I'm thinking of getting a Sangoma A108 and >> giving that a try. Same question with that one - any quirks I should >> be aware of? >> >> >> Girts >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7960 Queue Issue
The freepbx system has a primary number option in its ring group dialing which if selected as a ring strategy means it won't ring any further if the primary number is engaged. This is useful in follow me setups. I haven't dug into how its implemented but it works for ring groups and follow me on freepbx (asterisk 1.2 and 1.4) An article on the concepts. http://freepbx.org/2007/06/03/ring-group-and-follow-me-ring-strategies-1-of-2 It maybe useful to help figure out a way around your issue. Cheers Duncan on 11/05/07 14:09 Nick Brown said the following: > Thanks Eric, this is the case. A bit of a shame that it removes the > functionality for the member to see calls that have not come from a queue > however there is not much choice in the matter. > > FWIW to get this option a firmware upgrade was required (Now running > POS3-08-8-00). > > Cheers. > > > On 5/11/07 11:57 AM, "Eric Merkel" <[EMAIL PROTECTED]> wrote: > > >> On 11/4/07, Nick Brown <[EMAIL PROTECTED]> wrote: >> >>> Morning All, >>> >>> Quick question that has me stumped. Have a queue with several members >>> (Statically defined in queues.conf at this stage for testing) who use Cisco >>> 7960's. >>> >>> The queue is configured to use rrmemory and generally this works correctly. >>> However if a member is already on a call their phone will still ring (The >>> 7960 can show multiple incoming calls for one line). I really don't want >>> members who are on calls to get more calls. Especially when we start logging >>> out members who don't answer. >>> >>> Asterisk shows; >>> -- Called 1014 >>> -- SIP/1014-08f2e4d0 is ringing >>> -- Local/[EMAIL PROTECTED];1 is ringing >>> -- Nobody picked up in 15000 ms >>> >>> Short of disabling the feature to show multiple incoming calls on the 7960's >>> (Which I don't know if it can be done anyway), has anyone got any >>> suggestions? >>> >>> >> Yes, you can turn off this in the phone. Go into call preferences on >> the phone and turn off call waiting. Not optimal but can be done. >> >> -Eric >> >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
We build and maintain 7 Asterisk boxes for our customers, I have recently moved 3 to 1.4. I also use iaxmodem and on the last one 1.4.14 I was getting iax thread errors - which was reported as a bug in much earlier versions but apparently fixed. When 1.4.15 came out (two days later) it solved this problem, for me at least. I didn't dig any further but it did moderate my confidence somewhat. We run everything on ubuntu server 6.06 LTS and also use freepbx as the interface with some minor customisations. It works very well and we are now shifting some others to 1.4 but the issue is if anything goes wrong its too costly to fix, as part of maintenance we keep them uptodate. The main blocker for 1.4 was freepbx but now it supports 1.4 and seems to manage the transition really well. However being a small self employed group of two the main reason to stick with what works is the risk of cost. We don't generally do major upgrades without charging but there isn't any seriously missing functionality yet, and the effort involved to be sure it will be hassle free is significant. The clients have to see value in the upgrade. We also work with people still on version 1.0, because the risk of change to a working system is too high This seems to be the same issue already mentioned but my take on it is most people can't cope with any risk on production machines unless there is some significant gain. Its been a year now, generally I would think that means its probably starting to become stable but a year isn't very long really. Give it another year and the new installs will mostly be 1.4 and the migration process will be a lot more trusted. I don't think a year is really long enough to expect much more than where you are at. The debian stable, unstable, and testing model would be useful here, debian stable is so reliable it just rocks, if there was a version like that it would be fantastic (of course you trade access to the latest features for it) . We find ubuntu server a great balance between debian stability and getting the latest options. Is there a performance analysis of 1.2 vs 1.4 around or a clear business analysis of the distinctions in value for each? Cheers Duncan Lyle Giese wrote: > Olle E Johansson wrote: > >>>All I can say is with 1.6, if a change is made that causes something >>>that worked in 1.4 not to work in 1.6, please think twice, three >>>times or four times before making the change, or making the change >>>in such a way that it won't break dialplan stuff from 1.4. >>> >>> >>> >>Our policy is to never remove any functionality between two versions. >>We replace the functionality with new functionality and print out >>warnings whenever you use the deprecated functions. We also add this >>to the documenation in the software and the UPGRADE.TXT file. So the >>functionality that you lost in 1.4 was old 1.0 functions that was >>marked as deprecated in 1.2 and removed in 1.4. >> >>We might want to be more informative about those changes. We need to >>make a clear list of things you need to start changing as a user of >>1.4 to prepare for lost functionality in 1.6. This information already >>exist, but should maybe be a bit more public. >> >>In some cases we do have to change in a dramatic way and can't >>preserve the old functionality to solve a bug in the software. This >>requires thorough discussion in the developer group and is something >>we really want to avoid at all costs. If this happens, it's clearly >>documented in the software. >> >>Thank you for your feedback, it's important to us. >> >>/O >> >> >> > Along that this same line, I ran 1.0.something for a long time and it > was working just fine for my SOHO. I had a channel bank to interface > pots lines from the local Telco and feed the analog phones in the > house. Over time, I replaced most of those analog phones with SIP phones. > > An unfortunate incident caused us to lose that server and several sip > phones. When I recovered enough to rebuild *, I tried 1.4 and it > would not compile completely and zaptel did not load properly. I > download 1.2 and it worked with the same configs as 1.0, but the > quality was poor. That was due to hardware issues. > > I purchased a new motherboard and rebuilt using a newer Asterisk 1.4 > with the then current libpri and zaptel and the call quality came > back. But I had a hard time with syntax changes. Basically I was > jumping from 1.0.x to 1.4.x in one leap. > > My biggest gripe is that everything loaded and seemed to work. A day > later we found this did not work and discovered a syntax change. A > da
RE: [asterisk-users] Delays on E1 Delivered via SHDSL
I doubt it's the PRI itself SHDSL isn't part of the internet per se, its just an access technology. SHDSL is just synchronous DSL which can be used to deliver E1s over. ISDN PRI's are delivered in a 2Mbit/sec G703/G704 frame and will give you lots of alarms if they are having any issues It could be your toll provider at the end of it is routing calls in ways that cause delays, but less likely to be the PRI Cheers duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Sent: Thursday, 31 May 2007 12:18 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Delays on E1 Delivered via SHDSL I have an Asterisk system with a TE110P installed and connected to an ISDN E1 PRI that is delivered via a 2mb SHDSL connection. I am experiencing delays (the type of delay you would get on an international call) during calls. I am wondering if anyone could advise, would the problem be with any part of the Asterisk system or is the problem with the fact that the ISDN is delivered over the internet? ___ --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] SugarCRM Integration
If you look through the Trixbox without Tears by Ben Sharif - google for it, it's a good read for things you can do for asterisk Ch 31 has this below I would search the tribox and sugar forums for more info - really its just using click to dial from sugar, and potentially CID lookup - I am not sure if it is using Cheers Duncan SugarCRM is a contact management software that comes bundled with Trixbox. To set up SugarCRM, first, you need to open the SugarCRM application http:///crm using the default username of Admin and the password of password. For security reason you should change the Admin password. To do this, click on 'My Account' in the upper right-hand corner, then click on the 'Change Password' button underneath 'Users: Administrator (Admin) in the center-left of the screen. Change it to a new password and confirm your new password and click 'Save.' Now it's time to set up your contacts. I will start off setting up a couple of my internal extensions. Click on 'My Account' again and then click the 'Edit' button. Change 'Asterisk Phone Extension' to your Asterisk extension. My extension is 2001. While you are at it, change your time zone and date format as well. Click 'Save' to save that information. Let's add another one. Click on the contacts tab and then select 'Create Contact' from the left hand Shortcuts menu. Add another extension, in my case I chose my daughter's extension 2002: Firstname: Norsurya Last name: Sharif Home: 2002 Click 'Save' to save that information. Add another and another if you want to, using the method above. At this point, you may find that you are unable to make a phone call through SugarCRM. This is due to a little bug in the popup_picker.php (this bug may have been fixed by the time you read this, but at the time of writing, this bug exists). To fix this bug, you need to edit popup_picker.php by doing the following: >From your Linux CLI, log in as root. cd /var/www/html/crm/modules/Contacts nano Popup_picker.php Browse down to line 121 and change it from: $number = preg_replace ( "/[^\d\*]/", "", $number ); To $number = preg_replace ( "/[^\d\*]/", "", $display_number ); TRIXBOX Without Tears Page 136 of 209 You should now be able to dial from SugarCRM to your other internal extensions and to the outside world. Note: You can add multiple users who will each have their own settings/contacts/etc. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Saturday, 2 June 2007 7:25 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SugarCRM Integration Yes. You are right. You can integrate Sugar with Trixbox very easily. You can customize it also. Thanks, Chandra Joseph Bajin <[EMAIL PROTECTED]> wrote: I'd like to know as well about this. On 6/1/07, Diego Quintana Cruz wrote: > Hi folks, > I was wondering if there's a guide on how to configure sugarCRM > Integration with Asterisk. I was looking in google and all i found was > about Trixbox, which has sugarcrm integrated by default. > > Regards, > -- > Diego Quintana a.k.a. RouterMaN > Ingeniero de las Telecomunicaciones > Linux Registered User #382615 - http://counter.li.org/ > SIP # 1-747-633-6676 Ext. 1011 > FWD # 764839 Ext. 1011 > http://routerman.blogsome.com > http://gst.telecom.pucp.edu.pe > ___ > --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 _ Park yourself in front of a world of choices in alternative vehicles. Visit <http://us.rd.yahoo.com/evt=48246/*http:/autos.yahoo.com/green_center/;_ylc=X3oDMTE5cDF2bXZzBF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsD Z3JlZW4tY2VudGVy> the Yahoo! Auto Green Center. ___ --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] Slow list
This message arrived today 18 July NZ time Full headers below but most of my mail is like this - the offending bit seems to be: INXS.digium.internal which took 4 days to deliver it Cheers Duncan Return-path: <[EMAIL PROTECTED]> Envelope-to: [EMAIL PROTECTED] Delivery-date: Wed, 18 Jul 2007 09:33:50 +1200 Received: from pluto.e-simple.co.nz ([203.163.98.172]) by quotient.onesquared.net with esmtp (Exim 4.50) id 1IAug2-0006xz-17 for [EMAIL PROTECTED]; Wed, 18 Jul 2007 09:33:50 +1200 Received: by pluto.e-simple.co.nz (Postfix, from userid 1014) id 7C20823927F; Wed, 18 Jul 2007 09:33:48 +1200 (NZST) X-Spam-Checker-Version: SpamAssassin 3.1.7-deb (2006-10-05) on pluto X-Spam-Level: X-Spam-Status: No, score=-2.2 required=5.0 tests=AWL,BAYES_00, FORGED_RCVD_HELO autolearn=ham version=3.1.7-deb Received: from lists.digium.com (lists.digium.com [216.207.245.17]) by pluto.e-simple.co.nz (Postfix) with ESMTP id 2CF0223928A for <[EMAIL PROTECTED]>; Wed, 18 Jul 2007 09:33:44 +1200 (NZST) Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from <[EMAIL PROTECTED]>) id 1I9oPd-0002fh-Rm; Sat, 14 Jul 2007 15:40:21 -0500 Received: from exprod8mx52.postini.com ([64.18.3.152] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from <[EMAIL PROTECTED]>) id 1I9oPU-0002fV-Ja for asterisk-users@lists.digium.com; Sat, 14 Jul 2007 15:40:12 -0500 Received: from source ([82.103.132.43]) by exprod8mx52.postini.com ([64.18.7.10]) with SMTP; Sat, 14 Jul 2007 13:40:11 PDT Received: from localhost (threed001.three-dimensional.net [127.0.0.1]) by three-dimensional.net (Postfix) with ESMTP id 628F651C50 for ; Sat, 14 Jul 2007 22:40:10 +0200 (CEST) X-Virus-Scanned: Debian amavisd-new at three-dimensional.net Received: from three-dimensional.net ([127.0.0.1]) by localhost (mail.three-dimensional.net [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id I8DqCNjuf0hk for ; Sat, 14 Jul 2007 22:40:04 +0200 (CEST) Received: from lunteren.vanbaak.info (vanbaak.xs4all.nl [82.95.250.75]) by three-dimensional.net (Postfix) with ESMTP id EF00D51C4F for ; Sat, 14 Jul 2007 22:40:03 +0200 (CEST) Received: by lunteren.vanbaak.info (Postfix, from userid 1000) id 5814A50BB; Sat, 14 Jul 2007 22:39:57 +0200 (CEST) Date: Sat, 14 Jul 2007 22:39:56 +0200 From: Michiel van Baak <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Mail-Followup-To: asterisk-users@lists.digium.com References: <[EMAIL PROTECTED]> MIME-Version: 1.0 Content-Disposition: inline In-Reply-To: <[EMAIL PROTECTED]> X-PGP-Key: http://michiel.vanbaak.info/Files/pubkey.asc User-Agent: Mutt/1.5.12-2006-07-14 X-pstn-neptune: 0/0/0.00/0 X-pstn-levels: (S:91.99845/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108 M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from <[EMAIL PROTECTED]> [db-null] Subject: Re: [asterisk-users] Slow list X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.9 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion List-Id: Asterisk Users Mailing List - Non-Commercial Discussion List-Unsubscribe: <http://lists.digium.com/mailman/listinfo/asterisk-users>, <mailto:[EMAIL PROTECTED]> List-Archive: <http://lists.digium.com/pipermail/asterisk-users> List-Post: <mailto:asterisk-users@lists.digium.com> List-Help: <mailto:[EMAIL PROTECTED]> List-Subscribe: <http://lists.digium.com/mailman/listinfo/asterisk-users>, <mailto:[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, 15 July 2007 8:40 a.m. To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Slow list On 16:28, Thu 05 Jul 07, Philipp Kempgen wrote: > Since the list was switched over to API-Digital almost > every message I get is older than a week. Coincidence? > Is anyone else having trouble? > > Regards, > Philipp I got this message today July 14 Yes, I have the same. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer afficionados are both called users?" ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/l
Re: [asterisk-users] Slow list
I thought initially it was a pretty poor generalization about postgrey and our capabilities until I realized that this was sent a few weeks ago when this probably wasn't an as obvious issue. But it clearly is an issue now. I have checked my mail servers for failures, implicitly greylisting is working as the mails are coming from digium constantly - just a long time delayed, if postgrey was an issue there would still be retries and there have been none in over a week - as long as my logs go back, any decent mail server should have retried in much less than a week. Anyway - a discussion and investigation of issues is made pretty hard with 4 -10 day gaps in it. Since every other list works on time (+- a few hours) its looking like Digium from my view. I imagine someone will have sorted it before I see my own post, fingers crossed. Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed Sent: Friday, 6 July 2007 6:33 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Slow list On Thu, Jul 05, 2007 at 01:40:50PM -0400, Doug Lytle said: > >>Well, this is now the third active thread on this subject, but I guess > >>you won't see this message for a while. Has anyone dissected the > >>headers of a delayed message yet? We should be able to tell for sure > >>where the holdup is. All of the messages are coming through on time > >>for me, so it won't do much good for me to look. > > > Looks like mail is getting held up between INXS.digium.internal and > lists.digium.com > > INXS.digium.internal received it the first of July, lists.digium.com > received it on the 4th. > > drdos.info (ME) received it from lists.digium.com on that same day (Today). What you can't see without looking at the mail server logs on both ends is delivery attempts. Greylisting for example can totally hose you over depending on the implementation. Greylisting without whitelisting is irresponsible. How many tries did the digium server make before the message finally got through??? That's what we need to know. Only Digium can say. Before poking Digium too much, I would look at exactly what YOUR mail servers are doing that may potentially be the real cause of the delays. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?
I am yet to use 2.3 but have 2.2 on 8 ubuntu based installations with Asterisk 1.2.18 or greater FreePbx is really useful as an interface to all the config files, stats etc, its also really great if your customers need some control The documentation has recently been updated and there is a lot of life in the project so I would recommend it Just note, that like everything you still need to put some time into understanding what you are doing and how to get around the systems Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Friday, 14 September 2007 8:57 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly? On Thu, Sep 13, 2007 at 04:32:27PM -0400, Jay R. Ashworth wrote: > I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6 > stations, mostly IP (I'm looking at the Grandstream 201, to start), and > maybe X-lite on a couple of laptops via VPN. > > We've got a 4xFXO box we bought off eBay, which unfortunately I can't > find to quote a model number off of, but I *think* it's a Grandstream > as well. > > I've looked at several of the packages that turn Asterisk from a PBX > construction kit into an *actual* PBX, and so far FreePBX looks like > the one that matches my mental model of a small phone system best. > > Anyone have any first hand experiences with it that they'd like to > share? And I inadvertantly thread-jacked someone. Sorry. Fixed. Cheers -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g & SIP
Hi Salvatore Do you have a TFTP server that serves the phone configuration files? This is very separate to the phone, i.e. on a server/pc somewhere, and will log all the file requests it receives. You can check this irrespective of the phone Have you checked whether tftp requests are being made, usually they come before the system goes into the upgrading state. I have had that before and it was caused by having different load files from that specified in the OS79XX.TXT file which for my phones usually have P003-08-6-00 but for upgrading I start from P0S30202 For SIPDefault.cnf you also need the image version to match #Image Version image_version:P0S3-08-6-00 ; But for conversion I first go to this image image_version:P0S30202 ; And I go from that to this image_version:P0S3-06-2-00 ; then to the current version And I have these files on my tftpserver which are the respective firmwares -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2 -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin I can't recall if I need all the 08-6 versions Cheers Duncan Sasa wrote: > Hi Duncan, > I have tried more times to make the reset phone but is displays always and > only 'upgrading' and MAC address and I cann't access the phone > configuration. > Thanks. > > -- > >Salvatore. > > > > - Original Message - > From: "Duncan Turnbull" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, October 14, 2008 11:41 AM > Subject: Re: [asterisk-users] Cisco 7906g & SIP > > >> Hi Salvatore >> >> You need to look at the logs of the tftp server, not the phone. >> Hopefully you can see the ip address of the phone asking for files >> >> If there is nothing at all being requested from the tftp server then you >> probably want to reset the phone to defaults again. >> >> Usually it stalls when you have some mismatches in the config files. But >> it almost always asks for the default files. >> >> From the files requested you can determine whether its asking for SIP >> or SCCP files, and if SIP which version of firmware for the phone >> >> Cheers Duncan >> >> Sasa wrote: >>> Hi Dave, >>> I don't view nothing in tftp server because the phone is stopped on start >>> screen with displayed 'upgrading' and MAC address..I don't understand >>> what >>> happened after the reset. phone >>> Regards. >>> >>> -- >>> >>>Salvatore. >>> >>> >>> >>> - Original Message - >>> From: "David Gibbons" <[EMAIL PROTECTED]> >>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >>> >>> Sent: Monday, October 13, 2008 4:29 PM >>> Subject: Re: [asterisk-users] Cisco 7906g & SIP >>> >>> >>>> Hi Salvatore, >>>> >>>> I'm talking about the tftp logs on the tftp server: >>>> >>>> Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' >>>> should do the trick. >>>> >>>> Dave >>>> >>>> -Original Message- >>>> From: [EMAIL PROTECTED] >>>> [mailto:[EMAIL PROTECTED] On Behalf Of Sasa >>>> Sent: Monday, October 13, 2008 9:57 AM >>>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>>> Subject: Re: [asterisk-users] Cisco 7906g & SIP >>>> >>>> I cann't view phone log files because, after reboot, the phone is >>>> stopped >>>> on >>>> this screen ( 'upgrading' with MAC address) ! >>>> Regards. >>>> >>>> -- >>>> >>>> Salvatore. >>>> >>>> >>>> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g & SIP
Hi Salvatore You need to look at the logs of the tftp server, not the phone. Hopefully you can see the ip address of the phone asking for files If there is nothing at all being requested from the tftp server then you probably want to reset the phone to defaults again. Usually it stalls when you have some mismatches in the config files. But it almost always asks for the default files. From the files requested you can determine whether its asking for SIP or SCCP files, and if SIP which version of firmware for the phone Cheers Duncan Sasa wrote: > Hi Dave, > I don't view nothing in tftp server because the phone is stopped on start > screen with displayed 'upgrading' and MAC address..I don't understand what > happened after the reset. phone > Regards. > > -- > >Salvatore. > > > > - Original Message - > From: "David Gibbons" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, October 13, 2008 4:29 PM > Subject: Re: [asterisk-users] Cisco 7906g & SIP > > >> Hi Salvatore, >> >> I'm talking about the tftp logs on the tftp server: >> >> Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' >> should do the trick. >> >> Dave >> >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Sasa >> Sent: Monday, October 13, 2008 9:57 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Cisco 7906g & SIP >> >> I cann't view phone log files because, after reboot, the phone is stopped >> on >> this screen ( 'upgrading' with MAC address) ! >> Regards. >> >> -- >> >> Salvatore. >> >> >> ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g & SIP
Hi Salvatore Have you checked the tftp logs in any event? Its important to check the tftp logs and see if anything is being requested. I have had this before but usually its still trying to grab its first couple of files, and from that you can get an idea of where its getting stuck. If it says upgrading it means its trying to change from one version to another and failing, so you need to go backwards to a version it can cope with. If its not asking for any files then usually what I have done is to go to the lowest SIP version 2 or 3 for changing from the call manager to SIP and reset the phone to factory defaults and try and get it to start the change again Cheers Duncan Sasa wrote: > Hi Duncan, > yes I have a tftp server (I use also Cisco 7941G that use tftp server for > upload configuration) and I know this function, but now my problem is that > the phone is stopped on the initial screen that show 'upgrading' and MAC > address and the process not continued. > Thanks. > > -- > >Salvatore. > > > > - Original Message - > From: "Duncan Turnbull" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, October 14, 2008 8:52 PM > Subject: Re: [asterisk-users] Cisco 7906g & SIP > > >> Hi Salvatore >> >> Do you have a TFTP server that serves the phone configuration files? >> This is very separate to the phone, i.e. on a server/pc somewhere, and >> will log all the file requests it receives. You can check this >> irrespective of the phone >> >> Have you checked whether tftp requests are being made, usually they come >> before the system goes into the upgrading state. >> >> I have had that before and it was caused by having different load files >> from that specified in the OS79XX.TXT file which for my phones usually >> have P003-08-6-00 but for upgrading I start from P0S30202 >> >> For SIPDefault.cnf you also need the image version to match >> #Image Version >> image_version:P0S3-08-6-00 ; >> >> But for conversion I first go to this image >> image_version:P0S30202 ; >> >> And I go from that to this >> >> image_version:P0S3-06-2-00 ; >> >> then to the current version >> >> >> And I have these files on my tftpserver which are the respective firmwares >> >> -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2 >> -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads >> -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn >> -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin >> -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn >> -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin >> -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin >> >> I can't recall if I need all the 08-6 versions >> >> Cheers Duncan >> >> >> Sasa wrote: >>> Hi Duncan, >>> I have tried more times to make the reset phone but is displays always >>> and >>> only 'upgrading' and MAC address and I cann't access the phone >>> configuration. >>> Thanks. >>> >>> -- >>> >>>Salvatore. >>> >>> >>> >>> - Original Message - >>> From: "Duncan Turnbull" <[EMAIL PROTECTED]> >>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >>> >>> Sent: Tuesday, October 14, 2008 11:41 AM >>> Subject: Re: [asterisk-users] Cisco 7906g & SIP >>> >>> >>>> Hi Salvatore >>>> >>>> You need to look at the logs of the tftp server, not the phone. >>>> Hopefully you can see the ip address of the phone asking for files >>>> >>>> If there is nothing at all being requested from the tftp server then you >>>> probably want to reset the phone to defaults again. >>>> >>>> Usually it stalls when you have some mismatches in the config files. But >>>> it almost always asks for the default files. >>>> >>>> From the files requested you can determine whether its asking for SIP >>>> or SCCP files, and if SIP which version of firmware for the phone >>>> >>>> Cheers Duncan >>>> >>>> Sasa wrote: >>>>> Hi Dave, >>>>> I don't view nothing in tftp server because the phone is stopped on >>>>> start >>>>> screen with displayed 'upgrad
Re: [asterisk-users] top posting again [was: Re: CDR Design]
I like the discussion, I doubt it will end. I prefer top posting because I reply to all my customers that way, my mail client isn't that smart and I think technology should meet the needs rather than force you to adopt work arounds. I can fully understand though others preferring it, but I don't. All the presented evidence so far suggest bottom posting is a work around to a list archive function that is less than ideal or a politeness to get around a way of doing things that doesn't really apply so much anymore. I would have thought someone could make a better list archive model, I don't believe bottom posting is intuitive and therefore being picked up by many newcomers to the game. An alternate is to get a filter that sorts the whole thing out depending on preferences ;-), but who can be bothered. I haven't seen a signup requirement to this list requiring bottom posting, and neither have I on the many other lists I am on. In fact if I look at most of my lists the majority of posters over time have tended to top posting. Doesn't mean its right but it appears to be happening. Cheers Duncan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts
Try some of the shell scripts in the asteriskcookbook recipe heap http://asteriskcookbook.com/wiki/index.php/RecipeHeap Specifically http://asteriskcookbook.com/wiki/index.php/Asterisk_Brute_Force_Prevention Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Tuesday, 1 July 2008 07:33 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip extension compromised,need help blocking brute force attempts iptables -A INPUT -p tcp -s 74.52.112.162 -j DROP Good luck. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of spectro Sent: June 30, 2008 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] sip extension compromised, need help blocking brute force attempts Hello, yesterday one of the extensions on my asterisk server got compromised by brute-force attack. The attacker used it to try pull an identity theft scam playing a recording from a bank "your account has been blocked due to unusual activity, please call this number..." Attacker managed to make lots of calls for around 8 hours before I detected it and changed the password for that extension. As of this morning it is still attempting to brute force the password for that extension again. I need a way to block that IP from connecting to my asterisk server, please advice. --- sip debug --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 74.52.112.162 : 5060 (NAT) Found user '211' Reliably Transmitting (NAT) to 74.52.112.162:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 74.52.112.162:5060;branch=z9hG4bK3b28fa36;received=74.52.112.162;rport=5060 From: "ASLPLS" ;tag=as130a4d39 To: ;tag=as0c69057b Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- sip debug --- That box is currently running Trixbox 1.2.3. I have iptables disabled. If anybody can give me a simple ruleset that allows all traffic except ip 74.52.112.162 to port 5060 I will really appreciate it. Are there mechanisms in Asterisk to detect and automatically block these brute force attempts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astrundir not used
Are you using ubuntu? Usually I have to edit the Makefile in the else section of Global variable declaration based on architecture # ASTVARRUNDIR=$(localstatedir)/run ASTVARRUNDIR=$(localstatedir)/run/asterisk This seems to do it Cheers Duncan on 07/09/08 04:53 Cyril SCETBON said the following: > hi, > > I'im using asterisk 4.1.21 and astrundir is configured as followed in > /etc/asterisk/asterisk.conf : > > [global] > astetcdir => /etc/asterisk > astmoddir => /usr/lib/asterisk/modules > astvarlibdir => /var/lib/asterisk > astagidir => /usr/share/asterisk/agi-bin > astspooldir => /var/spool/asterisk > astrundir => /var/run/asterisk > astlogdir => /var/log/asterisk > > when I start asterisk it creates his pid file and the ctl socket in > /var/run and not in /var/run/asterisk > > How can I fix it ? Is it a known issue ? I did not get this error with > asterisk 1.4.10 > > Thanks > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gui issue in asterisk aa50
I had an issue where I put a comma in the prepend digits string pn call plans and then the call plan menu would no longer load. It parses the menu from the text file so I used the file editor to clear the offending line and my menu came back. Not sure if thats your issue but I was surprised I could enter text that broke the menus Cheers Duncan On 16/07/2008, at 10:27 AM, "Sydney Web Hosting" <[EMAIL PROTECTED] > wrote: > HI all, > > I am having issues with the gui on my AA50. > > under Voice Menus > Add new Step > Go to Time based rule. > > It allows me to select “Go to Time based rule” from the menu but > no options come up when selected. > > I’ve tried all browsers but no luck. > > > > Thanks > David. > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDMoE with Telco
You can use TDMoE to get an E1 running but its really designed to replicate an E1 end to end Its a standard and there is equipment out there that does it, e.g. from RAD and a few others. I didn't have any joy using the Asterisk code to get it going but it should in theory work. Its completely different to Dundi The challenge it is a protocol and needs two boxes talking TDMoE at each end. Telco's do not have this as an option, or at least none do that I have found Cheers Duncan Michael Graves wrote: > --Original Message Text--- > *From:* Yacine Boukaba > *Date:* Sun, 3 Aug 2008 18:54:08 +0100 > > Hello, is it possible with TDMoE to replace classic digital T1/E1 > interfaces like digium and sangoma cards connected to a telco. Or > TDMoE is only possible for connecting two asterisk boxes using their > NIC interfaces. if TDMoE can work with an T1/E1 connected with telco > how we can get the remote mac address of the telco interface ? > ThanksNo virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.5.10/1586 - Release Date: > 8/1/2008 6:59 PM > > I thought that TDMoE was largely depricated in the wake of DUNDi? > > Michael > -- > Michael Graves > mgravesmstvp.com > http://blog.mgraves.org > o713-861-4005 > c713-201-1262 > sip:[EMAIL PROTECTED] > skype mjgraves > [EMAIL PROTECTED] > > > > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >AstriCon 2008 - September 22 - 25 Phoenix, Arizona >Register Now: http://www.astricon.net > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AA50 using multiple outbound routes
Hi All I have an AA50 without inbound DDIs but each line has a separate number so based on analogue port it can be routed to different people. The challenge with this method is it appears to only allow the dial plan to use 1 outbound route so if all the analogue ports are split into individual lines it can only use 1 line for outbound, it won't allow it to step to other ports. This seems a less than ideal design and it could be I am missing something. If anyone knows how to make the outbound calls on an Digium AA50 appliance step through all its available ports (or at least a selected subset of them) I would love to know. Thanks very much Cheers Duncan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF
Its not so hard if the APs are purely just converting ethernet to wireless. If there is any authing on the AP then it would be tougher. And a centralised DHCP issuer is important i.e. just one address range across all APs so when moving APs there is no dhcp change, no auth change, just a client reconnecting to the SSID. I guess this is not exactly what you are talking about but organising such protocols to work properly does become much more complex, and no we didn't have so much joy that way. We have a wireless ISP in Wellington, New Zealand called CafeNET and thats all the APs do i.e. wireless to ethernet for large zones of the city homed back to a central controller. I have walked along our main central city street, Lambton Quay and carried on a conversation moving between at least 3 APs. I usually walked about 200 - 350m depending my destination so could be using quite a few APs The asterisk box in question is not blocked by the ISP. Driving is different, but walking is okay, and the street noise masks the other occasional glitches, so I may think its doing better than it is, it can be noisy and hard to hear in any event with a mobile as well. I don't do it so much any more because the cellphone charges got lower and I got tired of two devices, especially one that ran out of batteries. But I did for a few months to prove the point you are asking about. Cheers Duncan Michael Graves wrote: >On Sat, 30 Aug 2008 11:51:49 -0500, Karl Fife wrote: > > > >>Has anyone ever really, truly, actually held on to a Wi-SIP call while >>moving from the range of one AP to the range of another AP in the same >>network? >> >>Let's say a 'YES' only counts if you had a bona-fide handoff. In other >>words, you began in place 'A' (within range of AP#1 but OUTSIDE the >>range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but >>completely outside the range of AP#1) WITYOUT dropping the call. >> >>Supposedly it's possible with compliant hardware using 802.11f - >>Inter-Access Point Protocol (IAPP), but given how ALL standards ALWAYS >>work together PERFECTLY, 100% of the time :-), I'm guessing that it >>doesn't work. Can anyone speak to this from experience? >> >>-Karl >> >> > >Karl, > >I'm guessing that it was not common. 802.11f handoffs reportedly take >100ms which is considered too long for streaming applications like >voice and video. > >The 802.11r standard was only agreed upon and released days ago. This >specifies FAST BSS transition specifically to saisfy such applications. >Not sure if any hardware supports this as yet. > >http://en.wikipedia.org/wiki/IEEE_802.11r > >Michael >-- >Michael Graves >mgravesmstvp.com >http://blog.mgraves.org >o713-861-4005 >c713-201-1262 >sip:[EMAIL PROTECTED] >skype mjgraves >fwd 54245 > > > > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >AstriCon 2008 - September 22 - 25 Phoenix, Arizona >Register Now: http://www.astricon.net > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verbosity best practice
Its a good question I have lots of disk space so leave it high, I would rather have the detail if I need it It probably would seem sensible to revisit stable systems after a year and lower the verbosity, but then since I can afford the space I am not too fussed. Cheers Duncan Olivier wrote: > Hello, > > When managing a stable system, which verbosity level do you adopt ? > Leaving a higher level helps to catch root cause, if for any reason, > things go wrong. > Leaving a lower level saves resources if you need (have) to backup logs. > > What are current best practices ? > Do you change verbosity level during system lifecycle ? > > Regards > > > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >AstriCon 2008 - September 22 - 25 Phoenix, Arizona >Register Now: http://www.astricon.net > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g & SIP
Are you sure you have set the 7960 to SIP? By default they use SCCP, so you need to go through the process of changing them over, which ideally would just be done with the edits you have already in the load files but generally means going back to an early version of the SIP code then working upwards from there. You can check the current hardware in the status, if its SIP it will be something like POS-0806... (I haven't got a phone handy to check) but there is a reasonable amount of info on voipinfo about the process Cheers Duncan Sasa wrote: > Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk > 1.2.26. > I have uploaded in my tftp server the firmware > 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in > SEPmacaddress.cnf.xml I have: > > SIP11.8-0-4SR1S > > ..but in tftp log server I have: > > Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving > CTLSEPmacaddress.tlv to 192.168.0.155:49152 > Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving > SEPmacaddress.cnf.xml to 192.168.0.155:49153 > > ..and in asterisk CLI I have: > > -- Starting Skinny session from 192.168.0.155 > Device SEPmacaddress is attempting to register > > Now when 7906G started is loaded: > > load file: sccp11.8-3-2s > boot load id: tnp06.3-0-1-31.bin > > ..why isn't loaded sip firmware ?? > Thanks in advance. > > -- > >Salvatore. > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best way to get 60+ analogue extensions.
Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to get 60+ analogue extensions.
Thanks very much Rob & Stephen The channel banks look good. I am not sure if they are easily availble in NZ but we can get some in I am sure. Xorom make very positive comments about their astribanks and that you can have multiple channel banks on a server so they look pretty good (if they are honest). I can't tell the manufacturer of the other channel banks you were referring to. In Wellington, NZ, PRAs are pretty expensive and a 25Mbit/sec symmetrical fibre connection to a SIP provider is a better deal. On some of my other customers we have 15 SIP lines without issue using G711 and consuming about 80-100k per line if that. But I take the point so will revisit it in the design. Another reason for SIP is the Telepermited options available are limited over here, so to connect you really want to have an approved device in case you have any issues. But with SIP via a Provider you abstract that layer which is cleaner. If we need to have one E1 then having more for the Astribanks sounds fine. Cheers Duncan Rob Hillis wrote: >Duncan Turnbull wrote: > > >>Hi All >> >>I am looking at a replacement for a hotel PBX which requires at least 60 >>analogue extensions. >> >>I tend to use Sangoma equipment but haven't tried this many analogue >>extensions before. I am interested in anyone's experience of which >>server platform literally fits and copes well with multiple cards, and >>the choice of Digium vs Sangoma or something else. >> >> >> > >You have several options here, however due to the power requirements, I >wouldn't recommend you use either the Sangoma or Digium analogue cards >here - providing ring voltage to that many extensions is likely to >over-tax the power supply in the server. > >I'd either be looking at three channel banks (3 24 channel channel banks >would give you a total of 72 analogue channels) or two Xorcom Astribanks >which would likewise give you up to 64 channels. > >The Astribanks are probably a cheaper way to go since they connect to >your server via USB rather than T1/E1 ports. However, I haven't had any >experience with multiple Astribanks connected to the same server, so >there may be issues there that I'm not aware of. Channel banks are >certainly the proven and reliable technology, but will be significantly >more expensive since they connect to your Asterisk server via T1/E1 links. > > > >>I may need an ISDN PRA inbound but am working hard to have the inbound >>lines via SIP, but if I do that means at least 4 slots on this plan. >> >> >> > >You'd need to be very sure of the bandwidth and quality of connection to >your VoIP provider to go with SIP for more than half a dozen channels. >This kind of connection can easily be far more expensive than a >traditional T1/E1 line, so I wouldn't be pushing so hard for SIP. > >If you were to use channel banks, you would most likely end up with a >four port T1/E1 card and would only be using three of those channels, >leaving a spare one for an incoming T1/E1 line. > >If you were to use Astribanks, you would have plenty of space in the >server to include a T1/E1 card. > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic Calling Feature?
Not too hard to do, you can have a script generate a list of call files which automatically ring the callers in the list and play a message http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Cheers Duncan Christopher Stamper wrote: > Right now, my organization is using a commercial service > (OneCallNow.com), that gives telephone notifications to all numbers in > a predefined list. Example: > > -Admin records a voice message > -Service calls each number in the list, and plays the message back to them > > It's a pretty handy service, albeit a bit pricey. I've been wondering > if Asterisk could do this for me? I don't really want to have to write > scripts, but it would be great if it's already a feature. > > I don't have an Asterisk PBX running yet, but when I do it will > probably have multiple T1 PRI lines, making it possible to dial all > these numbers (100+) in a reasonable amount of time. > > Anyone know of a way to do this? > > -- > Christopher Stamper > > Email: christopherstam...@gmail.com <mailto:christopherstam...@gmail.com> > Web: http://tinyurl.com/2ooncg > gTalk: http://tinyurl.com/6e359r > Skype: cdstamper > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and openvpn and sip
Usually this is a routing error with openvpn setup and asterisk thinking it needs to route someway other than the vpn. If the originating packets have an external ip address asterisk might send them back out another route Have a look using tcpdump on the server to see where the returned packets are destined Cheers Duncan Giorgio Incantalupo wrote: > Hi all, > > I'm trying to connect one phone to a remote asterisk server via openvpn. > First of all, I put the vpn server on the box hosting asterisk and the > vpn client on another box, both with public ips. > Then I set the client ip as my phone IP gateway and the remote pbx ip as > the registrar and outbound proxy. > > I see in the phone log register packets are sent but nothing in return. > Asterisk console shows it tries to give back the packets but they seem > to be lost somewhere. > > I made some tests with my pc setting its gateway with the vpn client IP > and I can reach the pbx machine (ping, ssh,...) but sipsak gets no response. > It seems ping and ssh response packets are correctly routed but sip > packets aren't. > > I tried to set nat=yes in sip.conf but without result. > Is there any asterisk parameter to set to make it work with openvpn? > > Any help really appreciated. > > Thank you. > > Giorgio > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM mobile trunks
Yip the VoiceBlue SIP units are very good but a bit pricey Gordon Henderson wrote: > On Tue, 23 Jun 2009, Sasa Bobek wrote: > > >> Hi all, >> We have been planing for a long time to set up GSM mobile trunks for >> termination, and were planing on going with analog GSM adapters connected to >> a VoIP gateway. Should we be concerned with such a set-up as far as voice >> quality and other issues are concerned? Any experiences with GSM terminal >> chipsets? >> > > Why not SIP based GSM devices? e.g. Portech? > > Gordon > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about monitored calls storing
Trixbox I think uses FreePBX FreePbx has an option for each extension to set it to record all calls. It will record the extension in the file name and you can view it through the recordings app if you want a web view. There are all stored in a common dir /var/spool/asterisk/monitor - you can probably mod the code for the recording if you want more info in the filename Cheers Duncan peace keeper wrote: > Hello all, > how can I possibly make the monitoring for all calls through the > asterisk, and for those file to be stored with the name of the > initiator, in additional to know to whom this call is going, could > this functionality be implemented via configurations! > > in other words, could I configure the asterisk so that the > administrator to be able to hear calls coming from who going to whom, > as a having a record for each call, > I am using trixbox v2.6.2.1 > > should that functionality be implemented by an external application , > such as one written using asterisk-java !!! > > any help is appreciated? > thanks in advance, > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to sniff RTP and SIP traffic only
For Linux use tcpdump on the host you are after tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0 where 5060 is your SIP port and 1-16000 are your rtp ranges -s0 means snap length of 0 so capture all the packet rather than cutting off at a point And refine it by adding the host you are targetting and -w to write to a file. Then you can import the file in wireshark and use the voip utlities to listen to it fairly easily or use tcpdump -r to read it back and clean it out a bit more Cheers Duncan Xavier Cardil wrote: > Hi, do somebody knows how to sniff RTP and SIP traffic only for a > faster debugging ? > > Thanks. > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anonymous Connection form IP to use specific Context
If you create a peer definition and put the host address in it and the context you want it to go to you should be fine Cheers Duncan David Klaverstyn wrote: > > Hi All, > > > > I never saw a reply to this question. Is anyone able to assist? > > > > Regards > > David. > > > > *From:* asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David > Klaverstyn > *Sent:* Friday, 19 June 2009 2:28 PM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* [asterisk-users] Anonymous Connection form IP to use > specific Context > > > > Hi All, > > > > How can I force an anonymous SIP connection from a certain IP address > to use a specific context rather than the default one defined in sip.conf. > > > > I am using Asterisk 1.6.0.9 > > > > Regards > > *David Klaverstyn* > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call File Channel
If you use a Local channel to dial it then it will fall under the same rules Channel: Local/numbertod...@the-context-you-want This gets a CDR produced, it does pay to check everything works the same but it should be fine Cheers Duncan David Gibbons wrote: > > Context: is what the call is dumped into after it is answered, at > extension Extension:. I don’t think it’s related to how the call is > placed. > > I can dial the local extension SIP/170 but I’m not sure where that > gets me. > > Basically I want to have the same failover that I have for all other > outgoing calls on these automatic calls… > > Thanks > > Dave > > *From:* asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny > Nicholas > *Sent:* Wednesday, August 12, 2009 5:17 PM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* Re: [asterisk-users] Call File Channel > > Ok. Here’s how you would do that: > > Channel: SIP/170 (some local extension) > > CallerID: SIP/104 (another local extension) > > MaxRetries: 1 > > WaitTime: 60 > > retryTime: 5 > > Context: your_context > > Extension: s > > This should create an extension call using your context. The context > can then dial out as you write it. > > > > *From:* asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David > Gibbons > *Sent:* Wednesday, August 12, 2009 4:10 PM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* Re: [asterisk-users] Call File Channel > > Thanks Danny, > > I do have a dial cmd with multiple arguments in my normal outgoing > context. I guess my question really is: > > How do I tell the call file using “Channel: XXX” to use my outgoing > context instead of Zap/g1/xx or sip/trunk_x/xx directly? > > -Dave > > *From:* asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny > Nicholas > *Sent:* Wednesday, August 12, 2009 5:05 PM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* Re: [asterisk-users] Call File Channel > > Exten => s,1,Dial(SIP/trunk_x/#1&SIP/trunk_y/#2&ZAP/g1/#3,60) > > > > *From:* asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David > Gibbons > *Sent:* Wednesday, August 12, 2009 3:59 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Call File Channel > > I know I’m missing something here (been a long day)… > > How can I specify more than one channel in a call file? > > I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1… > > Thanks > > Dave > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk???
I am using the beta and its pretty good for remote access for clients It would help if they had some discount structure for volume Cheers Duncan Pascal Bruno wrote: > Not sure if anybody noticed, but it seems like Skype For Asterisk is out. > > $66 per channels, pretty pricey > > http://store.digium.com/productview.php?product_code=1SFA0001 > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing
Generally with FreePBX the ring options are set in the General Options - you can set the Dial options which are normally tr, but I guess that isn't working for you. The SIP files you could edit would have custom in their name, otherwise your changes will be overwritten when you reload freepbx You could put this in sip_general_custom.conf which will be included Cheers Duncan John A. Sullivan III wrote: > Oops! - You're using FreePBX - someone who knows more about FreePBX will > have to help you as I don't. May I also suggest that you bottom post in > future responses rather than top post; that makes it a little easier to > follow. Good luck - John > > On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote: > >> here is my sip.conf. i don't see it. >> ;; >> ; Do NOT edit this file as it is auto-generated by FreePBX. All >> modifications to ; >> ; this file must be done via the web gui. There are alternative files >> to make; >> ; custom modifications, details at: >> http://freepbx.org/configuration_files ; >> ;; >> ; >> >> [general] >> >> ; These files will all be included in the [general] context >> ; >> #include sip_general_additional.conf >> >> ;sip_general_custom.conf is the proper file location for placing any >> sip general >> ;options that you might need set. For example: enable and force the >> sip jitterbuffer. >> ;If these settings are desired they should be set the >> sip_general_custom.conf file. >> ; >> ; jbenable=yes >> ; jbforce=yes >> ; >> ;It is also the proper place to add the lines needed for sip nat'ing >> when going >> ;through a firewall. For nat'ing you'd need to add the following >> lines: >> ; nat=yes , externip= , localhost= , and optionally fromdomain= . >> ; >> #include sip_general_custom.conf >> >> ;sip_nat.conf is here for legacy support reasons and for those that >> upgrade >> ;from previous versions. If you have this file with lines in it >> please make >> ;sure they are not duplicated in sip_general_custom.conf, if so remove >> them >> ;from sip_nat.conf as sip_general_custom.conf will have precedence. >> #include sip_nat.conf >> >> ;sip_registrations_custom.conf is for any customizations you might >> need to do to >> ;the automatically generated registrations that FreePBX makes. >> ; >> #include sip_registrations_custom.conf >> #include sip_registrations.conf >> >> ; These files should all be expected to come after the [general] >> context >> ; >> #include sip_custom.conf >> #include sip_additional.conf >> >> ;sip_custom_post.conf If you have extra parameters that are needed for >> a >> ;extension to work to for example, those go here. So you have >> extension >> ;1000 defined in your system you start by creating a line [1000](+) in >> this >> ;file. Then on the next line add the extra parameter that is needed. >> ;When the sip.conf is loaded it will append your additions to the end >> of >> ;that extension. >> ; >> #include sip_custom_post.conf >> >> >> >>> From: jsulli...@opensourcedevel.com >>> To: asterisk-users@lists.digium.com >>> Date: Wed, 19 Aug 2009 12:17:15 -0400 >>> Subject: Re: [asterisk-users] outbound calls not ringing >>> >>> sip.conf >>> >>> On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: >>> >>>> we are using Aastra 57i >>>> >>>> i don't see that setting. where is it at? >>>> >>>> >>>>> From: jsulli...@opensourcedevel.com >>>>> To: asterisk-users@lists.digium.com >>>>> Date: Wed, 19 Aug 2009 11:07:21 -0400 >>>>> Subject: Re: [asterisk-users] outbound calls not ringing >>>>> >>>>> On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: >>>>> >>>>>> I put a post on here about my issues with outbound calls not >>>>>> >>>> ringing >>>> >>>>>> but i haven't resolved it. so i am trying again. >>>>>> >>>>>> When i dial any outside number i dont get a ring tone at all. >>>>>> >> when >> >>>> the >>>> >&
Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
I am a big fan of ubuntu LTS and freepbx and recently I saw mention of a custom module to add auto configuring endpoints for linksys (but i cna't find it again right now) Trixbox had too much stuff whereas the source install of just what you want is nice and clean Cheers Duncan Jeff LaCoursiere wrote: > On Mon, 31 Aug 2009, ilker Aktuna wrote: > > >> Thank you. >> That was quick and helpful :) >> >> Then I'll just "make" and "make install" >> What should I backup, in case of rollback requirement ? >> > > That's a bit tougher. At the least /usr/lib/asterisk/modules, > /etc/asterisk, and /usr/sbin/asterisk... someone else may need to chime > in here... > > I've always been a fan of trixbox, and I have done a lot of installations, > but when it comes down to it all I really want it for is for a quick > installations of asterisk and FreePBX. I don't think I actually use any > of the trixbox-only features. I've also been enamored with Ubuntu of > late, and have dumped CentOS. YMMV, but you might consider starting over > with a clean build of the linux of your choice, and doing asterisk + > addons + FreePBX from source. > > j > > >> Thanks. >> >> >> - Original Message - >> From: "Jeff LaCoursiere" >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> >> Sent: Monday, August 31, 2009 11:15 PM >> Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation >> >> >> >>> On Mon, 31 Aug 2009, ilker Aktuna wrote: >>> >>> >>>> Hi, >>>> >>>> My Trixbox 2.8.0.1 installation includes the following Asterik version: >>>> 1.6.0.9-samy-r27 >>>> >>>> I am having some problems with it and I think they might be solved if I >>>> use the latest Asterisk version. >>>> Is it a good idea to update Asterisk in Trixbox externally ? >>>> >>> I've done it in the 1.4 branch. >>> >>> >>>> Is it safe ? >>>> >>>> >>> Should be, as long as you stay within the same branch. That being the >>> case, I would stick with 1.6.0.14 if I were you. Make sure you don't >>> "make samples" :) >>> >>> j >>> >>> >>>> If so, which version should I prefer ? >>>> 1.6.1.5 or 1.6.0.14 ? >>>> >>>> Thanks, >>>> ilker >>>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >>> Register Now: http://www.astricon.net >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Today's problem: Inbound call routing
Usually that message comes up because the caller is anonymous and freepbx doesn't like anonymous calls by default. There is an option to accept anonymous calls, or set the incoming trunk to accept calls from the specific IP address Of course it could be something else Cheers Duncan Ben Schorr wrote: > Sorry, I'm brand new at Asterisk (and/or FreePBX). I'm going to have to > figure out what all those things are before I can show them. > > I'll have to get back to you. > > Ben M. Schorr > Chief Executive Officer > __ > Roland Schorr & Tower > www.rolandschorr.com > b...@rolandschorr.com > > > >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- >> boun...@lists.digium.com] On Behalf Of Tzafrir Cohen >> Sent: Friday, October 09, 2009 9:54 AM >> To: asterisk-users@lists.digium.com >> Subject: Re: [asterisk-users] Today's problem: Inbound call routing >> >> On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote: >> >>>> -Original Message- >>>> >>>>> Too simple, apparently, when I dial the number the caller gets a >>>>> recording that it's a non-working number and this is what I see >>>>> > in > >>> the >>> >>>>> CLI: >>>>> >>>>> Extension '8085255935' in context 'default' from '808xxx' >>>>> > does > >>> not >>> >>>>> exist. Rejecting call on channel 0/1, span 1 >>>>> >>>>> >>>> That is a pretty clear error message. >>>> >>> Yes, I thought so. But how do I fix it? >>> >>> >>>>> So...other than creating the inbound route and assigning it to >>>>> > an > >>>>> extension I apparently have to do something else. Any >>>>> > suggestions > >>> as >>> >>>>> to what that might be? >>>>> >>>> You manage your dialplan with FreePBX. This mailing list supports >>>> >>> Asterisk. I >>> >>>> have no problem with questions about FreePBX systems. But they >>>> should also be phrased as Asterisk questions. This is a FreePBX >>>> > question. > >>> I see, so this isn't an Asterisk problem it's a FreePBX problem? >>> >> "Creating an inbound route" is FreePBX speak. This is a FreePBX >> > question. > >> Please ask an Asterisk question. >> >> For instance, show a dialplan trace, show the respective dialplan, >> > show the > >> respective channel configuration. >> >> -- >>Tzafrir Cohen >> icq#16849755 jabber:tzafrir.co...@xorcom.com >> +972-50-7952406 mailto:tzafrir.co...@xorcom.com >> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: >> http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for hunt groups?
Freepbx comes with setup of ring groups and queues with different hunt strategies Also it has Flash Operator Panel which gives you the state of the system in real time graphical format No money - just a small bit of installation time and learning how to use it Cheers Duncan Ken D'Ambrosio wrote: > Hi, all. I've got an Asterisk box installed that I'd really like to > leverage -- and installing a GUI for hunt groups would be awesome. So > long as I can have a trial copy, I could even pay money. It would have to > be able to make use of both SIP and ZAP extensions. > > Suggestions? > > (Note: I wouldn't much care about the GUI, myself, but my boss is all over > one.) > > Thanks! > > -Ken > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4
I have the same challenge and issue, the server dies shortly after being fired up, although I am using Asterisk 1.2 Even with strace its very trying to work out whether the messages are errors or importance or just run of the mill All advice and options appreciated Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Thursday, 17 May 2007 11:37 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4 On Wed, May 16, 2007 at 03:22:35PM +0200, Jack wrote: > Hi, > > has anyone managed to get hudlite server working on a Debian Etch > based installation of Asterisk 1.4? > > So far I managed to eliminate all error messages, but the process is > killed directly after starting the hudlite server without showing any > error messages. > > I would be very happy if anyone can give me some hints or point me to > a installation guide. What I would do in such a situation, is run everything under strace. However, recall that you're dealing with a proprietary program here. The only ones who have the full information to help you are Fonality. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] DTMF not working using *98, but OK on inbound routes?
I have this happening with a Cisco 7960 - I can't see what the difference is, I have asterisk 1.2.13 and a number of 7960s which happily work, as well as some 7961s which also work. However one 7960 doesn't, although it dials quite happily but that's probably due to dtmf being put into SIP rather than inband. Why one works and the other doesn't I don't yet know. Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Sent: Thursday, 17 May 2007 2:40 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF not working using *98,but OK on inbound routes? Has anyone seen anything like this: I dial *98. Asterisk says "Password?" I punch in the password, and the system doesn't recognize the tones. However, if I dial my own number and ignore the incoming call, it goes to voicemail, and then I can get into voicemail. I have a sneaking suspicion that Asterisk is somehow not recognizing the DTMF tones somewhere along the way. This happens intermittently with Linksys ATAs and Polycom phones. Using a Cisco 3640 VOIP router. Any ideas on what to check? ___ --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] WiFi SIP phones
I have a recent dual gsm /wifi from e28 via Skyvoice. (http://myskyvoice.com/) Its built to use voip or gsm and is about the same price as existing wifi phones. My main hassle is it doesn' yet do WPA - WEP's okay and they say WPA is only a firmware load away ;-) , and it has a browser to login if you need to. So far so good and then to some degree I am not sure I would use a wifi only phone again That said wifi voip is still occasionally flaky but I much prefer it to soft clients on the laptop. Cheers Duncan _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, 24 May 2007 2:50 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] WiFi SIP phones I travel a lot for work. I frequently find hotels that have wifi, free or otherwise available. But I've yet to find it anywhere near sufficient to support voip applications. At least not good enough to compel me to not use my cell phone. If you have control of the host LAN then you can ensure it meets the needs of a wifi SIP phone, otherwise why bother. Has anyone ever seen anyone making a voip call on a wif handset ata public hotspot? While that would score many geek points I doubt it would work in many places. About 18 mo ago I bought the Hitachi Cable WIP5000 handset. It was seriously flawed so I resold it after a few months and settled on the Aastra desk phone. I do wish the cordless handsets were a little more like a Panasonic cordless phone...more buttons...easier to program, etc. Michael On Wed, 23 May 2007 21:59:03 -0400, Justin Moore wrote: >On 5/23/07, Michael Graves <[EMAIL PROTECTED]> wrote: >> I must say that I've VERY happy with my Aastra 4801 CT phones. I think that >> they're DECT. Each can have up to six cordless handsets. Technically its a 9 >> line phone, but if you use G.729 you can only sustain two calls at once. I >> can have a call on the portable and easily take another on the base. > >I am also an extremely happy user of an Aastra 480i CT. Awesome phone. >However, I was under the impression that the OP was looking for a WiFi >phone that could be carried from place to place, but I may be wrong... > >-- >Justin Moore >aka wantmoore >--- >www.wantmoore.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] Bottom line on fax reception
I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no baby sitting, I receive about 20 and it requires no baby sitting Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and hylafax lists for much bigger examples Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 29 May 2007 7:34 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Sorry but I must have missed it if someone else responded. If the built in fax reception doesn't work very well what about the 3rd party stuff mentioned on the Asterisk Wiki? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bottom line on fax reception Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of shadowym > Sent: Monday, May 28, 2007 11:20 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [asterisk-users] Bottom line on fax reception > > Anybody?? > > -Original Message- > From: shadowym [mailto:[EMAIL PROTECTED] > Sent: Thursday, May 24, 2007 9:35 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Bottom line on fax reception > > > > So what is the bottom line? Does it work or not. I've heard stories it > works, it doesn't work, it kinda sorta works when it's not raining out > side. > Everything under the rainbow. > > What's the bottom line with recent updates on 1.2.x? Is it production > ready for fax? By production ready I mean that it just works all the > time and > doesn't need any babysitting. Do I have to worry about dropped lines, > sometimes not detecting incoming fax toneyada yada. > > I know I don't have to use fax on Asterisk but I really want to for > various reasons. Mostly incoming but outgoing is a nice to have. > Should I use an > addon package and if so which one? Any help would be appreciated. > > > > ___ > --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
[asterisk-users] Upgrading an AA50 and finding zap drivers - lack of Digium support
Hi there I have a client who has an AA50 from DIgium. I am really challenged getting any support as the client doesn't have any of the original registration or subscription info, someone did the install and left without any records. I thought okay we can ask Digium, but you can't get help wthout registering your product and you can't register your product without your subscription info. I did get one response which was to email customer services and eventually found an email address for them but that seems to have fallen on deaf ears. Perhaps my expectations are too high but it was an email a week ago and no response, not even to say go away. Its really disappointing and amazingly hard to believe such a system exists. Perhaps no one else in the world loses their registration info. I need to upgrade the AA50 to the latest firmware, and just get some general support for the setup, as it doesn't seem to have picked up Zap and I think perhaps the CF card has died. Does anyone have any pointers as who to talk to. I see lots of digium people on the list so I am hoping someone can help me. Thanks very much Merry Xmas Cheers Duncan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading an AA50 and finding zap drivers - lack of Digium support
Thanks very much Kevin But its not that clear - in fact the support email address isn't listed on the support page - everything leads to logging in with your registered product. Its incredibly frustrating and I recommend you try looking and seeing how it works for yourself if you haven't got your registration details. Having some method to handle missing details would have sorted this all out but I get stuck in a big loop. You are probably right I should have called but I would have thought email would be possible at least. It wasn't a rush for a while. Also I am not in the US so I don't tend to call overseas as standard as most companies have ways of getting to support and need emails with details anyway. I agree a 4 day holiday is approaching but a week ago it wasn't and apart from an automated response not even a whiff of a reply Cheers Duncan On 24/12/2009, at 9:16 AM, Kevin P. Fleming wrote: > Duncan Turnbull wrote: > >> I did get one response which was to email customer services and eventually >> found an email address for them but that seems to have fallen on deaf ears. >> Perhaps my expectations are too high but it was an email a week ago and no >> response, not even to say go away. > > The contact information for Digium's support department is clearly > listed on our web site. There are multiple methods, including just using > the plain old PSTN to call us :-) > >> Its really disappointing and amazingly hard to believe such a system exists. >> Perhaps no one else in the world loses their registration info. >> >> I need to upgrade the AA50 to the latest firmware, and just get some general >> support for the setup, as it doesn't seem to have picked up Zap and I think >> perhaps the CF card has died. Does anyone have any pointers as who to talk >> to. I see lots of digium people on the list so I am hoping someone can help >> me. > > Given the serial number of the unit the support department will be able > to research the registration and get you squared away; the only issue is > that we are rapidly approaching a four-day holiday weekend at Digium so > you would likely not get any response until early next week. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mitel integration
Having looked at the outputs into PMS they are very simple stop start records. Line by line text that can easily be recreated. They have about 4-5 fields, origin number, destination, time of call, duration, or similar things Usually they go out via a serial port or TCP port expecting a terminal to receive them so plugging into them will quickly show you what you need. Its not that you need to match the Mitel, you need to match the PMS. Best to talk to them but I have looked at it for a couple of customers who are still deciding and helped them fix their PBXs when they broke and its pretty straight forward. You just need to be able to output to either a serial or TCP port . Cheers Duncan On 28/01/2010, at 6:01 AM, Jeff LaCoursiere wrote: > > > On Wed, 27 Jan 2010, Mark Wiater wrote: > >> the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware >> logs in the same manner, different ports. > > This particular model (need to get the model number) has a serial > connection. I'm all for putting a serial sniffer between them (if they > let me!), but was really hoping someone had already done this and could > give me a headstart. > > I'll investigate the ethernet options, though, as that would make more > sense anyway! If the PMS will talk over ethernet I'll try to pretend to > be a 3300. > > Cheers, > > j > >> >> On 1/27/2010 11:00 AM, Steve Howes said: >>> On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: >>>> Sounds good to me, but without the spec I'm stuck in a catch 22! >>> >>> tcpdump? (assuming IP). Bet its fairly simple plain text or something. >>> >>> Steve >>> >> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
The other way on Debian/Ubuntu is just to test the existence of the dir and create it if needed If you add this to the /etc/init.d/asterisk near the start you should be fine if ! [ -d /var/run/asterisk ] ; then mkdir /var/run/asterisk chown $AST_USER.$AST_GROUP /var/run/asterisk exit 0 fi Set the ownership as required Cheers Duncan On 11/02/2010, at 7:50 AM, Brian wrote: > On Wed, 2010-02-10 at 11:24 -0600, Jason Parker wrote: >> Brian wrote: >>> Each time the server is rebooted Asterisk duly >>> deletes the manually created /var/run/asterisk directory - quite why it >>> does this I just don't know - perhaps it is a bug? >>> >> >> Your assumption is incorrect. Some Linux distributions will empty /var/run/ >> on >> boot, just as they do with /tmp/. > Thanks Jason - that had never dawned on me, but I've just tested it and > indeed it does. > >> I do believe you're right, however, in >> suggesting that there is a bug in Asterisk. It appears that Asterisk >> creates >> /var/run/asterisk/ during install and assumes that it will always exist. > Agreed - that would make sense that by default it thinks the directory > is there. The workaround / fix is to take out the (!) > from /etc/asterisk/asterisk.conf and allowing the default setting of: > astrundir => /var/run to come into play. It then puts the .pid and .ctl > in the root of /var/run >> >> Some of the sample init scripts (Debian) create that directory before >> starting >> Asterisk. This should be done in all of them (or in Asterisk itself, >> maybe?). > The one I had didn't - but I could have added it. I just wanted to be > sure I was doing the right thing. >> >> Please report an issue on http://issues.asterisk.org/ > Done - but I'm a bit embarrassed as it seems so trivial. > > Thank you for your help. >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No RTP from asterisk?
On 1/03/2010, at 2:41 PM, Peter Serwe wrote: > I checked the firewall, iptables -L showed no rules whatsoever. No other > traffic has indicated it was blocked, iptables was set in allow all > everywhere mode. > > I went ahead and turned it off, still don't have RTP. No audio either > direction via lines registered. > > G729 is completely disabled from all trunk groups and users, only using G711 > at this point. > > Peter > > The asterisk rtp debug should show if asterisk is sending audio or receiving packets but its not nearly as useful as tcpdump. tcpdump udp port 5060 -s0 -A will give you all the SIP. But just dumping all traffic between asterisk and the host will give you a view on RTP - you should see it take off when a call is setup if its not blocked You should see a SIP Invite to setup a call with the audio destination - this should be your asterisk box and the far end depending on who is doing what. You should look at the address and also whether both sides are providing a mutually acceptable audio formats. If there are no agreed audio formats you won't get rtp. The c= in the session setup indicates the addresses each site is using for media. Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk system for church call center
Hi Frank I have found Freepbx on top of Asterisk a good solution for the church I look after and the rest of my customers, the callcentre functions you need are built in it and if they have someone technical then they can expand what they are doing It has both queues and ring groups (which are often all they need) I would imagine you just send them to an IVR or mailbox to ask for their name and details then move them into a higher priority second queue Elastix has Sugar in it I believe and looks okay but I like debian/ubuntu as a base distribution rather than Centos so haven't gone that way. However most of my customers still struggle with the concepts involved in telephony so while they are happy to look at it while I am there they forget quickly how to drive it or lose their nerve, especially the church which is faith focussed rather than tech focussed ;-) Because of that I think you want the easiest system for you to maintain remotely and elastix or freepbx is pretty easy. It also allows you to say this is available, this is not which is useful in narrowing down requirements. Cheers Duncan On 30/03/2010, at 9:58 AM, Frank Church wrote: > I have been asked by my church to recommend a VoIP system which can do > the following. > > They do internet radio shows which are sometimes broadcast on radio. > > They are looking for a system which does the following for about 5 > agents, exactly as they have described it. > > 1. Take incoming calls > > 2. Put them on hold if there is no one to handle the call immediately, > or transfer them to an available agent > > 3. Take down their details, and number, (if this can be retrieved and > saved from the caller id, thats better) > > 4. Get them to hold on after taking their details if they still want to hold > > 5. Call them back when the backlog is cleared up. > > I have a fairly good grasp of the hardware and programming part of > Asterisk, having compiled it more than a few times and implemented > A2Billing phone card and call shop system with it. > > But the type of software suited to the Call Center side is where my > knowledge gap lies. > > I am looking for solutions based on the usual Asterisk distributions > like AsteriskNow, trixbox, elastix etc, whether ready packaged or > requiring additional customization. > > > The matter of whether they will use soft phones, or regular phones > with headsets is also something to consider. Soft phones with good > GUI's may be preferred if more cost effective for them, although my > personal preferences are with hard phones. > > Any recommendations - the ease of software for the end users is the > main thing for me, and integration with the database for taking > customers details is the main thing for me. One of the distributions > with SugarCRM comes to mind here. > > Sorry for cross-posting, but ready made and commercially supported > systems are not ruled out, if they come within their budget. > > Regards > > > Frank Church > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial extension and play sound file from shell on asterisk server?
Have a look at the call files examples of voipinfo http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Its not too hard to do what you want Cheers Duncan On 8/04/2010, at 11:00 PM, Brian J. Murrell wrote: > I want to use Asterisk as a general message delivery system here. > > That is, I want to be able to have a (shell, perl, etc.) script on my > Asterisk server dial an extension, wait for it to be answered and then > play a sound file and then hang up, or even wait for a response or > reactions to some IVR. > > Certainly if I had a SIP library, I could have the script simply look > like a SIP extension but that seems like it should be an unnecessary > added complication. Instead I am looking for a more direct API to > Asterisk which allows a process to interact like a "phone" but not > actually be one. > > Is this at all possible? Anyone done anything like this? > > b. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK ISDN with Asterisk
Hi Henry staff member can take the call). If there is another way to do this with anaologue lines, i'm open to suggestions. I have looked at using a You used to be able (still can probably) do this with a thing called "auxiliary working" from BT on analogue lines. Two lines with one number, the second would take the call if the first were busy. I'd be tempted to steer clear of analogue though unless you really want it - I'm not sure how well it works directly hooked into the Asterisk server, but analogue on the voice gateways I've configured is just hard work a lot of the time. Good luck Dunc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Optus Australia Multiline SHDSL service
Hi, I am currently trying to find a replacement for a dinosaur PBX and want to replace it with a VoIP solution. We have just moved our lines over to an Optus Multiline from a Telstra ISDN Onramp 30 service with 100 lines. My question for you good people is what sort of hardware do I need to interface Asterix into the Optus Multiline? The Optus service is terminated in my office to a SHDSL NTU from Adtran and has two RJ45 conenctors on the back of it. Has anybody tried this yet? Thanks in advance. Duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
