Re: [Asterisk-Users] iptables rules that work?
Brian, Try these: ... -A INPUT -s x.x.x.x -p udp -m udp --dport 5060 -j ACCEPT -A INPUT -s x.x.x.x -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s x.x.x.x -p tcp -m tcp --dport 22 --syn -j ACCEPT ... Sunny --- Brian West [EMAIL PROTECTED] wrote: I'm trying to get some iptables rules that work with asterisk but for some reason I keep blocking everything and or locking myself out of the box.. mybad does anyone have any configs they would like to share that allow asterisk and ssh from x ip? TIA bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Sunny Woo email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iptables rules that work?
Ok it its working but what extra steps besides port 1720 need to be open for chan_h323 to play nice with this setup? Because right now its missing something... :/ bkw On Sat, 20 Sep 2003, Sunny Woo wrote: Brian, Try these: ... -A INPUT -s x.x.x.x -p udp -m udp --dport 5060 -j ACCEPT -A INPUT -s x.x.x.x -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s x.x.x.x -p tcp -m tcp --dport 22 --syn -j ACCEPT ... Sunny --- Brian West [EMAIL PROTECTED] wrote: I'm trying to get some iptables rules that work with asterisk but for some reason I keep blocking everything and or locking myself out of the box.. mybad does anyone have any configs they would like to share that allow asterisk and ssh from x ip? TIA bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Sunny Woo email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oops!!! Current CVS crashes
I don't know whether this ought to go to the bugtracker. I downloaded the current CVS last night and then again just a few minutes ago. In both cases I can crash asterisk very easily by the following method: 1. Call up and leave a voicemail. 2. Log in and listen that I have a new message. 3. Hit 1 to listen, and 'Hasta la vista' asterisk. I also noticed that the normal lines on the console showing vm-login etal aren't shown under the buggy code when a user tries to fetch vm. I noticed it the first time I tried to retrieve voicemail today, with code I built last night. Then I fetched the most recent code and it still behaves just the same. Perhaps it's an interaction with something else in my rather convoluted configs, but somebody else out there ought to give this a try. Reverting to my older CVS (CVS-09/11/03-00:32:05) fixes things immediately. FYI. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Samsung SKP 816H PBX !
Hi,Having Asterisk-0.4.0 with 2FXO port and Samsung SKP 816H PBX in 2 offices.I am able to make call between two offices. But the problem is that calldosen't hangup.Office A [Asterisk+2FXO+SamsungPBX] - I A X Office B [Asterisk+2FXO+SamsungPBX]Configuration files are given here..zapata.confbusydetect=yesbusycount=10callprogress=yesaccountcode=Off-A101callerid="Off-A"101channel = 1accountcode=Off-A102callerid="Off-A"102channel = 2--extensions.conf--exten = _1XX,1,StripMSD,1exten = _XX,2,Dial(Zap/g1/BYEXTENSION||H)exten = _2XX,1,Dial(IAX/[EMAIL PROTECTED]/[EMAIL PROTECTED])--indications.conf--[general]country=us[us]dial = 448/23busy = 424/460congestion = 424/250I have set the value of busycount=3 or busydetect=no and callprogress=no.But the result is same.Any help would be much appreciated!RegardsShimul
Re: [Asterisk-Users] Oops!!! Current CVS crashes
On Sun, 2003-09-21 at 08:50, Brian Capouch wrote: I don't know whether this ought to go to the bugtracker. I downloaded the current CVS last night and then again just a few minutes ago. In both cases I can crash asterisk very easily by the following method: 1. Call up and leave a voicemail. 2. Log in and listen that I have a new message. 3. Hit 1 to listen, and 'Hasta la vista' asterisk. It looks like the fix for this bug has not reached the CVS yet. I checked out CVS last night got the same but then applied the diffs. It then works. http://bugs.digium.com/bug_view_page.php?bug_id=292 Hope this helps. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iptables rules that work?
I'm trying to get some iptables rules that work with asterisk but for some reason I keep blocking everything and or locking myself out of the box.. mybad does anyone have any configs they would like to share that allow asterisk and ssh from x ip? This is what I use for Asterisk form my iptables file.. (mine is open to all IP's so if you need ti limit it down you will have to add -s to it) # SIP -A INPUT -p udp -m udp --dport 5060 -j ACCEPT # IAX2 -A INPUT -p udp -m udp --dport 4569 -j ACCEPT # IAX -A INPUT -p udp -m udp --dport 5036 -j ACCEPT # RTP -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT # SSH -A INPUT -p tcp -m tcp --dport 22 -j ACCEPT Hope that helps, If you see somthing I have left out that would help with my security let me know.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extract header(s) of SIP signalling messages
googled: yes, asked #asterisk: yes.. I need to extract headers of the SIP call signalling protocol on outbound calls (especially call setup response message/ 200 OK). Example: phone with extension 33 place a call to the number 12345 (or sip:[EMAIL PROTECTED]:5060) via SIP. The SIP proxy then sends a header within a call setup response which is important for further CDR processing. I appreciate any idea. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP segfault, problem loading modules, gdb output included
Last week I did aCVS update and since then I haven´t been able to run asterisk normally.The strange thing is that I have even go back to previous versions (0.5.0) andI am seening the same problems. Basically, when I try to load the zap module I get the following error: WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to specify channel 1: Device or resource busy (wxfxo loads fine) If I then do not load zap I get a problem when trying to load iax2 WARNING:Unable to bind to 0.0.0.0 port 4569: Address already in use If I then do not load iax2 asterisk starts fine. However when I try to place a SIP call it segfaults right away. The output of gdb asterisk /etc/asterisk/core.2906 follows: (gdb) bt #0 0x401507f1 in ?? () #1 0x444c7fab in ?? () #2 0x4002c020 in ?? () (gdb) Can someone please help me. Dan
Re: [Asterisk-Users] MY Sql CDR
Look here: http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysql /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Oops!!! Current CVS crashes
This is already fixed in CVS. Mark On Sun, 21 Sep 2003, Brian Capouch wrote: I don't know whether this ought to go to the bugtracker. I downloaded the current CVS last night and then again just a few minutes ago. In both cases I can crash asterisk very easily by the following method: 1. Call up and leave a voicemail. 2. Log in and listen that I have a new message. 3. Hit 1 to listen, and 'Hasta la vista' asterisk. I also noticed that the normal lines on the console showing vm-login etal aren't shown under the buggy code when a user tries to fetch vm. I noticed it the first time I tried to retrieve voicemail today, with code I built last night. Then I fetched the most recent code and it still behaves just the same. Perhaps it's an interaction with something else in my rather convoluted configs, but somebody else out there ought to give this a try. Reverting to my older CVS (CVS-09/11/03-00:32:05) fixes things immediately. FYI. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: how many production systems are there?
On Saturday 20 September 2003 14:29, Steve Totaro wrote: ...i think the nec runs dos... NEC PBX' run a derivative of BSD. -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] built in dial functions?
The implementation of *72 is done for FXS port (the one that gives the dialtone). However you could implement that with some extensions.conf logic. regards Martin On Sat, 20 Sep 2003, Rich Adamson wrote: Martin, That makes sense... but how would one actually use *72#, as an example, when * has two x100p FX ports? i.e, can one enable call forwarding on one fx port and not the other? If I want to call forward my extension (say extn 3000) to extn 3001, is there a way for the user to do that without changing config files? Rich These functions are implemented only for chan_zap (zaptel hardware) and work for FXS/FXO ports. Exception is *8 (remote call pickup) as far as I know. regards Martin On Fri, 19 Sep 2003, Rich Adamson wrote: Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200, ata186, links to fwd and iaxtel, two x100p incoming fx lines, MoH, etc. Everything attempted to date is now working fine. However, testing the above list tends to suggest they don't work (or at least they don't work as I would expect them to.) Example, from a C7960 I dial *78# and hang up. From another sip phone I Dial that extenstion and the 7960 rings. I expected the call to roll over to voicemail or something. Am I missing something here, or are these functions not expected to work on a per-extension basis? I was assuming (probably incorrectly) these functions were custom calling features implemented within * for all extensions. Are my assumptions wrong or do I have to implement something for these to work? TIA, Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Does this thread help? http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html Thanks, this is exactly what I was looking for. I tried experimenting with different codecs myself, and GSM seems to be the only one that works... neither iLBC or Speex went thru. I'm using XLite v1.x Asterisk 0.5.0, wonder if it's a softphone's problem? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN BRI hardware
Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. Is there any particular BRI card that works better with Asterisk than any other? Also, can the BRI interface cards participate in conference, etc., since they aren't a Zaptel interface? Thanks, M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI hardware
We sell: AVM B1 for development environment Eicon Diva Server BRI card for live system (on-board echo canceller) Tan www.telappliant.com - Original Message - From: Mark Hagler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, September 21, 2003 10:43 PM Subject: [Asterisk-Users] ISDN BRI hardware Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. Is there any particular BRI card that works better with Asterisk than any other? Also, can the BRI interface cards participate in conference, etc., since they aren't a Zaptel interface? Thanks, M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Very bad echo (appears that...)
The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options seem to make any difference). Is there any debugging we can turn on to see what the problem may be, this definitely will hurt production of this environment. Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 512-698-8647(V) 425-944-6391 (F) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls being interrupted, analog signalling problems
I'm having trouble with a WX100USB adapter and a Siemens Gigaset cordless phone. If I select fxols as a signalling method, calls are being disconnected. Usually after about 4 minutes, and asterisk just says that the phone has hung up. If I choose fxogs, I immediately get a LINE IN USE message on my phone and I can't even get a dialtone. If I choose fxoks, it mostly works, but sometimes after making a call the adapter will get stuck in a LINE IN USE state, too. I don't know of a proper way to correct it, sometimes disconnecting the USB adapter and reloading the drivers and asterisk fixes that, sometimes not. What is the proper signalling method? What do you people use? I'd appreciate any advice. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very bad echo (appears that...)
Hi, Uncomment the following line in the makefile in /usr/src/zaptel KFLAGS+=-DAGGRESSIVE_SUPPRESSOR Do a make clean install in this directory and reload wcfxo driver (rmmod and then modprobe). Try again and see if there is any improvement. Tan www.telappliant.com - Original Message - From: Asterisk PBX [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, September 21, 2003 11:02 PM Subject: [Asterisk-Users] Very bad echo (appears that...) The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options seem to make any difference). Is there any debugging we can turn on to see what the problem may be, this definitely will hurt production of this environment. Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 512-698-8647(V) 425-944-6391 (F) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very bad echo (appears that...)
I bet your jack is wired backwards.. :) Try checking that out. bkw On Sun, 21 Sep 2003, Asterisk PBX wrote: The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options seem to make any difference). Is there any debugging we can turn on to see what the problem may be, this definitely will hurt production of this environment. Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 512-698-8647(V) 425-944-6391 (F) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MY Sql CDR
I have a question regarding MySQL CDR's: For a given extension I need to limit the number of minutes it can use in a given week. I was thinking about using the CDR information in the MySQL table to see the usage for the week and then if exceeded, STOP the call and play a message. does anybody have a suggestion on how to query the database so I don't have to add-up all the minutes this particular extension have used during the week? Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Sunday, September 21, 2003 10:20 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MY Sql CDR Look here: http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysql /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Very bad echo (appears that...)
Oh, I forgot to say, zaptel/wcfxo is compiled with: KFLAGS+=-DECHO_CAN_MARK2 KFLAGS+=-DAGGRESSIVE_SUPPRESSOR (and, Brian, my jack is wired correct..) -Original Message- From: Lenny Tropiano [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk PBX Sent: Sunday, September 21, 2003 5:02 PM To: '[EMAIL PROTECTED]' Subject: Very bad echo (appears that...) The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options seem to make any difference). Is there any debugging we can turn on to see what the problem may be, this definitely will hurt production of this environment. Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 512-698-8647(V) 425-944-6391 (F) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outgoing limit in chan_sip not working as described
Hi, I just tried to test this feature with fwd. I defined an incoming and outgoing limit of 1. The following comand verifies it: *CLI sip show inuse UsernameincomingLimit outgoingLimit fwd 0 1 0 1 10100 N/A 0 1 I then place a call from 1010 to fwd: -- Executing Goto(SIP/1010-7cbd, fwd1|BYEXTENSION|1) in new stack -- Goto (fwd1,22357,1) -- Executing Dial(SIP/1010-7cbd, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd-6c65 is making progress passing it to SIP/1010-7cbd -- SIP/fwd-6c65 answered SIP/1010-7cbd -- Attempting native bridge of SIP/1010-7cbd and SIP/fwd-6c65 ...and the call connects fine...but the channel limit does not show(and does not work either since I can place multiple concurrent calls): *CLI sip show inuse UsernameincomingLimit outgoingLimit fwd 0 1 0 1 10101 N/A 0 1 If I call a local phone it does seem to work fine. With DEBUG on it mentions something about the dialed number not being local...not sure if it is related to the problem though: DEBUG[12301]: File chan_sip.c, Line 942 (find_user): 22357 is not a local user Any ideas? Patrick? Thanks, Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iptables rules that work?
I'm trying to get some iptables rules that work with asterisk but for some reason I keep blocking everything and or locking myself out of the box.. mybad does anyone have any configs they would like to share that allow asterisk and ssh from x ip? This is what I use for Asterisk form my iptables file.. (mine is open to all IP's so if you need ti limit it down you will have to add -s to it) # SIP -A INPUT -p udp -m udp --dport 5060 -j ACCEPT # IAX2 -A INPUT -p udp -m udp --dport 4569 -j ACCEPT # IAX -A INPUT -p udp -m udp --dport 5036 -j ACCEPT # RTP -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT # SSH -A INPUT -p tcp -m tcp --dport 22 -j ACCEPT Hope that helps, If you see somthing I have left out that would help with my security let me know.. Why are you using -m to achieve what your -p is already doing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Very bad echo (appears that...)
Hi, I am having the same issue with the echo wit that configuration. Were you able to resolve it? Thanks, Kevin -Original Message- From: Asterisk PBX [mailto:[EMAIL PROTECTED] Sent: Sunday, September 21, 2003 6:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Very bad echo (appears that...) The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options seem to make any difference). Is there any debugging we can turn on to see what the problem may be, this definitely will hurt production of this environment. Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 512-698-8647(V) 425-944-6391 (F) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Very bad echo (appears that...)
So 'zap show chanenl channel-no' shows that the echocan is turned on ? Martin On Sun, 21 Sep 2003, Asterisk PBX wrote: Oh, I forgot to say, zaptel/wcfxo is compiled with: KFLAGS+=-DECHO_CAN_MARK2 KFLAGS+=-DAGGRESSIVE_SUPPRESSOR (and, Brian, my jack is wired correct..) -Original Message- From: Lenny Tropiano [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk PBX Sent: Sunday, September 21, 2003 5:02 PM To: '[EMAIL PROTECTED]' Subject: Very bad echo (appears that...) The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options seem to make any difference). Is there any debugging we can turn on to see what the problem may be, this definitely will hurt production of this environment. Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 512-698-8647(V) 425-944-6391 (F) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MY Sql CDR
You could have an AGI script that runs after an outbound call to update a running-total figure with the amount of either the last call or all calls to date in the current period? That way you're just checking a stored value before allowing/denying an outbound call? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Very bad echo (appears that...)
My partner found it!! Problem solved... The error was a syntax error in the zapata.conf channel=1 Should have been written as: channel=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MY Sql CDR
Agree, I can run an AGI script after the outbound call. But where do I invoke the AGI script? it can't be in extensions.conf since, I believe, when either party hang-up, the next priority is not invoked, or am I mistaken? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Crick Sent: Sunday, September 21, 2003 8:33 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MY Sql CDR You could have an AGI script that runs after an outbound call to update a running-total figure with the amount of either the last call or all calls to date in the current period? That way you're just checking a stored value before allowing/denying an outbound call? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Very bad echo (appears that...)
You are kidding,I hope. This typo would manifest itself as an echo problem? May be the parser needs to put out a warning of some kind. That is my 2cents. URiel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Asterisk PBX Sent: Sunday, September 21, 2003 8:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: Very bad echo (appears that...) My partner found it!! Problem solved... The error was a syntax error in the zapata.conf channel=1 Should have been written as: channel=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Very bad echo (appears that...)
Just on a side note can you please put a realname in your name field on your email client. Everytime I see Asterisk PBX I think gee more voicemail. bwk On Sun, 21 Sep 2003, Asterisk PBX wrote: My partner found it!! Problem solved... The error was a syntax error in the zapata.conf channel=1 Should have been written as: channel=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Very bad echo (appears that...)
I wouldn't mess with the gains if I were you. Mark On Sun, 21 Sep 2003 [EMAIL PROTECTED] wrote: Hi, I am having the same issue with the echo wit that configuration. Were you able to resolve it? Thanks, Kevin -Original Message- From: Asterisk PBX [mailto:[EMAIL PROTECTED] Sent: Sunday, September 21, 2003 6:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Very bad echo (appears that...) The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options seem to make any difference). Is there any debugging we can turn on to see what the problem may be, this definitely will hurt production of this environment. Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 512-698-8647(V) 425-944-6391 (F) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone 800 numbers working?
TC wrote: I have seen nufone die, if the callerid is not a cid from us 48 try setting your sic to This seems to not be an issue any more as we have many Canadian customers sending Canadian caller*id's. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone 800 numbers working?
First off the Asterisk mailing list is not a proper place to be discussing these details. Secondly, I have sent you an example config to the email address that you sent the payment from, when you signed up with us. The solution to your problem: You need to register to our system. As of 9:54pm you are not registered to our switch-1 and my debug shows you haven't even attempted a registration in the last 5 days... iax.conf: [general] register = username:[EMAIL PROTECTED] Next time please use [EMAIL PROTECTED] Jeremy McNamara marrandy wrote: On Thursday 18 September 2003 06:50 am, [EMAIL PROTECTED] wrote: Well, I can do outbound calls via NuFone, but nothing on inbound. I get a message that saysThe person you are calling is not reachable, please try again later. IAX2 debug shows nothing. After some time, I copied my config files elsewhere and started with a clean slate, simplified with just one phone (zap/2-1) using NuFone only. Still the same. I can call out but no inbound and no iax2 debug info. I've asked Jeremy (NuFone) to provide the absolute minimum config files to call in and out on a zap/2-1 phone, in the hope that it either work or show a problem elsewhere. Unfortunately, he seems unable to do that. So, if anyone has a working inbound/outbound Nufone connection with a zap/2-1, I'd like their configs - zapata.conf, iax.conf extensions.conf, replace actual password with 'password' of course. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h.323 - success
You have to enable ring indications exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr Jeremy McNamara Roy Sigurd Karlsbakk wrote: hi seems like things are closing in to something that might look like success. I have one problem left: I don't get ring indicator when I dial out from the h.323 phone... Sound is good, so it doesn't look like a codec problem. I'm using chan_capi with early B3. I also use gnugp to route the calls from the phones to asterisk, as the dlink dph-100h requires this. Debug output follows: Any ideas? roy DEBUG - *CLI exten b4: 98013356 -- Executing Dial(H323/ip$10.47.0.1:39307/29476, CAPI/22545070:b98013356|300|T) in new stack -- Called 22545070:b98013356 us: 0.0.0.0:6124 them: 0.0.0.0:0 info: 0.0.0.0:6124 us: 0.0.0.0:6124 them: 0.0.0.0:0 info: 0.0.0.0:6124 -- CAPI[contr1/22545070]/8 is ringing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MY Sql CDR
On Sunday 21 September 2003 18:16, Uriel Carrasquilla wrote: I have a question regarding MySQL CDR's: For a given extension I need to limit the number of minutes it can use in a given week. I was thinking about using the CDR information in the MySQL table to see the usage for the week and then if exceeded, STOP the call and play a message. does anybody have a suggestion on how to query the database so I don't have to add-up all the minutes this particular extension have used during the week? I'm guessing you're looking for a query formula? mysql select sum(billsec) from cdr where calldate '2003-09-01 00:00:00' and '2' in (src,dst); +--+ | sum(billsec) | +--+ | 173 | +--+ 1 row in set (0.03 sec) where '2' is the extension you want to limit. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP NAT QUESTIONS
Hi, Is there anyway to use xlite though a nat I have a xlite - nat- asterisk. * is on a public IP. When I do this, I get an error on the asterisk server because it is trying to use the dirty ip of the computer running xlite. All of the settings in xlite seem to have no effect! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming phone line rollover / hunt?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go about providing a call rollover (single advertised phone number, but allow multiple incoming calls) Thanks in advance, Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/bmJ86gq3eQ0gpNURApYRAKDViZhTUrygxcM3yqlPkdifK4jpuwCfeKVU wympd2pcbcUW1LA4HDeRLzY= =DLug -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming phone line rollover / hunt?
Hi Leif, Hunting, or roll-over has to be done at the CO. The CO is the only place that knows if a line is busy or not **AND** have the ability to redirect the call setup request to a different line On Sun, Sep 21, 2003 at 10:46:20PM -0400, Leif Madsen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go about providing a call rollover (single advertised phone number, but allow multiple incoming calls) Thanks in advance, Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/bmJ86gq3eQ0gpNURApYRAKDViZhTUrygxcM3yqlPkdifK4jpuwCfeKVU wympd2pcbcUW1LA4HDeRLzY= =DLug -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming phone line rollover / hunt?
Check the zapata.conf.sample for the keyword 'group' Jeremy McNamara Leif Madsen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go about providing a call rollover (single advertised phone number, but allow multiple incoming calls) Thanks in advance, Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/bmJ86gq3eQ0gpNURApYRAKDViZhTUrygxcM3yqlPkdifK4jpuwCfeKVU wympd2pcbcUW1LA4HDeRLzY= =DLug -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming phone line rollover / hunt?
how does a PBX control the call setup of inbound calls from the PSTN?? unless you are doing something like ATM an your switch is going to handle processing a call setup request, I don't see how * can deal with hunting from a PSTN side. Certainly from the station or SIP or IAX or H323 side it can deal with it, but I'd be surprised if from the PSTN side. On Sun, Sep 21, 2003 at 11:02:29PM -0400, Jeremy McNamara wrote: Check the zapata.conf.sample for the keyword 'group' Jeremy McNamara Leif Madsen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go about providing a call rollover (single advertised phone number, but allow multiple incoming calls) Thanks in advance, Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/bmJ86gq3eQ0gpNURApYRAKDViZhTUrygxcM3yqlPkdifK4jpuwCfeKVU wympd2pcbcUW1LA4HDeRLzY= =DLug -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MY Sql CDR
I like it. I am thinking of putting this query in a C++ but I am a bit concern on 1) scalability 2) delays in setting up the calls shoud I be concerned? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher Sent: Sunday, September 21, 2003 10:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MY Sql CDR On Sunday 21 September 2003 18:16, Uriel Carrasquilla wrote: I have a question regarding MySQL CDR's: For a given extension I need to limit the number of minutes it can use in a given week. I was thinking about using the CDR information in the MySQL table to see the usage for the week and then if exceeded, STOP the call and play a message. does anybody have a suggestion on how to query the database so I don't have to add-up all the minutes this particular extension have used during the week? I'm guessing you're looking for a query formula? mysql select sum(billsec) from cdr where calldate '2003-09-01 00:00:00' and '2' in (src,dst); +--+ | sum(billsec) | +--+ | 173 | +--+ 1 row in set (0.03 sec) where '2' is the extension you want to limit. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming phone line rollover / hunt?
I am planning on getting 4 analog trunk lines from my carrier (SBC).
~US$14/month/each
And a block of 20 DID numbers for these trunk lines.
(~US$15/month/block of 20) (a block of 20 is the smallest)
Inbound calls come in, and the lines (on the * side) are set to the
same context. (which contain the DID extensions)
If you are not going to use DID, the lines still come into the same
context, just you do not handle the DID extensions. You would just
answer and provide a menu of some sort.
The rollover feature is really just busy call forwarding.
You can buy 4 residential lines. (do not get call waiting)
Setup busy call forwarding:
1 - 2
2 - 3
3 - 4
Then advertise the number for line 1.
Outbound calls would be handled by the group feature of Zapata.
You put the 4 lines in the same group (in zapata.conf) and the
extension.conf would have Dial(Zap/ggroupnumber/${EXTEN})
quote who=John Brown
how does a PBX control the call setup of inbound calls from
the PSTN??
unless you are doing something like ATM an your switch is going to
handle processing a call setup request, I don't see how * can
deal with hunting from a PSTN side.
Certainly from the station or SIP or IAX or H323 side it can
deal with it, but I'd be surprised if from the PSTN side.
On Sun, Sep 21, 2003 at 11:02:29PM -0400, Jeremy McNamara wrote:
Check the zapata.conf.sample for the keyword 'group'
Jeremy McNamara
Leif Madsen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
I have a simple question about incoming phone line rollovers. How are
these usually done? Is this done at the phone company usually, or is
this something that Asterisk or channel bank is capable of? I just
need
someone to give me a brief explanation how it usually works, and if
someone was implementing an Asterisk system, how they would go about
providing a call rollover (single advertised phone number, but allow
multiple incoming calls)
Thanks in advance,
Leif Madsen.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (Cygwin)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQE/bmJ86gq3eQ0gpNURApYRAKDViZhTUrygxcM3yqlPkdifK4jpuwCfeKVU
wympd2pcbcUW1LA4HDeRLzY=
=DLug
-END PGP SIGNATURE-
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