Re: [Asterisk-Users] Bri problem
Hi Do you have immediate=no in your zapata.conf ? immediate = yes makes asterisk pass all incoming calls to s extension. Hope that helps you --- Altus Snyman [EMAIL PROTECTED] escribió: Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension s?? Is this something to do with the telecoms provider or a asterisk config? Please Help ore advice Thanks Altus ___ 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 __ Renovamos el Correo Yahoo!: ¡250 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es ___ 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
Re: [Asterisk-Users] Bri problem
Thanks Will have a look On Fri, 2005-02-11 at 09:59, Edin Kozo wrote: Hi Do you have immediate=no in your zapata.conf ? immediate = yes makes asterisk pass all incoming calls to s extension. Hope that helps you --- Altus Snyman [EMAIL PROTECTED] escribió: Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension s?? Is this something to do with the telecoms provider or a asterisk config? Please Help ore advice Thanks Altus ___ 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 __ Renovamos el Correo Yahoo!: ¡250 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es ___ 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 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
Re: [Asterisk-Users] Why echo occurs
On Fri, 2005-02-11 at 15:32 +0800, Steve Underwood wrote: What you said was not actually wrong. However, 9 out of 10 people reading it will see echo is something that affects only analogue phones. People keep saying this. Its even in comments in the * source code. Its wrong. Yeah, I should have added that we where at a point of picking nits. -- Steven Critchfield [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Bri problem
On Fri, 11 Feb 2005, Edin Kozo wrote: --- Altus Snyman [EMAIL PROTECTED] escribió: I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension s?? Is this something to do with the telecoms provider or a asterisk config? Do you have immediate=no in your zapata.conf ? immediate = yes makes asterisk pass all incoming calls to s extension. Another possibility is that the pstn sends the incoming DID as overlap digits. There are telcos that do, but they are rare so it is unlikely. If all else fails do a pri intense debug span XX and look at the results. Peter ___ 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
Re: [Asterisk-Users] Searchable Mailing Lists NooB Question
Rich Adamson wrote: Looks like your numbers add the transmit and receive data rates together, which is not a realistic way to discuss bandwidth consumption. An IAX link consumes about 22kb/s (round it to 30kb/s, who cares) in the transmit direction, and another 22kb/s in the receive direction. (There's your 60kb/s.) You are right. I was not watching bandwidth directions seperately. I have a mangle rule that marks all voip traffic, then have it set up in a queue tree to give it priority. I was watching the bandwidth for that individual mangle rule, which is comprised of all transmitted and received voip connections. Actual traffic is 32kb/s for GSM, and about 60kb/s for G711. Thank you for clearing that up for me. Dan ___ 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] i want to load chan_h323.so
I use Fedora core 2, and openssl-0.9.7, expat-1.95.7 is installed by rpm packages. I downloaded pwlib-1.5.2 and openh323-1.12.2 at my home directory(/root/root_src), asterisk 1.0.4 at directory /usr/src/ and have installed successfully. Asterisk runs normally, but cannot load module chan_h323.so. The message is : # asterisk vvvgc . . . Asterisk Ready. *CLI load chan_h323.so /root/root_src/openh323/lib/libh323_linux_x86_r.so.1.12.2: undefined symbol: _Z13v pb_dial_synciPc Unable to load module chan_h323.so *CLI Please give me your solutions. Thank you for your reading. My working log is : # tar xvfz pwlib-1.5.2.tar.gz # tar xvfz openh323-1.12.2.tar.gz # cd /root/root_src/pwlib # ./configure # make # cd /root/root_src/openh323 # ./configure # make opt #cd /usr/src #exportCVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot #cvslogin #cvsco-rv1-0asterisk # echo $PWLIBDIR /root/root_src/pwlib # echo $OPENH323DIR /root/root_src/openh323 # echo $LD_LIBRARY_PATH /root/root_src/pwlib/lib:/root/root_src/openh323/lib # cd /usr/src/asterisk/channels/h323 # make # cd /usr/src/asterisk # make install == ___ 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] i want to load chan_h323.so
I use Fedora core 2, and openssl-0.9.7, expat-1.95.7 is installed by rpm packages. I downloaded pwlib-1.5.2 and openh323-1.12.2 at my home directory(/root/root_src), asterisk 1.0.4 at directory /usr/src/ and have installed successfully. Asterisk runs normally, but cannot load module chan_h323.so. The message is : # asterisk vvvgc Asterisk Ready. *CLI load chan_h323.so /root/root_src/openh323/lib/libh323_linux_x86_r.so.1.12.2: undefined symbol: _Z13v pb_dial_synciPc Unable to load module chan_h323.so *CLI Please give me your solutions. Thank you for your reading. My working log is : # tar xvfz pwlib-1.5.2.tar.gz # tar xvfz openh323-1.12.2.tar.gz # cd /root/root_src/pwlib # ./configure # make # cd /root/root_src/openh323 # ./configure # make opt #cd /usr/src #exportCVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot #cvslogin #cvsco-rv1-0asterisk # echo $PWLIBDIR /root/root_src/pwlib # echo $OPENH323DIR /root/root_src/openh323 # echo $LD_LIBRARY_PATH /root/root_src/pwlib/lib:/root/root_src/openh323/lib # cd /usr/src/asterisk/channels/h323 # make # cd /usr/src/asterisk # make install ___ 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] How to monitor externip automatically?
Hello list, I am behind a NAT router and therefore need to have externip= in sip.conf. Whenever the isp resets its DHCP server, I have to change the setting in order to make Asterisk work. Is there a way to make Asterisk get the external ip automatically in case I am able to do it from wherever I am? Thor Atle Rustad ___ 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] Proper handling of incoming IAX/SIP callerids to be able to call back - why is calleridnum stripping dots out of number ?
Hi,
I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323
callerids so they can be called back if needed.
I have three incoming contexts for sip, iax and h323 calls.
To each incoming call I'd like to prepend certain number that will be
catched with pattern matching on output calls. For instance for iax I have:
[from-iax]
exten = s,1,NoOp(IAX call from outside ${CALLERID}: Name: ${CALLERIDNAME},
Number: ${CALLERIDNUM})
exten = s,2,Wait,2
exten = s,3,SetCIDNum(1${CALLERIDNUM})
exten = s,4,NoOp(IAX call from outside ${CALLERID} changed : Name:
${CALLERIDNAME}, Number: ${CALLERIDNUM})
;exten = s,3,SetCallerID(${CALLERIDNUM})
exten = s,5,Goto(from-pstn,s,1)
and when executed :
-- Accepting unauthenticated call from 193.77.90.224, requested format =
2, actual format = 2
-- Executing NoOp(IAX2/[EMAIL PROTECTED]/2, IAX call from outside Robert
Rozman [EMAIL PROTECTED]: Name: Robert Rozman| Number:
[EMAIL PROTECTED]) in new stack
-- Executing Wait(IAX2/[EMAIL PROTECTED]/2, 2) in new stack
-- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]/2, [EMAIL PROTECTED]) in
new
stack
-- Executing NoOp(IAX2/[EMAIL PROTECTED]/2, IAX call from outside Robert
Rozman [EMAIL PROTECTED] changed : Name: Robert Rozman| Number:
[EMAIL PROTECTED]) in new stack
-- Executing Goto(IAX2/[EMAIL PROTECTED]/2, from-pstn|s|1) in new stack
-- Goto (from-pstn,s,1)
-- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2,
1?from-pstn-reghours|s|1:) in
new stack
So in Variable ${CALLERID} is right (with dots in iax address) but not the
case with ${CALLERIDNUM} that has dots stripped off. So now I cannot
callback cause proper address is lost. On outgoing part I have:
[outbound-iax]
exten = _2.,1,NoOp(Outbound IAX call from local extension ${CALLERID} to
${EXTEN:1})
exten = _2.,2,NoOp
exten = _2.,3,Dial(IAX2/${EXTEN:1})
exten = _2.,4,Congestion()
but it won't work, cause dots were stripped out
Is there something wrong with my way of handling this problem? Is there any
better way to handle incoming calls, so they can be called back with click
on recent calls ?
How to handle this when there is equpment that cannot show callerid names
(like BT100) - for each extension separately or somehow different ?
Has anyone working example of proper handling of incoming nontrunk
iax/sip/h323 calls ? Any advice ?
Thanks in advance,
regards,
Rob.
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Re: [Asterisk-Users] asterisk@home scary log
On Thu, Feb 10, 2005 at 01:30:03PM -0600, Steven Critchfield wrote: If you are going to rely on keys, you need to have both directions identified. Nothing like sending a valid key to a man-in-the-middle. That's indeed one atvantage of keys over passwords. Even if the server is compromised, your secret keys are safe. The server only needs to know your public keys, and some proofs that you have the matching private key (using it to sign some random data the server sends). Anyway, with ssh you'll normally be notified of a spoofed host, because the host key won't match. A decent ssh client won't let you to connect or will give you a very nasty warning. Unless it is the first time you connect from that host/account to the server. sshophilicly yours -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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
Re: [Asterisk-Users] asterisk@home scary log
On Thu, Feb 10, 2005 at 10:51:33AM -0600, Rich Adamson wrote: There are multiple password guessing tools commonly available on the Internet. I eval'ed one of the tools and it took five seconds to guess a password that was five characters in length. It took an hour to guess a password that was eight characters, and around twenty-four hours to guess a password that was eight characters made up of uppercase, lowercase and non-alpha characters (eg, complex). Regardless, the guessing process is simply how much time does one want to devote to doing it (eg, what's the return value for spending the time exploiting a system). Sorry, not in my tests. I used John the Ripper (http://openwall.com/john/ ), which is a tool for cracking passwords from password files using dictionaries and brute force. The password files had passwords in varrying quality, and cracking time was indeed affected. all-numbers password were guessed almost immidietly. [*] Well-composed passwords of 8 characters were not cracked by brute-force in resonable time. [*] passwords that should be dialed from phones are relatively short and all-numbers. Are they never exposed to the internet? It doesn't make much difference whether one exposes telnet or ssh. Both can be exploited. But, the more complex you make the password, the more time-consuming and difficult it is to guess it. So, if you must expose either telnet or ssh, make your passwords very long and complex. If your O/S has the capability to lockout the account after 'xx' failed passwords, then do that. And allow crackers to lock you out. A silly and effective DoS attack. Automatically resetting the process after 'y' minutes disrupts the guessing process without the hacker knowing it, but still allows you access after that auto reset. Using something like seven failed attempts with a five minute reset is more then adequate in most cases. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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
Re: [Asterisk-Users] i want to load chan_h323.so
On Fri, Feb 11, 2005 at 06:09:06PM +0900, ?? wrote: If you actually sent text instead of an HTML only email, you increase the chance someone will actually read your message... ___ 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] Multiple incomming contexts
Hi list I'm trying to implement sourcerouting on a distributed installation, but I can't get contexts to work right. My goal is to do a Dial([EMAIL PROTECTED]) and vary the somecontext based on different criteria. This is going on over trunked IAX2 links. How do I set up my IAX-accounts to manage this? I have tried to play around with 'context' and 'peercontext' on the server being dialed, but no luck. Is it legal to have multiple 'context' lines in one object? Is what I'm trying to do possible? Any help appreciated. Eivind Trondsen LinuxLabs AS -- Eivind TrondsenTlf: +47 23 89 71 85 LinuxLabs AS Mob: +47 928 40 009 --- http://www.linuxlabs.no--- Drift - Overvåkning - Rådgivning ___ 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
Re: [Asterisk-Users] asterisk@home scary log
On Thu, Feb 10, 2005 at 11:44:37AM -0600, Steven Critchfield wrote: I know for a fact that Debian does NOT allow root logins except from console. Debian isn't allowing root logins from X anymore due to the likely hood for you to try and use root for more than administration. Debian does not disable root logins on ssh by default, at least not in testing (Sarge) in the package ssh. I don't know about other sshds gdm disables root login. I don't know about other DMs. telnetd and rshd both use login and login's pam file checks the securetty module, so root login is indeed disabled for them. (The latter two are quite standard among linux distros). I know Mandrake does annoying things if you try to login as root on anything but console to also discourage it's use. SuSE uses a cute annoyance (or used, in one version) : if you login as root, your default wallpaper is a scary red bomb with a lit fuse. I hopes that this delivers the message. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] Help with dial command and h, H and g parameters
I'm trying to find some live examples on how to use the h, H and g parameters on the dial command (http://www.voip-info.org/wiki-Asterisk+cmd+dial) Any ideas? I was testing with the code below but after pressing * nothing happens (only after a long pause the goodye file was played) [testset] exten = 1023,1,NoCDR() exten = 1023,2,Dial(SIP/1023,30,Hg) exten = h,1,Background(goodbye) exten = h,2,Hangup exten = i,1,Hangup exten = t,1,Hangup ___ 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
Re: [Asterisk-Users] How to monitor externip automatically?
Have you considered one of the dynamic DNS services, (e.g. http://www.dyndns.org/services/dyndns/)? Looking through the Wiki it seems that there is no requirement to use an actual IP address for externip=, in fact I found one posting that explicitly discouraged their use. Richard Thor Atle Rustad wrote: Hello list, I am behind a NAT router and therefore need to have externip= in sip.conf. Whenever the isp resets its DHCP server, I have to change the setting in order to make Asterisk work. Is there a way to make Asterisk get the external ip automatically in case I am able to do it from wherever I am? Thor Atle Rustad ___ 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 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] Re: Detect hangup
Hi, indications.conf has nothing to do with hangup detection. Instead it defines how to signal the line state to theremote party, e.g. what tone you will hereas a busy indication. If you wantasterisk to recognize hangups and standard busy detection doesn't work for you, you would have to adjust dsp.c. ForZaptel channels you have to configure busydetect=yes and busycount=3 for instance in zapata.conf. Tobias ___ 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] Dial and congestion
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Can the Dial() command tell the difference between busy and congestion? At the moment it seems to be treating them both the same on my server. I want to route the calls out via a SIP gateway unless that is congested, in which case dial out through my POTS line (using an X100P). It seems a bit pointless to try dialling the POTS line when the SIP dial is busy instead of congested. (I expected Dial() to treat congestion like other network error conditions such as a timeout) - Steve Jabber: [EMAIL PROTECTED] Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) Comment: Public key available at http://www.nexusuk.org/pubkey.txt iD8DBQFCDKQQ5zUOsIV3bqERArs+AKCndMZ5x/mKdv36ifwKP7eI7LczOgCbB0Zn hl6fBlPDAPJ7FOKxWvG0hCo= =g8j+ -END PGP SIGNATURE- ___ 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] Transfers to engaged extensions
Hi, I'm using zaptel with three way calling and call transfers with a hookflash. If I try transfering a call to an extension that is engaged I get an engaged tone. This is fine, but how do I get back to the caller? If I hookflash again I seem to put on a three-way call and the caller can hear the beeping. I can press hookflash again but I'd prefer the caller to hear only the hold music and then me speaking. Is this intentional or am I doing something wrong? Robie. ___ 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
Re: [Asterisk-Users] zaphfc - problems with hangup detection?
If you check the latest info in bristuff this bug is now fixed in 0.2.0-RC7, too bad that zaphfc is not yet finished for single HFC-S cards but if you have quad or octobri it should work On Fri, 11 Feb 2005, Stefan Gofferje wrote: Hi folks, in the past, I have encountered several situations where internal ISDN phones, connected to a HFC-S card, running in NT mode, continued ringing despite the call has been hung up. THe conditions were not reproduceable. Once it was a simple Dial() statement in an incoming context, another time it was in a complex script and again another time it was from a queue. Also, SIP phones which were rung in the same script or queue DID stop ringing when the call was hung up. Does anyone has a hint on what is going on here? Are there know issues with hangup on zaphfc? I have been searching mantis and Google but haven't found anything. Regards, Stefan ___ 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] How can agent logout manually ?
Hi, I don't know how to logout agent. The trick from Wiki (stated below) doesn't work (I have CVS stable from yesterday). I get invalid login if don't specifiy Agent ID. regards, Rob. Logging off the queue manually 1.. call the extension for AgentCallbackLogin 2.. enter your password followed by # 3.. when asked for the extension number just press # You will hear a voice prompt that confirms that the agent has been logged off. ___ 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
Re: [Asterisk-Users] Dial and congestion
Hi!
Can the Dial() command tell the difference between busy and congestion?
At the moment it seems to be treating them both the same on my server.
With bristuff 0.1.0 and later a patch to Dial() is included as follows:
app_dial modification (jumps to +201 if channel is unavailable)
Apart from the above you might want to look here for ${HANGUPCAUSE}:
http://www.voip-info.org/tiki-
index.php?page=Asterisk%20variable%20hangupcause
Cheers, Philipp
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[Asterisk-Users] Newbie: ISDN E1 the same in all countries?
Hi. Im looking at ordering a 30-channel ISDN connection from telia (a swedish operator) and then using a Wildcard TE110P card with that and asterisk to do IVR. Can I be certain that the TE110P card will work with that ISDN connection? A 30 channel ISDN certainly sounds like an E1 connection, but I couldnt get any clear answers from the operator if it is. Has anyone used the TE110P card in Sweden with telia? Thanks /sverrir ___ 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
Re: [Asterisk-Users] Newbie: ISDN E1 the same in all countries?
On Fri, 11 Feb 2005, Sverrir Valgeirsson wrote: I'm looking at ordering a 30-channel ISDN connection from telia (a swedish operator) and then using a Wildcard TE110P card with that and asterisk to do IVR. Can I be certain that the TE110P card will work with that ISDN connection? A 30 channel ISDN certainly sounds like an E1 connection, but I couldn't get any clear answers from the operator if it is. An 30-channel isdn PRI is always delivered over E1 (or perhaps several aggregated into an E3 or similar). However, an E1 can be delivered over two phisucal interfaces, 75 ohm unbalanced coaxial cables terminated in two BNC connectors or as an 120 ohm balanced twisted pair terminated in an RJ45 connector. You want to order the later. The TE410P works well in Sweden so the TE110P should as well. The implementation notes that specify the isdn interfaces on the Telia network are available from http://www.skanova.se/index.asp?lev=1636. Remember that there are lots of options on isdn. Unless you are familiar with isdn telephony yourself you may consider getting some help in ordering the line and setting up your system. Peter ___ 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] chan_capi or chan_mISDN vs bristuff
Hi list! I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. I'm having continuous problems where inbound calls will not work after some time of operation (the number then appears as not in use to the caller) or also outbound calls do not work. The solution is to unload the modules, stop asterisk, re-load the modules and start asterisk again. The machine (Athlon64) already hung several times when unloading the modules (I guess the same bug/problem is is reported for SMP boxes). This problem occurs every single day and giving me really grey hairs. If I ditch the HFC-S card and replace it with another card that will work with mISDN or chan_capi will this solve my problems? Thanks for any hints / tips! Remco ___ 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
Re: [Asterisk-Users] Newbie: ISDN E1 the same in all countries?
Sverrir, 30 channel ISDN (generally known as primary rate ISDN, PRI) is a layer that runs on top of E1, just as Internet Protocol can run over Ethernet. I haven't personally worked with Telia PRIs, but have with many other telcos throughout Europe, and they're all very similar with only minor software configuration variations between them. I would be amazed if Telia were any different. You should be pretty safe with this configuration. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Sverrir Valgeirsson wrote: Hi. Im looking at ordering a 30-channel ISDN connection from telia (a swedish operator) and then using a Wildcard TE110P card with that and asterisk to do IVR. Can I be certain that the TE110P card will work with that ISDN connection? A 30 channel ISDN certainly sounds like an E1 connection, but I couldnt get any clear answers from the operator if it is. Has anyone used the TE110P card in Sweden with telia? Thanks /sverrir ___ 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 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
Re: [Asterisk-Users] Multiple SIP registrations for one account?
Hi! canreinvite=yes does not affect call accounting in any way. U sure? What for example if later on the SIP device forwards the call (note: not using #) and itself steps out of the line? Cheers, Philipp ___ 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] transferring a IAX call into a conference
When I make a call out on the Faktortel number I am then able to transfer to call to my asterisk meetme room of 801 by hitting transfer then 801 then send on my grandstream phone. This connects my faktortel trunk (and who ever is on the other end) to my conference room I can then make another call using my local pstn service and set up a 3 way (or whatever number in a conference call) Now the problem is this; If someone calls me in on my faktortel number I cant transfer them to the conference call room. It literally disconnects them each time I transfer? Why is this? What can I do to prevent this. Cheers, Dean ___ 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
Re: [Asterisk-Users] Newbie: ISDN E1 the same in all countries?
On Fri, 11 Feb 2005, Alistair Cunningham wrote: 30 channel ISDN (generally known as primary rate ISDN, PRI) is a layer that runs on top of E1, just as Internet Protocol can run over Ethernet. IP is run over just about anything that passes data, serial lines, atm, ethernet, you name it, it has probably had ip run over it. Isdn on the other hand is very unusual over anything but the various BRI configurations and E1 / T1 for PRI. Some may have multiple E1s over E3 or some other aggregated connection, but nothing more exotic. Peter ___ 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
Re: [Asterisk-Users] Re: Detect hangup
hi thanks very much I need to look into the dsp.c file.. the tone generate is abou 2 seconds on. then 0.5 off. busy changing the pabx connection so i can plug directley into a co-line. this would enable the pabx and * to use loop start.. and i think that will solve my problems thanks and enjoy weekend - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 11, 2005 1:44 PM Subject: [Asterisk-Users] Re: Detect hangup Hi, indications.conf has nothing to do with hangup detection. Instead it defines how to signal the line state to theremote party, e.g. what tone you will hereas a busy indication. If you wantasterisk to recognize hangups and standard busy detection doesn't work for you, you would have to adjust dsp.c. ForZaptel channels you have to configure busydetect=yes and busycount=3 for instance in zapata.conf. Tobias ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.___ 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
Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
Remco Barende wrote: I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. [..] I have three different locations with HFC cards. I had the same stability problems on ALL of the installations. Since RC5 plus the florz patch *ALL* of the stability problems have vanished. No more seconds of silence, no more unavailability messages. It just works now. I won't touch the installations for a long time :-) If I ditch the HFC-S card and replace it with another card that will work with mISDN or chan_capi will this solve my problems? I have good results with an AVM C4 card using the CAPI drivers. I started out with an old ISA AVM B1 card which had echo problems, which got fixed with some later chan_capi driver releases. -- Best regards Peer Oliver Schmidt ___ 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] Caller ID
How can I change that when there's no Caller ID instead of displaying asterisk it display something like Unknown. Because everyone is confuse when they see a call coming from asterisk. Thanks Martin ___ 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
Re: [Asterisk-Users] transferring a IAX call into a conference
If someone calls me in on my faktortel number I cant transfer them to the conference call room. It literally disconnects them each time I transfer? Why is this? What can I do to prevent this. Any CLI log from when you try that ? Help us helping you :) ___ 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] chan_capi and asterisk
Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36 WARNING[2535]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Feb 11 13:50:36 WARNING[2535]: loader.c:391 load_modules: Loading module chan_capi.so failed! my lsmod shows: Module Size Used by mISDN_capi 85312 0 kernelcapi 45088 1 mISDN_capi hfcpci 28716 0 mISDN_dsp 197248 0 l3udss132008 0 mISDN_l2 38272 0 mISDN_l1 10632 0 mISDN_core 77732 6 mISDN_capi,hfcpci,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 md5 4352 1 ipv6 235840 24 parport_pc 25024 1 lp 12396 0 parport42696 2 parport_pc,lp dm_mod 55444 0 uhci_hcd 31896 0 3c59x 36776 0 floppy 59568 0 ext3 116744 2 jbd74904 1 ext3 and my modules.conf : load = chan_capi.so [global] chan_capi.so=yes what seems to be the problem can someone help me? tahnk´s ___ 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
Re: [Asterisk-Users] asterisk@home scary log
On Thu, 10 Feb 2005 17:49:23 +, Clive Carter [EMAIL PROTECTED] wrote: I hesitated before sending this, as I have been flamed before for being a beginner. but I am newish to linux/asterisk, and I am running an ssh server. It is still running with default settings, (I dont know yet how/where to change it), and I CAN logon remotely as root. (Haven't figured out how to 'su' yet !) This is using the Rapid Xorcomm v 1.0 cd, which I believe (may be wrong) is based on a very recent version of Debian ? Perhaps xorcom have changed the default setting ? -- Clive Email : [EMAIL PROTECTED] Tel : 08444844790 Alt : 08450043366 Fax : 08444844813 SIP : [EMAIL PROTECTED] Mobile : 07031945504 Hey Clive. I thought it was mentioned earlier before in the thread, but if not, all you need to do is edit your sshd_config file. In Debian, this is located at /etc/ssh/sshd_config, but it could be different for other distros. Open that up in a text editor and then locate the line that says PermitRootLogin yes, and change that to PermitRootLogin no. Save it, and then restart SSH. On Debian, you type in /etc/init.d/ssh restart, but on other distros it might be different. Note that you'll have to be root to edit that file and restart that service. -- Dana ___ 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
Re: [Asterisk-Users] asterisk@home scary log (OT)
On Thu, 10 Feb 2005 14:05:38 -0600, Kristian Kielhofner [EMAIL PROTECTED] wrote: Derek Whitten wrote: I also call bullshit.. OpenBSD does NOT allow ssh root login by default.. why do you think that they have such an excellent security track record.. Derek, I am sorry to say, that in fact, OpenBSD does allow SSH root logins by default: http://www.openbsd.org/cgi-bin/cvsweb/~checkout~/src/usr.bin/ssh/sshd_config?rev=1.70content-type=text/plain BTW, OpenBSD's track record of security has nothing to do with whether they allow root logins by default or not. If the admin isn't wise enough to pick a -decent- root password they shouldn't be running a box connected to the internet. I think people need to start to provide HARD FACTS in some of these posts. I don't see what any of this has to do with Asterisk... -- Kristian Kielhofner Unless I'm missing something, the only line that is ENABLED in that file is this one: Subsystem sftp/usr/libexec/sftp-server The rest appear to be commented out with #, unless I'm not understanding how that all works... -- Dana ___ 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] Not register SIP and IAX
Hi all, My Asterisk server is facing some problem that I can´t even find, any registrarion for that, into the error log file. It runs normally for while and suddenly stop registering even IAX and SIP. Acting like that all my softphones and equipments once registered stop working and the only way to start working again is applying a STOP NOW command. Is there anybody there has faced into this problem someday that could help me? Thank´s jmoura ___ 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
Re: [Asterisk-Users] asterisk@home scary log (OT)
On Fri, 11 Feb 2005 09:00:49 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Thu, 10 Feb 2005 14:05:38 -0600, Kristian Kielhofner [EMAIL PROTECTED] wrote: Derek Whitten wrote: I also call bullshit.. OpenBSD does NOT allow ssh root login by default.. why do you think that they have such an excellent security track record.. Derek, I am sorry to say, that in fact, OpenBSD does allow SSH root logins by default: http://www.openbsd.org/cgi-bin/cvsweb/~checkout~/src/usr.bin/ssh/sshd_config?rev=1.70content-type=text/plain BTW, OpenBSD's track record of security has nothing to do with whether they allow root logins by default or not. If the admin isn't wise enough to pick a -decent- root password they shouldn't be running a box connected to the internet. I think people need to start to provide HARD FACTS in some of these posts. I don't see what any of this has to do with Asterisk... -- Kristian Kielhofner Unless I'm missing something, the only line that is ENABLED in that file is this one: Subsystem sftp/usr/libexec/sftp-server The rest appear to be commented out with #, unless I'm not understanding how that all works... -- Dana Nevermind. I see how it is... Good thing I'm not a BSD admin. -- Dana ___ 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
RE: [Asterisk-Users] transferring a IAX call into a conference
I'm using an [EMAIL PROTECTED] installation. I dialed out on my packet8 service using a '9' And dialed back in my faktortel iax service. I have tried this with people dialing into my Faktortel service as well using my cell phone but same thing happens. asterisk1*CLI asterisk1*CLI asterisk1*CLI -- Executing Macro(SIP/30-e7e2, dialout|1|961283073503) in new stack -- Executing SetVar(SIP/30-e7e2, length=1) in new stack -- Executing Dial(SIP/30-e7e2, ZAP/g0/61283073503) in new stack -- Called g0/61283073503 -- Zap/1-1 answered SIP/30-e7e2 -- Accepting AUTHENTICATED call from 202.125.42.141, requested format = 256, actual format = 1024 -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/5, 0?from-pstn-reghours|s|1:) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/5, 0?from-pstn-afthours|s|1:) in new stack -- Executing GotoIfTime(IAX2/[EMAIL PROTECTED]/5, 5:55-23:59|*|*|*?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/5, 1?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours-nofax,s,1) -- Executing SetVar(IAX2/[EMAIL PROTECTED]/5, intype=GRP-700) in new stack -- Executing Cut(IAX2/[EMAIL PROTECTED]/5, intype=intype|-|1) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/5, 0?4:5) in new stack -- Goto (from-pstn-reghours-nofax,s,5) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/5, 1?6:7) in new stack -- Goto (from-pstn-reghours-nofax,s,6) -- Executing Goto(IAX2/[EMAIL PROTECTED]/5, ext-group|700|1) in new stack -- Goto (ext-group,700,1) -- Executing SetVar(IAX2/[EMAIL PROTECTED]/5, GROUP=30|32|33|) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/5, RINGTIMER=30) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/5, PRE=4357) in new stack -- Executing Macro(IAX2/[EMAIL PROTECTED]/5, rg-group) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/5, GRP=30|32|33|) in new stack -- Executing SetGroup(IAX2/[EMAIL PROTECTED]/5, ) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/5, FROMCONTEXT=rg-group) in new stack -- Executing SetCIDName(IAX2/[EMAIL PROTECTED]/5, 4357) in new stack -- Executing Macro(IAX2/[EMAIL PROTECTED]/5, dial|30|tr|30|32|33|) in new stack -- Executing AGI(IAX2/[EMAIL PROTECTED]/5, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: request = dialparties.agi -- dialparties.agi: priority = 1 -- dialparties.agi: extension = s -- dialparties.agi: language = en -- dialparties.agi: accountcode = -- dialparties.agi: uniqueid = 1108130997.18 -- dialparties.agi: channel = IAX2/[EMAIL PROTECTED]/5 -- dialparties.agi: callerid = 4357 -- dialparties.agi: context = macro-dial -- dialparties.agi: type = IAX2 -- dialparties.agi: rdnis = unknown -- dialparties.agi: enhanced = 0.0 -- dialparties.agi: dnid = unknown dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 30 to extension map -- dialparties.agi: Added extension 32 to extension map -- dialparties.agi: Added extension 33 to extension map -- dialparties.agi: Extension 33 cf is disabled -- dialparties.agi: Extension 32 cf is disabled -- dialparties.agi: Extension 30 cf is disabled -- dialparties.agi: Extension 33 do not disturb is disabled -- dialparties.agi: Extension 32 do not disturb is disabled -- dialparties.agi: Extension 30 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 33 has call waiting disabled dialparties.agi: Extension 32 has call waiting disabled dialparties.agi: Extension 30 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial dialparties.agi: Dial still has extensions - continuing -- dialparties.agi: DbDel CALLTRACE/33 - Caller ID is not defined -- dialparties.agi: DbDel CALLTRACE/32 - Caller ID is not defined dialparties.agi: About to execute Dial(IAX2/33SIP/32|30|tr) -- AGI Script Executing Application: (Dial) Options: (IAX2/33SIP/32|30|tr) -- Called 32 -- SIP/32-30dc is ringing -- SIP/32-30dc answered IAX2/[EMAIL PROTECTED]/5 -- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/5 -- Stopped music on hold on IAX2/[EMAIL PROTECTED]/5 dialparties.agi: Dial return value was -1 and dialstring was IAX2/33SIP/32|30|tr dialparties.agi: Setting Priority to 22 from 2 -- AGI Script dialparties.agi completed, returning 0 == Channel 'IAX2/[EMAIL PROTECTED]/5' jumping out of macro 'dial' == Channel 'IAX2/[EMAIL PROTECTED]/5' jumping out of macro 'rg-group' -- Executing Macro(IAX2/[EMAIL PROTECTED]/5, hangupcall) in new stack -- Executing ResetCDR(IAX2/[EMAIL
Re: [Asterisk-Users] chan_capi and asterisk
I don't know about your problem, but since you use mISDN, why not use the specific chan_mISDN? http://www.beronet.com/?PageID=3017 It's Free Software (GPL) Regards Marco Menardi btw, if you login in their bug tracker, the home page has alink to a document that tells you how install their boards, mISDN and, AFAIR, use their chan_mISDN with asterisk. Anabela Abreu wrote: Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36 WARNING[2535]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Feb 11 13:50:36 WARNING[2535]: loader.c:391 load_modules: Loading module chan_capi.so failed! my lsmod shows: Module Size Used by mISDN_capi 85312 0 kernelcapi 45088 1 mISDN_capi hfcpci 28716 0 mISDN_dsp 197248 0 l3udss132008 0 mISDN_l2 38272 0 mISDN_l1 10632 0 mISDN_core 77732 6 mISDN_capi,hfcpci,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 md5 4352 1 ipv6 235840 24 parport_pc 25024 1 lp 12396 0 parport42696 2 parport_pc,lp dm_mod 55444 0 uhci_hcd 31896 0 3c59x 36776 0 floppy 59568 0 ext3 116744 2 jbd74904 1 ext3 and my modules.conf : load = chan_capi.so [global] chan_capi.so=yes what seems to be the problem can someone help me? tahnk´s ___ 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 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] Re: chan_capi and asterisk
Try to check if you have /dev/capi20 ? If not, you can create it with: mknod /dev/capi20 c 68 0 chown root.dialout /dev/capi20 chmod 660 /dev/capi20 That worked for me on one instalation (Debian Sarge) that somehow finished without making /dev/capi20. Regards, Nenad Radosavljevic Message: 11 Date: Fri, 11 Feb 2005 13:55:30 + From: Anabela Abreu [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_capi and asterisk To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36 WARNING[2535]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Feb 11 13:50:36 WARNING[2535]: loader.c:391 load_modules: Loading module chan_capi.so failed! my lsmod shows: Module Size Used by mISDN_capi 85312 0 kernelcapi 45088 1 mISDN_capi hfcpci 28716 0 mISDN_dsp 197248 0 l3udss132008 0 mISDN_l2 38272 0 mISDN_l1 10632 0 mISDN_core 77732 6 mISDN_capi,hfcpci,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 md5 4352 1 ipv6 235840 24 parport_pc 25024 1 lp 12396 0 parport42696 2 parport_pc,lp dm_mod 55444 0 uhci_hcd 31896 0 3c59x 36776 0 floppy 59568 0 ext3 116744 2 jbd74904 1 ext3 and my modules.conf : load = chan_capi.so [global] chan_capi.so=yes what seems to be the problem can someone help me? tahnk´s ___ 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] Menu Selections Only Work Internally
All,
Funny problem. During my greating, the menu selections only work if
one calls from an internal sip line. The greating plays for all
including calls over the t1. But pressing 9 for directory or any other
mapped button will only work if I call from inside. If I arrive to the
menu from an outside line SIP or POTS pressing the button does
nothing. Any ideas?
extensions.conf
--
[MainMenu]
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(main-menu)
exten=_3XX,1,Goto(sip,${EXTEN},1)
exten=0,1,Goto(sip,301,1)
[sip]
;Main Number
exten = 300,1,Goto(MainMenu,s,1)
--
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[Asterisk-Users] Weird Echo Problem
Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), then it stop and come back all the way until the call is finish. It does the same thing on outgoing calls from my Cisco 7960 phone to the PSTN line. I have no problem when it's an internal call from one 7960 to another one. I tried a lot of different config in zapata.conf and the one that seems to work the best for now is this one : context=incoming signalling=fxs_ks echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=0 txgain=0 immediate=no busydetect=no callprogress-no musiconhold=default usecallerid=yes callerid=asreceived group=1 channel = 1-8 Any suggestion why it start doing echo after 5 minutes or so? Thanks Martin ___ 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
Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
On Fri, 11 Feb 2005, Peer Oliver Schmidt wrote: Remco Barende wrote: I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. [..] I have three different locations with HFC cards. I had the same stability problems on ALL of the installations. Since RC5 plus the florz patch *ALL* of the stability problems have vanished. No more seconds of silence, no more unavailability messages. It just works now. I won't touch the installations for a long time :-) Thanks. I did look in the wiki and the webpage of florz but thought that the patch was only for multi card installations, therefore I never applied it. Will try it tonight. Just out of interest, why was that patch never incorporated in bristuff? ___ 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
Re: [Asterisk-Users] chan_capi and asterisk
i was using chan_mISDN with asterisk and it works, but i trying to setup my isdn pci card with hylafax and i read that i add to use chan_capi. I don´t know if is possible to do this with chan_mISDN. Em Fri, 11 Feb 2005 15:12:31 +0100 Marco Menardi [EMAIL PROTECTED] escreveu: I don't know about your problem, but since you use mISDN, why not use the specific chan_mISDN? http://www.beronet.com/?PageID=3017 It's Free Software (GPL) Regards Marco Menardi btw, if you login in their bug tracker, the home page has alink to a document that tells you how install their boards, mISDN and, AFAIR, use their chan_mISDN with asterisk. Anabela Abreu wrote: Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36 WARNING[2535]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Feb 11 13:50:36 WARNING[2535]: loader.c:391 load_modules: Loading module chan_capi.so failed! my lsmod shows: Module Size Used by mISDN_capi 85312 0 kernelcapi 45088 1 mISDN_capi hfcpci 28716 0 mISDN_dsp 197248 0 l3udss132008 0 mISDN_l2 38272 0 mISDN_l1 10632 0 mISDN_core 77732 6 mISDN_capi,hfcpci,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 md5 4352 1 ipv6 235840 24 parport_pc 25024 1 lp 12396 0 parport42696 2 parport_pc,lp dm_mod 55444 0 uhci_hcd 31896 0 3c59x 36776 0 floppy 59568 0 ext3 116744 2 jbd74904 1 ext3 and my modules.conf : load = chan_capi.so [global] chan_capi.so=yes what seems to be the problem can someone help me? tahnk´s ___ 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 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 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
Re: [Asterisk-Users] asterisk@home scary log
On Thu, 10 Feb 2005 17:49:23 +, Clive Carter [EMAIL PROTECTED] wrote: I hesitated before sending this, as I have been flamed before for being a beginner. but I am newish to linux/asterisk, and I am running an ssh server. It is still running with default settings, (I dont know yet how/where to change it), and I CAN logon remotely as root. (Haven't figured out how to 'su' yet !) This is using the Rapid Xorcomm v 1.0 cd, which I believe (may be wrong) is based on a very recent version of Debian ? Perhaps xorcom have changed the default setting ? Hey Clive. I thought it was mentioned earlier before in the thread, but if not, all you need to do is edit your sshd_config file. In Debian, this is located at /etc/ssh/sshd_config, but it could be different for other distros. Open that up in a text editor and then locate the line that says PermitRootLogin yes, and change that to PermitRootLogin no. Save it, and then restart SSH. On Debian, you type in /etc/init.d/ssh restart, but on other distros it might be different. Note that you'll have to be root to edit that file and restart that service. -- Dana Thanks for that. I did not see it before, and I was afraid to ask in case I got jumped on again ! Thanks again -- -- Clive Email : [EMAIL PROTECTED] Tel : 08444844790 Alt : 08450043366 Fax : 08444844813 SIP : [EMAIL PROTECTED] Mobile : 07031945504 ___ 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
Re: [Asterisk-Users] chan_capi and asterisk
i try to do that and it didn´t work i continue to have the same problem. Em Fri, 11 Feb 2005 14:58:31 +0100 Stefan Gofferje [EMAIL PROTECTED] escreveu: Anabela Abreu schrieb: Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36 WARNING[2535]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Feb 11 13:50:36 WARNING[2535]: loader.c:391 load_modules: Loading module chan_capi.so failed! my lsmod shows: Module Size Used by mISDN_capi 85312 0 kernelcapi 45088 1 mISDN_capi hfcpci 28716 0 mISDN_dsp 197248 0 l3udss132008 0 mISDN_l2 38272 0 mISDN_l1 10632 0 mISDN_core 77732 6 mISDN_capi,hfcpci,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 md5 4352 1 ipv6 235840 24 parport_pc 25024 1 lp 12396 0 parport42696 2 parport_pc,lp dm_mod 55444 0 uhci_hcd 31896 0 3c59x 36776 0 floppy 59568 0 ext3 116744 2 jbd74904 1 ext3 AFAIK, chan_capi is for FritzCards with original AVM capi4linux only. Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ 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 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
Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
Remco Barende wrote: Since RC5 plus the florz patch *ALL* of the stability problems have vanished. No more seconds of silence, no more unavailability messages. It just works now. I won't touch the installations for a long time :-) Thanks. I did look in the wiki and the webpage of florz but thought that the patch was only for multi card installations, therefore I never applied it. Will try it tonight. Just out of interest, why was that patch never incorporated in bristuff? I *assume* it has to do with the license. Junghanns wants to keep all commercial rights to the driver. But maybe Kapejod or Florz are able to shed some light on the issue. -- Best regards Peer Oliver Schmidt ___ 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
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
Wait a sec, so let me get this straight... Previously if you checked out cvs 1.0.5 stable you would get the caller id bug and now there's a 1.0.5 stable that doesn't have the bug? Errr, someone please explain to me the versioning scheme being used here. Seems to me that if stable is released with a bug then the only way to change that is to issue a new release (like 1.0.6) without the bug. -mark On Feb 10, 2005, at 5:49 PM, Nicolás Gudiño wrote: Paul, 1.0.5 stable suffers from caller id issues as well, at least for SIP channels. What fixed things for me was swapping in app_dial.c from 1.0.2 stable (didn't try others). You could also just diff app_dial.c between versions to find the problem but I took the lazy way out the first time around. Drumkilla reverted the callerid changes on the latest stable (thanks Russell!). You will be fine if you checkout stable from CVS now. Regards, -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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
RE: [Asterisk-Users] Codec passthrough patch for IAX
Hmm... What's the status of this? This would allow me to declare one of my incoming DIDs a fax-number by forcing it to use ulaw. -Original Message- From: Michael Giagnocavo [mailto:[EMAIL PROTECTED] Sent: Thursday, February 10, 2005 5:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Codec passthrough patch for IAX Hi there, I had a problem, basically, I have 4 different types of end users (gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider. My provider supports all 4 codecs. The issue is then: When an incoming call comes in, a codec is negotiated (usually ULAW), later on, when the extension is dialed, we'll see we're doing GSM, and thus transcode. Here's an example dialplan: [incoming] exten = 123,1,Dial(IAX2/gsmUser) exten = 456,2,Dial(IAX2/ilbcUser) exten = 789,3,Dial(IAX2/g729User) You're pretty much forced to accept ULAW, and then transcode. Not fun if your provider does it for you (that's what you pay them for, right?). So, with this patch, just add a new config file. codec_passthrough.conf: [iax_my-did-provider] 123=gsm 456=ilbc 789=g729 Now, when an incoming call comes in, the user/extension will be found, and your preferred codec changed. No more transcoding. http://bugs.digium.com/bug_view_page.php?bug_id=0003553 My main question is: Can this be done without this patch? I've heard it's impossible, and it sure seems that way. Any suggestions? ___ 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
RE: [Asterisk-Users] Menu Selections Only Work Internally
Does your incoming context include the MainMenu?
-Original Message-
From: Philip Siegrist [mailto:[EMAIL PROTECTED]
Sent: Friday, February 11, 2005 8:17 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Menu Selections Only Work Internally
All,
Funny problem. During my greating, the menu selections only
work if one calls from an internal sip line. The greating
plays for all including calls over the t1. But pressing 9 for
directory or any other mapped button will only work if I call
from inside. If I arrive to the menu from an outside line SIP
or POTS pressing the button does nothing. Any ideas?
extensions.conf
--
[MainMenu]
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(main-menu)
exten=_3XX,1,Goto(sip,${EXTEN},1)
exten=0,1,Goto(sip,301,1)
[sip]
;Main Number
exten = 300,1,Goto(MainMenu,s,1)
--
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Re: [Asterisk-Users] Menu Selections Only Work Internally
On Fri, 2005-02-11 at 09:16 -0500, Philip Siegrist wrote:
All,
Funny problem. During my greating, the menu selections only work if
one calls from an internal sip line. The greating plays for all
including calls over the t1. But pressing 9 for directory or any other
mapped button will only work if I call from inside. If I arrive to the
menu from an outside line SIP or POTS pressing the button does
nothing. Any ideas?
extensions.conf
--
[MainMenu]
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(main-menu)
exten=_3XX,1,Goto(sip,${EXTEN},1)
exten=0,1,Goto(sip,301,1)
[sip]
;Main Number
exten = 300,1,Goto(MainMenu,s,1)
--
It is hard to say without knowing what context your incoming calls are
getting dumped into. Usually your external and internal calls should be
dropped into different contexts and will include different contexts into
them. My guess is you need to rethink what you have done above. Most of
the time, Gotos are not needed unless you are working within a single
extension, or if you are transferring to a different context or a
different extension than was typed.
--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Asterisk not accepting multiple SIP phone logins
Remember that SIP uses DNS SRV entries, maybe one of the phones you use efectively use the DNS SRV entry and the other not. Some VoIP phones have a flag where you can deactivate this functionality for SIP. If not, make sure you have in your local DNS a SRV entry for SIP. Hope this helps. Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Juki Sent: Thursday, February 10, 2005 11:08 PM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [Asterisk-Users] Asterisk not accepting multiple SIP phone logins Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201 callerid=101 2125 nat=yes My extensions.conf has: exten = 101,1,Dial(SIP/101,20,tr) exten = 101,2,VoiceMail,u101 exten = 101,102,VoiceMail,b101 My voicemail.conf has: 101 = 2348,Emma, [EMAIL PROTECTED] Any ideas are most welcome. -- Rgds, Juki ___ 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 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
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
Mark Eissler wrote: Wait a sec, so let me get this straight... Previously if you checked out cvs 1.0.5 stable you would get the caller id bug and now there's a 1.0.5 stable that doesn't have the bug? Errr, someone please explain to me the versioning scheme being used here. Seems to me that if stable is released with a bug then the only way to change that is to issue a new release (like 1.0.6) without the bug. -mark Anything that comes out of CVS always incorporates recent changes. If you check out 1.0.5 from CVS right now, you'll get 1.0.5 + the changes that will be part of 1.0.6. The are two ways to get 1.0.5 as it was released back on XYZ date, first is to download one of the 1.0.5 tarballs, the other is to download the CVS 1.0.5 telling CVS to pull the source AS OF that XYZ date. It basically works out that CVS *is always a moving target*, tarballs are the only things that don't change. Make sense? -Chris ___ 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] chan_misdn and hylafax
Was anyone put hylafax working with chan_misdn? ___ 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] Asterisk as a UAC forwarded by SER
Hi everybody, I have a SER Server (Sip Proxy / REGISTRAR) and a Asterisk Server (PSTN and other services). I've got some clients that make calls to each other through my SER Server, that's to say, non external or international calls. I would like my clients to make external and international calls through my server but for that they must authenticate at another server to have a valid VoIP phone number. I want them to authenticate at Go2Call VoIP Server to make internatinoal and external calls, but for that i need my ser to forward every call begginning with 1 to Asterisk and Asterisk to auth at Go2Call Server sending username and password (Funcionality that ser doesn't have) ending the VoIP tunnel and making the call complete. My SER Server is working perfectly, Asterisk is Receiving Ser forwards normally but it can't forward to Go2Call, that is the configuration i'm in need of. I've read many articles and books about the subject at www.voip-info.org but I haven't make up my mind on how to configure Asterisk to do so. Could anyone who have done that give me a hand ? Thanks in Advance. ___ 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
Re: [Asterisk-Users] dtmfmode and IAX protocol
There's enough information if he's using FWD's 8XX-gateway for his toll-free calls to UPS, the bank, etc. First of all, if the wiki says inline (yes, okay, it does) it probably means as inline data as opposed to inband. But the fact is that iax2 ALWAYS sends DTMF out-of-band. While it's true that some VOIP phones (SIP adapters, etc) can be configured to send DTMF inband, I would think that doing so while using IAX is going to result in digits being reproduced twice at the destination--once from the DTMF reproduced from out-of-band and once from the inband DTMF. So if you're using IAX as your trunking protocal you need to use out-of-band DTMF on your IP phones (and adapters) as well. Why does DTMF work sometimes and not all of the time? Heck, if I knew that then I wouldn't have this problem where inband DTMF hardly ever works properly for any of my inbound calls over IAX from Voicepulse. I'm starting to think that Asterisk's support for DTMF over IAX has issues but I'm too stubborn to switch to SIP and test that. I know I don't have any (zero, nada, keine, rien, etc.) problems navigating Asterisk IVR menus via my SIP adapter. It's important to keep in mind, however, that the telco environment beyond your Asterisk box, beyond FWD (Voicepulse, Broadvoice, Vonage, etc.) is a complicated environment where everyone isn't playing by the same set of technical specs. The fact that any of this stuff actually works as well as it does is just amazing in itself. -mark On Feb 11, 2005, at 1:17 AM, Rich Adamson wrote: Joseph has been working at bringing up an asterisk box as kind of a newbie, and I think he's using a Sipura as his fxs interface into asterisk. He's having a problem with asterisk passing dtmf to FWD, but didn't say how he's accessing the bank or fedex. So, without a fair amount more detail from him, there's no way to answer his questions or guess at the problem. Exactly. (I was hoping he'd come to his own conclusions.) So... if the Sipura does not do IAX, then it's quite possible that you're not doing IAX on the Sipura. Which means the whole dtmfmode and IAX protocol is moot... -Michael - No. Can the Sipura SPA-3000 do IAX? -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Thursday, February 10, 2005 10:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] dtmfmode and IAX protocol Actually, I don't know what might be the problem. I'm using Sipura SPA-3000 unit connected to standard cordless phone and connecting to FWD over IAX 1.) If I call FedEx or Bank and enter my account number using numeric keys it works 2.) If I dial UPS 1-800-742-5877 and try to use one of the option provided it doesn't work. Could it be their phone system? -- #Joseph On Thu, 2005-02-10 at 21:36 -0600, Michael Giagnocavo wrote: Actually, there are some phones that will do inband DTMF over IAX2. So if he's using one of these, he has to make sure his settings are correct. Yes, the PA168 phones. The correct setting is RFC2833 for IAX (inside these phones). Otherwise it's inband. The other options they provide just cut the call. -Michael ___ 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 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 ---End of Original Message- ___ 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 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 ---End of Original Message- ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
Re: [Asterisk-Users] Multiple SIP registrations for one account?
Philipp von Klitzing wrote: U sure? What for example if later on the SIP device forwards the call (note: not using #) and itself steps out of the line? That has nothing to with canreinvite=yes. Setting canreinvite=no does _not_ in any way restrict the ability of the SIP peer to redirect/forward or otherwise manipulate the call. It has only one effect: it stops Asterisk from trying to send the media directly to that peer. Realistically, the name of this option is completely and utterly wrong. I've suggested changing it, but been told no, it's OK. In my mind, it's not OK, but oh well... ___ 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
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
Chris Wade wrote: Anything that comes out of CVS always incorporates recent changes. snip It basically works out that CVS *is always a moving target*, tarballs are the only things that don't change. Make sense? Please be aware, these statements are intentional generalities, there are exceptions to every rule. -Chris ___ 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
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
Chris Wade wrote: If you check out 1.0.5 from CVS right now, you'll get 1.0.5 + the changes that will be part of 1.0.6. The are two ways to get 1.0.5 as it was released back on XYZ date, first is to download one of the 1.0.5 tarballs, the other is to download the CVS 1.0.5 telling CVS to pull the source AS OF that XYZ date. No, that is _not_ correct. If you do a CVS checkout and specify v1-0, you will get the _current 1.0 branch_. This means you will get the latest 1.0 stable official release, plus any changes that have been made since then but not released yet. If you do a CVS checkout and specify v1-0-5, you will _always_ get 1.0.5, regardless of any changes that have been made since then. There is never any need to tell CVS to pull by date unless you are trying to track down a particular problem; that is why the CVS tree is tagged with these branch/tag identifiers. ___ 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] RE:mandrake linux install of zaptel
Extreme N00b, I am getting the error message a target does not exist when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, February 11, 2005 1:37 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 7, Issue 168 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Searchable Mailing Lists NooB Question (Rich Adamson) 2. RE: TelIAX troubles (Scott Bussinger) 3. Re: Asterisk not acceptingmultiple SIP phone logins (Juki) 4. Re: Asterisk not accepting multiple SIP phone logins (Juki) 5. RE: dtmfmode and IAX protocol (Rich Adamson) 6. RE: dtmfmode and IAX protocol (Michael Giagnocavo) 7. Re: Why echo occurs (Rich Adamson) 8. RE: dtmfmode and IAX protocol (Rich Adamson) 9. Re: Why echo occurs (Steven Critchfield) 10. RE: Searchable Mailing Lists NooB Question (Ed Guy) 11. Re: Why echo occurs (Steve Underwood) 12. Re: Why echo occurs (Steven Critchfield) 13. RE: Zombie SIP channels (Florian Overkamp) 14. Re: Why echo occurs (Steve Underwood) -- Message: 1 Date: Thu, 10 Feb 2005 23:32:01 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Searchable Mailing Lists NooB Question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 Looks like your numbers add the transmit and receive data rates together, which is not a realistic way to discuss bandwidth consumption. An IAX link consumes about 22kb/s (round it to 30kb/s, who cares) in the transmit direction, and another 22kb/s in the receive direction. (There's your 60kb/s.) When comparing my numbers to things like 256,000 bits/sec of DSL bandwidth, you truly are comparing apples to apples. So, if you could orchestrate all IAX calls to be just exactly perfect across the 256,000 bits/sec DSL bandwidth, that DSL circuit could supposedly handle about eight simultanous gsm calls (256,000 divided by 30,000). However, there are lots of other real world issues that would preclude it from actually supporting anything close to eight calls. Four to six might be realistic if nothing else is using the DSL circuit. Very rough numbers: iax-gsm consumes about 22kb/s, I see about 60kb/s g711 about 80kb/s on I see 155kb/s Is that normal? This is an IAX link to voicepulse. I see all these lower numbers posted around but fail to see that on my connections. Using G711, Its only possible to have one connection at anytime, do to my upload capped at 256kb/s. So I use GSM, sounds fine anyway. Just wondering about the numbers. Dan same link unless you can set up QoS, etc. Lots of good info on the wiki ( www.voip-info.org ) for reference. ___ 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 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 ---End of Original Message- -- Message: 2 Date: Thu, 10 Feb 2005 21:44:05 -0800 From: Scott Bussinger [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] TelIAX troubles To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii We're just getting our Asterisk server setup with TelIAX and it's working fine. I did have to play with settings a bit. Basically I just used the setting they recommended instead of the generic settings I started with. Here are the significant settings we're using in IAX.CONF: [general] disallow=all allow=gsm register=username:[EMAIL PROTECTED] [teliax] type=friend context=tollfree host=voip.teliax.com auth=md5 secret=password Good Luck! --
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
I'm confused about the part that you can check out a stable version after 1.0.5. IMHO if you check out what is tagged as 1.0.5 at any time then you should get exactly what is in the 1.0.5 tarball. If you check out head then you should get all of the latest stuff in CVS which may or may not build cleanly (and may segfault or whatever). If you could check out 1.0.6-rc1 (release candidate 1) or something like that you would get everything after 1.0.5 that may or may not build properly but is no longer a moving target (features have been frozen). It just doesn't make sense to me that there would be a 1.0.5 that has changed since 1.0.5 was released unless you tag it 1.0.5.1 (or something). I mean, why even bother trying to constantly maintain a new stable version without having a formal release? 1.0.5 is what it is with whatever bugs it came with upon release. Obviously, just my opinion on How things should work!. ;-) -mark On Feb 11, 2005, at 10:06 AM, Chris Wade wrote: It basically works out that CVS *is always a moving target*, tarballs are the only things that don't change. Make sense? -Chris -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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
Re: [Asterisk-Users] Searchable Mailing Lists NooB Question
On Thu, 2005-02-10 at 21:44 -0600, Steven Critchfield wrote: So you probably want to still turn off the webserver and jabber server, they would be better off coloed anyways and there are a lot of cheap colo places for non critical hosting. As a sidenote, you can also set up traffic shaping to prioritize particular traffic/ports. I.e. if it's OK for you to starve web and jabber clients during voip calls, you can still run those servers without impairing your voice streams. Regards, Bruno. ___ 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
RE: [Asterisk-Users] Codec passthrough patch for IAX
I have it running on my server now for a day and seems to be working fine. -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, February 11, 2005 9:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Codec passthrough patch for IAX Hmm... What's the status of this? This would allow me to declare one of my incoming DIDs a fax-number by forcing it to use ulaw. -Original Message- From: Michael Giagnocavo [mailto:[EMAIL PROTECTED] Sent: Thursday, February 10, 2005 5:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Codec passthrough patch for IAX Hi there, I had a problem, basically, I have 4 different types of end users (gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider. My provider supports all 4 codecs. The issue is then: When an incoming call comes in, a codec is negotiated (usually ULAW), later on, when the extension is dialed, we'll see we're doing GSM, and thus transcode. Here's an example dialplan: [incoming] exten = 123,1,Dial(IAX2/gsmUser) exten = 456,2,Dial(IAX2/ilbcUser) exten = 789,3,Dial(IAX2/g729User) You're pretty much forced to accept ULAW, and then transcode. Not fun if your provider does it for you (that's what you pay them for, right?). So, with this patch, just add a new config file. codec_passthrough.conf: [iax_my-did-provider] 123=gsm 456=ilbc 789=g729 Now, when an incoming call comes in, the user/extension will be found, and your preferred codec changed. No more transcoding. http://bugs.digium.com/bug_view_page.php?bug_id=0003553 My main question is: Can this be done without this patch? I've heard it's impossible, and it sure seems that way. Any suggestions? ___ 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 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
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
Kevin P. Fleming wrote: Chris Wade wrote: If you check out 1.0.5 from CVS right now, you'll get 1.0.5 + the changes that will be part of 1.0.6. The are two ways to get 1.0.5 as it was released back on XYZ date, first is to download one of the 1.0.5 tarballs, the other is to download the CVS 1.0.5 telling CVS to pull the source AS OF that XYZ date. No, that is _not_ correct. If you do a CVS checkout and specify v1-0, you will get the _current 1.0 branch_. This means you will get the latest 1.0 stable official release, plus any changes that have been made since then but not released yet. If you do a CVS checkout and specify v1-0-5, you will _always_ get 1.0.5, regardless of any changes that have been made since then. There is never any need to tell CVS to pull by date unless you are trying to track down a particular problem; that is why the CVS tree is tagged with these branch/tag identifiers. I stand corrected, sorry for the mis-information. -Chris ___ 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
Re: [Asterisk-Users] asterisk@home scary log
Sorry about this. The new verison of [EMAIL PROTECTED] has a message in the install docs warning users to set thier root passwords. --- Jean-Louis curty [EMAIL PROTECTED] wrote: Hi everybody, I'm testing [EMAIL PROTECTED] 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from [EMAIL PROTECTED] at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw this Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) Feb 9 20:30:07 asterisk1 sendmail[10077]: j1A1U7CY010077: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30068, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7Ns010089 Message accepted for delivery) Feb 9 20:30:17 asterisk1 sendmail[10094]: j1A1U7Ns010089: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30348, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? thanks jl ___ 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 __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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
Re: [Asterisk-Users] Menu Selections Only Work Internally
yes. it get's to the Menu prompt which is defined under [MainMenu].
The input buttons simply do not work.
On Fri, 11 Feb 2005 09:06:26 -0600, Jay Milk [EMAIL PROTECTED] wrote:
Does your incoming context include the MainMenu?
-Original Message-
From: Philip Siegrist [mailto:[EMAIL PROTECTED]
Sent: Friday, February 11, 2005 8:17 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Menu Selections Only Work Internally
All,
Funny problem. During my greating, the menu selections only
work if one calls from an internal sip line. The greating
plays for all including calls over the t1. But pressing 9 for
directory or any other mapped button will only work if I call
from inside. If I arrive to the menu from an outside line SIP
or POTS pressing the button does nothing. Any ideas?
extensions.conf
--
[MainMenu]
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(main-menu)
exten=_3XX,1,Goto(sip,${EXTEN},1)
exten=0,1,Goto(sip,301,1)
[sip]
;Main Number
exten = 300,1,Goto(MainMenu,s,1)
--
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RE: [Asterisk-Users] No dialtone in a E1
Thank you Peter, how can I add the options to Dial to generate ringback??? do you have an example??? By the way, it is a PRI E1, with 30 bchannels and 1 dchannel. For a little background, I'm intending to replace my actual PBX with Asterisk, and everything is just working fine, until yesterday when I realized that when a call was made from some external lines, this lines didn't receive a dialtone. For this reason, I began to make some exhaustive test cases, and began to make calls from distinct providers to my E1. In all this testing I received a dialtone, except for a GSM cellular phone from a specific Telco. I tested some others GSM cellulars from the same Telco, and got always the same functionality, they didn't receive a dialtone. I think that if Asterisk can generate a ringback, this is going to solve all my problems with this little issue. Thank you in advance Peter for your help. Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Peter Svensson Sent: Thursday, February 10, 2005 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No dialtone in a E1 On Thu, 10 Feb 2005, Marco Castillo wrote: Hi, I'm having a little problem when trying to make a call from asterisk. I connect a SIP phone to asterisk, and in the asterisk box I have a TE110P card connected to a E1. When a SIP client makes a call through the E1, I received no dialtone in the SIP client. In the same manner, when somebody from the POTS network makes a call to a SIP client (through * and the E1) he doesn't receive the apropiate tone of call progress. Does anyone has some ideas about this? Are you talking about an ISDN E1 or another form of E1? On isdn dialtone is an optional feature of the specification and there are many implementations of isdn. I think it is mandatory on EuroISDN. Since asterisk normally generates the dialtone itself there should be little nead for the dialtone from the pstn. We use the dialtone from the network ourselves, but asterisk could provide it as well. In band call progress is also a feature of the net on isdn. If the net does not provide it you will have to do so yourself. Just add the proper options to Dial to generate ringback and if the call fails you generate the matching sound (Busy etc). Peter ___ 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 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
Re: [Asterisk-Users] Re: asterisk@home scary log {Scanned}
[EMAIL PROTECTED] uses the CentOS default sendmail config that does not forward mail. --- David Shaw [EMAIL PROTECTED] wrote: Cat your maillog. Grep out the msg ID. cat /var/log/maillog | grep j1A1U7Q1010071 j1A1U7Q1010071 is the [EMAIL PROTECTED] j1A1U7mf010088 is email from root to??? Have you checked root's email?? Your might want to edit /etc/aliases and forward root: [EMAIL PROTECTED] Also check sendmail deamon ports. cat /etc/mail/sendmail.cf | grep DaemonPortOptions This mains only 127.0.0.1 can relay. O DaemonPortOptions=Port=smtp,Addr=127.0.0.1, Name=MTA Good luck, David On Thu, 2005-02-10 at 17:53 +0100, Bruno Hertz wrote: On Thu, 2005-02-10 at 11:09 -0500, Jason Stewart wrote: There's a chance that you may have been hacked, but the logs you post look more like your mailserver is an open relay. You sure? I run postfix myself and am not proficient in analyzing sendmail logs, but looking at those lines Feb 9 20:30:07 asterisk1 sendmail[10088]: j1A1U7mf010088: from=[EMAIL PROTECTED], size=329, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=asterisk1.local [127.0.0.1] Feb 9 20:30:07 asterisk1 sendmail[10071]: j1A1U7Q1010071: [EMAIL PROTECTED], ctladdr=root (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=30049, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (j1A1U7mf010088 Message accepted for delivery) I find the relay (accepting host) is 127.0.0.1. So, even if ignoring the envelope 'from', there seems to be no doubt which host this mail was sent from. Regards, Bruno. ___ 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 -- David Shaw [EMAIL PROTECTED] ___ 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 __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ 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
Re: [Asterisk-Users] Menu Selections Only Work Internally
Sounds like maybe the wrong DTMF setting ?
On Fri, 11 Feb 2005 10:57:38 -0500, Philip Siegrist [EMAIL PROTECTED] wrote:
yes. it get's to the Menu prompt which is defined under [MainMenu].
The input buttons simply do not work.
On Fri, 11 Feb 2005 09:06:26 -0600, Jay Milk [EMAIL PROTECTED] wrote:
Does your incoming context include the MainMenu?
-Original Message-
From: Philip Siegrist [mailto:[EMAIL PROTECTED]
Sent: Friday, February 11, 2005 8:17 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Menu Selections Only Work Internally
All,
Funny problem. During my greating, the menu selections only
work if one calls from an internal sip line. The greating
plays for all including calls over the t1. But pressing 9 for
directory or any other mapped button will only work if I call
from inside. If I arrive to the menu from an outside line SIP
or POTS pressing the button does nothing. Any ideas?
extensions.conf
--
[MainMenu]
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(main-menu)
exten=_3XX,1,Goto(sip,${EXTEN},1)
exten=0,1,Goto(sip,301,1)
[sip]
;Main Number
exten = 300,1,Goto(MainMenu,s,1)
--
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[Asterisk-Users] Question about DID
Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk it passing the DID and CallerID info to the server? I seen this was done with HylaFax. Any help would be great!! Here is my configs cat zaptel.conf#PRI to Telco span=1,1,0,esf,b8zsbchan=1-23dchan=24 # PRI to Fax serverspan=2,0,0,esf,b8zsbchan=25-47dchan=48 zapata.conf[channels]context=from-analogsignalling=pri_cpeswitchtype=dms100group=1usecallerid=yeshidecallerid=norestrictcid=nousecallingpres=nouseincomingcalleridonzaptransfer=yescallerid=asreceivedfaxdetect=nomusiconhold=defaultchannel = 1-23 context=from-sip-internalswitchtype=dms100signalling=pri_netgroup=2overlapdial=yesusecallerid=yeshidecallerid=norestrictcid=nousecallingpres=nouseincomingcalleridonzaptransfer=yescallerid=asreceivedfaxdetect=nomusiconhold=default channel = 25-47 ___ 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
RE: [Asterisk-Users] Echo Cancellation
We use a product from oriontelecom.com. The interface is rough, but we have not had a single problem since putting this in. Stephen Dolloff DLS Internet Services 847-854-4799 x256 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Cook Sent: Wednesday, February 09, 2005 5:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Echo Cancellation Can anyone provide a good manufacturer of echo cancellation equipment for a PRI? -- Richard Cook [EMAIL PROTECTED] T: 705-497-9320 x2010 image001.gif___ 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
Re: [Asterisk-Users] Caller ID
http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
http://voip-info.org/wiki-Asterisk+cmd+SetCIDName
http://voip-info.org/wiki-Asterisk+cmd+SetCIDNum
example:
exten = 1,1,SetCallerID(${CALLERID})
or
exten = 1,1,SetCallerID(Your Name (555)555-)
On Fri, 2005-02-11 at 05:45, Martin Roy wrote:
How can I change that when there's no Caller ID instead of displaying
asterisk it display something like Unknown. Because everyone is confuse
when they see a call coming from asterisk.
Thanks
Martin
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--
Derek Whitten [EMAIL PROTECTED]
kFuQ Productions
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RE: [Asterisk-Users] Menu Selections Only Work Internally
In your [mainmenu] use the include = context_for_internal_numbers, or at
least the ones you want peaple to call.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philip
Siegrist
Sent: 11 February 2005 15:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Menu Selections Only Work Internally
yes. it get's to the Menu prompt which is defined under [MainMenu].
The input buttons simply do not work.
On Fri, 11 Feb 2005 09:06:26 -0600, Jay Milk [EMAIL PROTECTED] wrote:
Does your incoming context include the MainMenu?
-Original Message-
From: Philip Siegrist [mailto:[EMAIL PROTECTED]
Sent: Friday, February 11, 2005 8:17 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Menu Selections Only Work Internally
All,
Funny problem. During my greating, the menu selections only
work if one calls from an internal sip line. The greating
plays for all including calls over the t1. But pressing 9 for
directory or any other mapped button will only work if I call
from inside. If I arrive to the menu from an outside line SIP
or POTS pressing the button does nothing. Any ideas?
extensions.conf
--
[MainMenu]
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(main-menu)
exten=_3XX,1,Goto(sip,${EXTEN},1)
exten=0,1,Goto(sip,301,1)
[sip]
;Main Number
exten = 300,1,Goto(MainMenu,s,1)
--
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[Asterisk-Users] Asterisk-MySQL: Not loading voicemail config from MySQL
Folks,
I'm trying to get Asterisk to load my voicemail
configuration from MySQL. I've followed the
instructions at:
http://www.voip-info.org/wiki-Asterisk+voicemail+database
I restarted Asterisk, but no luck: the voicemail.conf
does not get updated. I started with a sample
voicemail.conf that I found on the Wiki. Or was it
from Voicepulse? I can't remember. For initial
testing, I added extensions and 100 in the
[voicepulse_connect_context] with appropriate settings
in extensions.conf to direct incoming calls to those
mailboxes, and that works. I was expecting that after
I added in the db details, reloading or restarting
Asterisk would add the new extension from MySQL's
'users' table into the voicemail.conf. It doesn't.
As soon as I type (the beginning of the mailbox
and also the extension number), I get the message:
*CLI Feb 11 08:38:38 WARNING[5224]:
app_voicemail.c:1539 leave_voicemail: No entry in
voicemail config file for ''
If I add a line for into my voicemail.conf, all
works well.
Please help me understand what is going on here!
Thanks,
Maya.
--- My configuration ---
My 'users' table has 1 row only, for testing purposes:
+---++--+--+---+---+++
| context | mailbox| password |
fullname | email | pager | options
| stamp |
+---++--+--+---+---+++
| voicemail_connect_context | | 1234 |
Moron Tester | [EMAIL PROTECTED] | | attach=yes
| 00 |
+---++--+--+---+---+++
--
The appropriate settings from extensions.conf:
[voicepulse_connect_context] ; -- Should match the
context you have
; under [voicepulse-in-01] in
iax.conf
exten = 100,1,Playback(tt-monkeys)
exten = 100,2,Record(/tmp/asterisk-recording:gsm)
;exten = 100,3,Wait(2)
exten = 100,3,Playback(/tmp/asterisk-recording)
;exten = 100,5,Wait(2)
exten = ,1,Playback(transfer,skip)
exten = ,2,VoiceMail,u
exten = ,102,VoiceMail,b
exten = ,1,VoiceMail,u
--
My complete voicemail.conf looks like this:
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
format=wav49|gsm|wav
; Who the e-mail notification should appear to come
from
[EMAIL PROTECTED]
; Should the email contain the voicemail as an
attachment
attach=yes
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Minimum length of a voicemail message in seconds
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
; How many miliseconds to skip forward/back when
rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the
recording
maxsilence=10
; Silence threshold (what we consider silence, the
lower, the more sensitive)
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail is left, delivered, or your
voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail password is changed,
; uncomment this:
;externpass=/usr/bin/myapp
; For the directory, you can override the intro file
if you want
;directoryintro=dir-intro
; The character set for voicemail messages can be
specified here
;charset=ISO-8859-1
; The ADSI feature descriptor number to download to
;adsifdn=000F
; The ADSI security lock code
;adsisec=9BDBF7AC
; The ADSI voicemail application version number.
;adsiver=1
; Skip the [PBX]: string from the message title
pbxskip=yes
; Change the From: string
fromstring=The Asterisk PBX
;
;Change the From: string for pager messages
;pagerfromstring=The Asterisk PBX
;
; Change the email body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX,
VM_CALLERID, VM_DATE
;
; Note: The emailbody config row can be up to 512
characters due to a limitation in
; asterisk config files.
;emailsubject=New VM (${VM_MSGNUM}) - ${VM_DUR} long
in mailbox ${VM_MAILBOX} from ${VM_CALLERID}
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you
know you were just left a ${VM_DUR} long message
(number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from
${VM_CALLERID}, on ${VM_DATE} so you might\nwant to
check it when you get a chance. Thanks!
;
; You can override the default program to send e-mail
if you wish, too
;
;mailcmd=/usr/sbin/sendmail -t
;
; Users may be located in different timezones, or may
have different
; message announcements for their introductory message
when they enter
; the voicemail system. Set the message and the
timezone each user
; hears here. Set the user into one of these zones
with the tz= attribute
; in
Re: [Asterisk-Users] RE:mandrake linux install of zaptel
On Feb 11, 2005, at 16:28, [EMAIL PROTECTED] wrote: Extreme N00b, I am getting the error message a target does not exist when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. I think everyone would appreciate if... - you wrote a new mail instead of highjacking an existing thread by answering it and replacing the subject line - you would not keep 5 miles of completely unrelated stuff in your email message - you could provide a better problem description that includes specific error messages and message stacks. Thanks! jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about DID
How are you telling Asterisk to send the call to the fax group? You
should have something in extensions.conf like
exten = _4135551234,1,Dial($FAXTRUNKS/${EXTEN})
Asterisk should send the EXTEN down as a DID to the fax server
-Matt
On Feb 11, 2005, at 11:05 AM, Eric Hall wrote:
Hello Group
I have a Asterisk server running with 2 Digium T1 cards installed. 1
card connects to Telco via a PRI. The 2nd card is connected to a fax
server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to
have Asterisk route the calls based on DID or FAX tones. Everything is
working great so far. The only problem is the Fax server does not see
the DID. How can I tell if Asterisk it passing the DID and CallerID
info to the server? I seen this was done with HylaFax.
Any help would be great!!
Here is my configs
cat zaptel.conf
#PRI to Telco
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
# PRI to Fax server
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
zapata.conf
[channels]
context=from-analog
signalling=pri_cpe
switchtype=dms100
group=1
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=no
musiconhold=default
channel = 1-23
context=from-sip-internal
switchtype=dms100
signalling=pri_net
group=2
overlapdial=yes
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=no
musiconhold=default
channel = 25-47
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Re: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config fromMySQL
What version of asterisk?
-Matthew
- Original Message -
From: beonice [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 11, 2005 10:36 AM
Subject: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config
fromMySQL
Folks,
I'm trying to get Asterisk to load my voicemail
configuration from MySQL. I've followed the
instructions at:
http://www.voip-info.org/wiki-Asterisk+voicemail+database
I restarted Asterisk, but no luck: the voicemail.conf
does not get updated. I started with a sample
voicemail.conf that I found on the Wiki. Or was it
from Voicepulse? I can't remember. For initial
testing, I added extensions and 100 in the
[voicepulse_connect_context] with appropriate settings
in extensions.conf to direct incoming calls to those
mailboxes, and that works. I was expecting that after
I added in the db details, reloading or restarting
Asterisk would add the new extension from MySQL's
'users' table into the voicemail.conf. It doesn't.
As soon as I type (the beginning of the mailbox
and also the extension number), I get the message:
*CLI Feb 11 08:38:38 WARNING[5224]:
app_voicemail.c:1539 leave_voicemail: No entry in
voicemail config file for ''
If I add a line for into my voicemail.conf, all
works well.
Please help me understand what is going on here!
Thanks,
Maya.
--- My configuration ---
My 'users' table has 1 row only, for testing purposes:
+---++--+--+
---+---+++
| context | mailbox| password |
fullname | email | pager | options
| stamp |
+---++--+--+
---+---+++
| voicemail_connect_context | | 1234 |
Moron Tester | [EMAIL PROTECTED] | | attach=yes
| 00 |
+---++--+--+
---+---+++
--
The appropriate settings from extensions.conf:
[voicepulse_connect_context] ; -- Should match the
context you have
; under [voicepulse-in-01] in
iax.conf
exten = 100,1,Playback(tt-monkeys)
exten = 100,2,Record(/tmp/asterisk-recording:gsm)
;exten = 100,3,Wait(2)
exten = 100,3,Playback(/tmp/asterisk-recording)
;exten = 100,5,Wait(2)
exten = ,1,Playback(transfer,skip)
exten = ,2,VoiceMail,u
exten = ,102,VoiceMail,b
exten = ,1,VoiceMail,u
--
My complete voicemail.conf looks like this:
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
format=wav49|gsm|wav
; Who the e-mail notification should appear to come
from
[EMAIL PROTECTED]
; Should the email contain the voicemail as an
attachment
attach=yes
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Minimum length of a voicemail message in seconds
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
; How many miliseconds to skip forward/back when
rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the
recording
maxsilence=10
; Silence threshold (what we consider silence, the
lower, the more sensitive)
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail is left, delivered, or your
voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail password is changed,
; uncomment this:
;externpass=/usr/bin/myapp
; For the directory, you can override the intro file
if you want
;directoryintro=dir-intro
; The character set for voicemail messages can be
specified here
;charset=ISO-8859-1
; The ADSI feature descriptor number to download to
;adsifdn=000F
; The ADSI security lock code
;adsisec=9BDBF7AC
; The ADSI voicemail application version number.
;adsiver=1
; Skip the [PBX]: string from the message title
pbxskip=yes
; Change the From: string
fromstring=The Asterisk PBX
;
;Change the From: string for pager messages
;pagerfromstring=The Asterisk PBX
;
; Change the email body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX,
VM_CALLERID, VM_DATE
;
; Note: The emailbody config row can be up to 512
characters due to a limitation in
; asterisk config files.
;emailsubject=New VM (${VM_MSGNUM}) - ${VM_DUR} long
in mailbox ${VM_MAILBOX} from ${VM_CALLERID}
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you
know you were just left a ${VM_DUR} long message
(number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from
${VM_CALLERID}, on ${VM_DATE} so you
Re: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config fromMySQL
--- Matthew Boehm [EMAIL PROTECTED] wrote: What version of asterisk? -Matthew Asterisk CVS-v1-0-12/12/04-15:58:29 built by [EMAIL PROTECTED] on a i686 running WhiteBox Enterprise Linux By the way, I _have_ created an ast_config db and the content of my ast_config table is: ++++---++--+--+-+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+-+ | 1 | 0 | 0 | 0 | voicemail.conf | default | | | ++++---++--+--+-+ I've also created etc/asterisk/configs/res_odbc.conf as described in: http://voip-info.org/wiki-Asterisk+res_config My extconfig.conf says: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf = odbc voicemail.conf = odbc Unfortunately, I'm not sure what values to put in for [mysql1] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes and for [ENV] VAR=VALUE I suspect this MAY be the problem. :) I'm unable to guess what to substitute as an appropriate dsn value and what to put into the [ENV] section. Thanks, Maya __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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
Re: [Asterisk-Users] Caller ID
Derek, this gives a workaround, and we all know about this workaround,
however it also means that we have to change the Dialplan and rewrite
everything twice, one for no callerid, and the other for callerid.
What Martin is trying to do is change the code in asterisk that sends
the name asterisk as caller id when the caller id is unnknown to
something else, like unknown.
On Fri, 11 Feb 2005 08:33:46 -0800, Derek Whitten [EMAIL PROTECTED] wrote:
http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
http://voip-info.org/wiki-Asterisk+cmd+SetCIDName
http://voip-info.org/wiki-Asterisk+cmd+SetCIDNum
example:
exten = 1,1,SetCallerID(${CALLERID})
or
exten = 1,1,SetCallerID(Your Name (555)555-)
On Fri, 2005-02-11 at 05:45, Martin Roy wrote:
How can I change that when there's no Caller ID instead of displaying
asterisk it display something like Unknown. Because everyone is confuse
when they see a call coming from asterisk.
Thanks
Martin
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--
Derek Whitten [EMAIL PROTECTED]
kFuQ Productions
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Re: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config fromMySQL
By the way, I did fix the typo in my users table so now the context is 'voicepulse_connect_context', just like in the extensions.conf. That didn't fix the problem. Cheers, Maya --- Matthew Boehm [EMAIL PROTECTED] wrote: What version of asterisk? -Matthew - Original Message - From: beonice [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 11, 2005 10:36 AM Subject: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config fromMySQL Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I added extensions and 100 in the [voicepulse_connect_context] with appropriate settings in extensions.conf to direct incoming calls to those mailboxes, and that works. I was expecting that after I added in the db details, reloading or restarting Asterisk would add the new extension from MySQL's 'users' table into the voicemail.conf. It doesn't. As soon as I type (the beginning of the mailbox and also the extension number), I get the message: *CLI Feb 11 08:38:38 WARNING[5224]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '' If I add a line for into my voicemail.conf, all works well. Please help me understand what is going on here! Thanks, Maya. --- My configuration --- My 'users' table has 1 row only, for testing purposes: +---++--+--+ ---+---+++ | context | mailbox| password | fullname | email | pager | options | stamp | +---++--+--+ ---+---+++ | voicemail_connect_context | | 1234 | Moron Tester | [EMAIL PROTECTED] | | attach=yes | 00 | +---++--+--+ ---+---+++ -- The appropriate settings from extensions.conf: [voicepulse_connect_context] ; -- Should match the context you have ; under [voicepulse-in-01] in iax.conf exten = 100,1,Playback(tt-monkeys) exten = 100,2,Record(/tmp/asterisk-recording:gsm) ;exten = 100,3,Wait(2) exten = 100,3,Playback(/tmp/asterisk-recording) ;exten = 100,5,Wait(2) exten = ,1,Playback(transfer,skip) exten = ,2,VoiceMail,u exten = ,102,VoiceMail,b exten = ,1,VoiceMail,u -- My complete voicemail.conf looks like this: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav49|gsm|wav ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] ; Should the email contain the voicemail as an attachment attach=yes ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Minimum length of a voicemail message in seconds ;minmessage=3 ; Maximum length of greetings in seconds ;maxgreet=60 ; How many miliseconds to skip forward/back when rew/ff in message playback skipms=3000 ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 ; Max number of failed login attempts maxlogins=3 ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this: ;externnotify=/usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail password is changed, ; uncomment this: ;externpass=/usr/bin/myapp ; For the directory, you can override the intro file if you want ;directoryintro=dir-intro ; The character set for voicemail messages can be specified here ;charset=ISO-8859-1 ; The ADSI feature descriptor number to download to ;adsifdn=000F ; The ADSI security lock code ;adsisec=9BDBF7AC ; The ADSI voicemail application version number. ;adsiver=1 ; Skip the [PBX]: string from the message title pbxskip=yes ; Change the From: string fromstring=The Asterisk PBX ; ;Change the From: string for pager messages ;pagerfromstring=The Asterisk PBX ; ; Change the email body and/or subject, variables: ; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_DATE ; ; Note: The emailbody config
Re: [Asterisk-Users] Multiple SIP registrations for one account?
On Fri, 11 Feb 2005 14:21:48 +0100, Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! canreinvite=yes does not affect call accounting in any way. U sure? What for example if later on the SIP device forwards the call (note: not using #) and itself steps out of the line? It still has to contact * about the forward (unless you are doing IP to IP forward), your dialplan comes from somewhere. Cheers, Philipp ___ 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 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] *.conf files not parsing
Has anyone ever seen Asterisk fail to parse files referenced by an #include by a *.conf command? e.g.: #include /etc/asterisk/sip-phones.d/*.conf Where the dir sip-phones.d contains sip extension conf files. This worked fine for nearly a month and then mysteriously stopped working for me last night! Regards, /rg ___ 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
RE: [Asterisk-Users] Question about DID
I have is like so
exten = 6149233422,1,Dial(Zap/g2/9233422)
Also I found some config file that ask about the following.. This is not an
Asterisk problem but I can't think of a better group of people to help with
this problem...
Address Type (International, National, Network, Subscriber, Abbreviated)
Numbering Plan (ISDN, Data, Telex, National, Private)
Subaddress Type (NSAP, User)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Crocker
Sent: Friday, February 11, 2005 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Question about DID
How are you telling Asterisk to send the call to the fax group? You should
have something in extensions.conf like
exten = _4135551234,1,Dial($FAXTRUNKS/${EXTEN})
Asterisk should send the EXTEN down as a DID to the fax server
-Matt
On Feb 11, 2005, at 11:05 AM, Eric Hall wrote:
Hello Group
I have a Asterisk server running with 2 Digium T1 cards installed. 1
card connects to Telco via a PRI. The 2nd card is connected to a fax
server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to
have Asterisk route the calls based on DID or FAX tones. Everything is
working great so far. The only problem is the Fax server does not see
the DID. How can I tell if Asterisk it passing the DID and CallerID
info to the server? I seen this was done with HylaFax.
Any help would be great!!
Here is my configs
cat zaptel.conf
#PRI to Telco
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
# PRI to Fax server
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
zapata.conf
[channels]
context=from-analog
signalling=pri_cpe
switchtype=dms100
group=1
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=no
musiconhold=default
channel = 1-23
context=from-sip-internal
switchtype=dms100
signalling=pri_net
group=2
overlapdial=yes
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=no
musiconhold=default
channel = 25-47
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Re: [Asterisk-Users] Caller ID
Stefan Gofferje wrote: ...what probably would be a good idea, because a call from asterisk really looks strange... I have been searching for the position in source but haven't found it yet. Didn't spend too much effort anyway... But if one of the maintainers would do that, it would be nice... I assuming this is when using SIP. I was annoyed by this and make an adjustment which works nicely. channels/chan_sip.c around line 132 look for #define DEFAULT_CALLERID asterisk swap that to Unknown and you're in good shape. ___ 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] polycom ip phones + asterisk
hi all, Anybody could help me to configure soundpoint ip polycom with asterisk in order to get Instant message and presence . Regards harry Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ ___ 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] SIP in the Philippines
I have a frend in Manilla who is trying to connect to an Asterisk-based VoIP provider here in Western Canada. Has anyone had difficulties with SIP in the Philippines ? I'm having a lot of trouble getting info from the provider there (PLDT) and it seems as if the device can't access a port that will allow it to get out and REGISTER with the switchboard even because the provider never sees it make a request. Is it possible that port 5060 is being blocked ? I'm unclear as to whether or not the required port must indeed be 5060 or if Asterisk is somehow able to recognize SIP / UDP on any incoming port and correctly port forward it. They're using a SPA-1001. Thanks, Kurtz ___ 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] IAX Registration Refused
I have a remote * box connected via a dial-up 128K ISDN line into the main * server. I am now getting an error message every time the remote * box gets a new IP address. error messages: Feb 11 11:34:39 NOTICE[11337]: chan_iax2.c:6577 socket_read: Registration of 'wdsdl' rejected: Registration Refused both * boxes where working just fine until I upgraded last night both * servers are able to talk and registor with each other after I type reload in the CLI both asterisk boxes are running: CVS-HEAD-02/11/05-01:18:45 iax.conf [wdsdl] type=friend host=dynamic secret=x context=default permit=0.0.0.0/0.0.0.0 disallow=all allow=ilbc allow=gsm Regards, Tim ___ 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
Re: [Asterisk-Users] IAX Registration Refused
Tim Lewis wrote: I have a remote * box connected via a dial-up 128K ISDN line into the main * server. I am now getting an error message every time the remote * box gets a new IP address. I'm almost certain there is a bug in the current HEAD tree causing that problem. I upgrade almost every day. I'm having this same problem on all the servers that were upgraded after the peer/friend patches yesterday. None of the machines upgraded before that point have been affected. FWIW. B. ___ 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] Asterisk@Home 0.5 Released today
We are releasing a new version of our one-button Asterisk install, [EMAIL PROTECTED], today. This release includes a redesigned web interface and auto-detection of Digium fxo and fxs cards. We have also fixed a lot of bugs and added numerous customer requested enhancements. [EMAIL PROTECTED] is now more secure with passwords on the web pages and better Linux security. http://asteriskathome.sourceforge.net/ __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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
Re: [Asterisk-Users] IAX Registration Refused
On February 11, 2005 01:47 pm, Brian Capouch wrote: I'm almost certain there is a bug in the current HEAD tree causing that problem. It is, I am seeing the same problem. -A. ___ 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
Re: [Asterisk-Users] SIP in the Philippines
Can they ping the box successfully? Do they have enough bandwidth? Are you seeing ANY failed or successful registrations? You can change the SIP port in your SIP.CONF, though I don't know if you can use both ports at the same time. Perhaps. Worth reading the wiki to see. /edg --On Friday, February 11, 2005 10:16 AM -0800 kurtz [EMAIL PROTECTED] wrote: I have a frend in Manilla who is trying to connect to an Asterisk-based VoIP provider here in Western Canada. Has anyone had difficulties with SIP in the Philippines ? I'm having a lot of trouble getting info from the provider there (PLDT) and it seems as if the device can't access a port that will allow it to get out and REGISTER with the switchboard even because the provider never sees it make a request. Is it possible that port 5060 is being blocked ? I'm unclear as to whether or not the required port must indeed be 5060 or if Asterisk is somehow able to recognize SIP / UDP on any incoming port and correctly port forward it. They're using a SPA-1001. Thanks, Kurtz ___ 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 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
Re: [Asterisk-Users] IAX Registration Refused
What is a know good version of cvs? how do I roll back my ver? -Thanks Tim On Fri, 2005-02-11 at 12:47, Brian Capouch wrote: Tim Lewis wrote: I have a remote * box connected via a dial-up 128K ISDN line into the main * server. I am now getting an error message every time the remote * box gets a new IP address. I'm almost certain there is a bug in the current HEAD tree causing that problem. I upgrade almost every day. I'm having this same problem on all the servers that were upgraded after the peer/friend patches yesterday. None of the machines upgraded before that point have been affected. FWIW. B. ___ 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 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] Polycom IP 3000 configuration
I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid=Conference Room Polycom extensions.conf exten = 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten = 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper and gateway in the polycom config. To date nothing seems to work and Polycom is now on a week return a support call to the reseller that sold us the unit. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ 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
RE: [Asterisk-Users] dtmfmode and IAX protocol
On Fri, 2005-02-11 at 00:17 -0600, Rich Adamson wrote: Joseph has been working at bringing up an asterisk box as kind of a newbie, and I think he's using a Sipura as his fxs interface into asterisk. He's having a problem with asterisk passing dtmf to FWD, but didn't say how he's accessing the bank or fedex. So, without a fair amount more detail from him, there's no way to answer his questions or guess at the problem. Exactly. (I was hoping he'd come to his own conclusions.) So... if the Sipura does not do IAX, then it's quite possible that you're not doing IAX on the Sipura. Which means the whole dtmfmode and IAX protocol is moot... -Michael - No. I'm using Sipura-3000 unit to connect to PSTN and my cordless phone. The prefer an external unit over internal card; as I went enough several internal modems on my PC over time. If there was a bus change and I changed the board my modem cards were just piece of junk. In addition it is easer to switch to backup PC if the main one goes down - all I need to do is to change the IP address on the Sipura-3000 and my sip phone (in all 30sec in total including starting *-on backup PC). With an internal card the downtime is a bit longer. If I could find an external Adapter with native IAX connection with 2-4 ports I would buy it today. I know digium has one but it is only one port unit and I would need minimum one-FXS/FXO ports. So, Mark Eissler might be right the Telco system is complicated environment and not everybody follows standards. So the this might be the case between FedEx and UPS. -- #Joseph ___ 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
Re: [Asterisk-Users] Asterisk@Home 0.5 Released today
[EMAIL PROTECTED] wrote: We are releasing a new version of our one-button Asterisk install, [EMAIL PROTECTED], today. This release includes a redesigned web interface and auto-detection of Digium fxo and fxs cards. We have also fixed a lot of bugs and added numerous customer requested enhancements. [EMAIL PROTECTED] is now more secure with passwords on the web pages and better Linux security. Is there an upgrade for current users of .04? http://asteriskathome.sourceforge.net/ __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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 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
Re: [Asterisk-Users] dtmfmode and IAX protocol
On Fri, 2005-02-11 at 10:15 -0500, Mark Eissler wrote: There's enough information if he's using FWD's 8XX-gateway for his toll-free calls to UPS, the bank, etc. First of all, if the wiki says inline (yes, okay, it does) it probably means as inline data as opposed to inband. But the fact is that iax2 ALWAYS sends DTMF out-of-band. While it's true that some VOIP phones (SIP adapters, etc) can be configured to send DTMF inband, I would think that doing so while using IAX is going to result in digits being reproduced twice at the destination--once from the DTMF reproduced from out-of-band and once from the inband DTMF. So if you're using IAX as your trunking protocal you need to use out-of-band DTMF on your IP phones (and adapters) as well. I just check Sipura-3000 setup on Line1 where my phone is connected to and DTMF Tx Method: Auto Auto includes: InBand AVT INFO Auto InBand+INFO AVT+INFO There is no Out-of-band setting. -- #Joseph ___ 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
Re: [Asterisk-Users] Wireless LANs and Asterisk
Mike Meyer wrote: snip Also we had one bridge that seemed to be a week puppy in the litter. It could only muster 60-70% signal strength. It seemed to have problems under all configurations. Finally we positioned it such that it too works well running WEP 64b. I wonder if having 3 wireless bridges in close proximity would have anything to do with the signal strength? I would doubt it though. My memory fails me but for at least one of the wireless standards (802.11a or .11b or .11g or 802.16) there is power control for the rf output of access points. Having several points close together would cause a reduction of power output. I know this isn't a full answer but Don Pobanz ___ 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
Re: [Asterisk-Users] asterisk@home scary log
There are multiple password guessing tools commonly available on the Internet. I eval'ed one of the tools and it took five seconds to guess a password that was five characters in length. It took an hour to guess a password that was eight characters, and around twenty-four hours to guess a password that was eight characters made up of uppercase, lowercase and non-alpha characters (eg, complex). Regardless, the guessing process is simply how much time does one want to devote to doing it (eg, what's the return value for spending the time exploiting a system). Sorry, not in my tests. I used John the Ripper (http://openwall.com/john/ ), which is a tool for cracking passwords from password files using dictionaries and brute force. The password files had passwords in varrying quality, and cracking time was indeed affected. all-numbers password were guessed almost immidietly. [*] Well-composed passwords of 8 characters were not cracked by brute-force in resonable time. I never use products that rely on pre-staged password files; they are no better then the person that assembled the password file and run about the same level of mentality as the script kitties. Try one of the tools that simply starts with a, then b, etc, then aa, ab, etc. There is no preconceived notion as to what the password should be, and will guess _any_ password given enough time. That was the key point, and one of only a few true mechanism to defeat that process is a short duration lockout. (Exception is the use of keys as noted in previous postings.) [*] passwords that should be dialed from phones are relatively short and all-numbers. Are they never exposed to the internet? And that statement is exactly why the fed and state banking examiners are raising all kinds of red flags relative to Internet Banking Systems. Complex passwords aren't a choice for telephone banking, but certainly are for PC Internet banking. One of the controls used to mitigate that risk is a backend system (sort of a batch process) that attempts to analyze customer banking patterns and alerts on unusal events. Lots of banks and international credit card companies use the process, and even the small rural banks use a manual process to do the same. (The majority of banks also use the account lockout mechanism even for the simplest telephone banking system.) If you apply the above discussions to asterisk, how hard do you really think it might be to write a small script to guess the password used to register a sip phone (as an example)? Given what you've already seen on this list, it would not take long at all to determine the IP address of anyone's exposed asterisk box that posts to the list, and beat on their asterisk box to guess the phone's assigned secret. That is exactly one of the common trade journal complaints relative to VoIP security. (Mark has added some code that essentially is an account lockout mechanism to help defeat that process. Not sure if that is cvs head only or if it was moved into stable as well.) It doesn't make much difference whether one exposes telnet or ssh. Both can be exploited. But, the more complex you make the password, the more time-consuming and difficult it is to guess it. So, if you must expose either telnet or ssh, make your passwords very long and complex. If your O/S has the capability to lockout the account after 'xx' failed passwords, then do that. And allow crackers to lock you out. A silly and effective DoS attack. Call it what you want, but a five minute lockout in my book (and a very large number of very professional security folks) is not a DOS at all. Of coarse, if you're one of the few that want to expose common userid's like root, then you're just creating the DOS problem for yourself. Moving ssh or telnet to another tcp/udp port is nothing more then security by obsurity. For anyone in the security business, that step only adds about ten minutes to the process of discovering which services are actually exposed (on any of 65,000 ports) and then beating on those services to exploit them. Very easy task (and since those tasks are automated, who cares about the extra ten minutes). The bottom line for those asterisk readers that have actually read this far is to use complex lenthy passwords where possible, and some sort of alerting mechansim when xx number of passwords are guessed incorrectly (such as an account lockout mechanism with alerts as just one of many available choices). ___ 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
Re: [Asterisk-Users] Asterisk@Home 0.5 Released today
you can always download the tgz from there site and run the install that way. you may want to backup your configs though. --Dalon On Fri, 11 Feb 2005 14:51:10 -0500, Ariel Batista [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: We are releasing a new version of our one-button Asterisk install, [EMAIL PROTECTED], today. This release includes a redesigned web interface and auto-detection of Digium fxo and fxs cards. We have also fixed a lot of bugs and added numerous customer requested enhancements. [EMAIL PROTECTED] is now more secure with passwords on the web pages and better Linux security. Is there an upgrade for current users of .04? http://asteriskathome.sourceforge.net/ __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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 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 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
