[Asterisk-Users] Asterisk and CDRTool
Does anyone have the CDRTool working well with Asterisk? I have been trying to figure out the tricks to getting pricing to show up when searching on CDR's in the tool. I have searched on Google and the mailing archives, but have not found anything too helpful. If anyone has any insight that would be great! Thanks, dj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap Channel error using 4-port FXO TDM400P
On Tue, 6 Jul 2004 12:19:30 -0700, Darrin Johnson [EMAIL PROTECTED] wrote: I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though there is not inbound PSTN call. A part of this activity was generation of a CDR that would get logged. What version of asterisk are you using a fix went in last week for spurious call detections on the TDM400P. I should try downloading the latest zaptel drivers and asterisk code. Jason I upgraded to the latest zaptel code from CVS and ran a modprobe -r wcfxs and modprobe wcfxs and that did fix the problem. Thanks for the help! dj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Channel error using 4-port FXO TDM400P
I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though there is not inbound PSTN call. A part of this activity was generation of a CDR that would get logged. After this error was detected I configured an extension to send incoming calls to an extension that uses the NoCDR() application so that we do not keep sending these bogus CDR's to the DB. The messages displayed on the console window would report this activity at varying intervals sometimes as short as 2 or 3 seconds over and over and over again: -- Starting simple switch on 'Zap/1-1' Jul 6 12:08:48 WARNING[-1418982480]: chan_zap.c:4706 ss_thread: CallerID returned with error on channel 'Zap /1-1' -- Executing NoCDR(Zap/1-1, ) in new stack Jul 6 12:08:48 WARNING[-1418982480]: cdr.c:108 ast_cdr_free: CDR on channel 'Zap/1-1' not posted Jul 6 12:08:48 WARNING[-1418982480]: cdr.c:110 ast_cdr_free: CDR on channel 'Zap/1-1' lacks end -- Executing Wait(Zap/1-1, 90) in new stack == Spawn extension (incoming-pstn, s, 2) exited non-zero on 'Zap/1-1' Jul 6 12:08:56 DEBUG[-1418982480]: chan_zap.c:1717 zt_hangup: Hangup: channel: 1 index = 0, normal = 21, cal lwait = -1, thirdcall = -1 Jul 6 12:08:56 DEBUG[-1418982480]: chan_zap.c:1134 zt_disable_ec: disabled echo cancellation on channel 1 Jul 6 12:08:56 DEBUG[-1418982480]: chan_zap.c:2097 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1 -1 Jul 6 12:08:56 DEBUG[-1418982480]: chan_zap.c:1077 update_conf: Updated conferencing on 1, with 0 conference users -- Hungup 'Zap/1-1' I thought that perhaps there was something on the analog line itself that was causing Asterisk and FXO card to respond as if a call was inbound. I have pulled the analog line from the TDM400P and this is still occurring which means that something between the TDM400P and Asterisk is either not configured correctly or there is a code error. In the archives I noticed some similar issues early one that had to do with callprogress=yes and busydetect=yes, but neither of those things are set. Is there a default for those? The Zapata.conf file for this install looks like this (I have X'd out the phone number of the configured channel for this email): [channels] context=incoming-pstn signalling=fxs_ks hidecallerid=no callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=15.0 txgain=0.0 immediate=no callerid=BrianHomePhone(XXX) XXX- channel = 1 callerid=NotYetConnected2(222) 222- channel = 2 callerid=NotYetConnected3(333) 333- channel = 3 callerid=NotYetConnected4(444) 444- channel = 4 Any thoughts or ideas on what might be the issue are greatly appreciated! Thanks, dj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FireFly client and echo problems with IAX
Hello, I am having horrible echo problems when using the FireFly client on both the caller and callee sides of the call. When I use another IAX soft client like IAXcomm or IAXPhone I do not have the same echo problems. Has anyone else experienced this and do you know what might be the problem? Thanks, dj
[Asterisk-Users] RE: FireFly client and echo problems with IAX
An update to this issue is that the echo only occurs on the softphone that initiates the call regardless of what softphone it is. The receiving client does not hear an echo during the conversation. -Original Message- From: Darrin Johnson [mailto:[EMAIL PROTECTED] Sent: Monday, July 05, 2004 10:11 AM To: '[EMAIL PROTECTED]' Subject: FireFly client and echo problems with IAX Hello, I am having horrible echo problems when using the FireFly client on both the caller and callee sides of the call. When I use another IAX soft client like IAXcomm or IAXPhone I do not have the same echo problems. Has anyone else experienced this and do you know what might be the problem? Thanks, dj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX to IAX call with really bad echo
All, I have spent the last couple of days looking through the mail archives and the documentation on the Wiki, but have not been able to find a solution to the problem. The version of code I am running is from CVS as of 6/30/04. What happens is that when I make an IAX call to another IAX client the caller receives a really bad echo. All of the documentation I found around using -DAGRESSIVE_SUPPRESSION and MARK2 or one of those options seems to be around echo when going out an FXO port through a zaptel channel. It wasn't obvious to me if this same process could be done on the IAX channels and if so how? For reference my iax.conf looks likes: [general] port=5036 bindaddr=0.0.0.0 bandwidth=low echocancel=yes jitterbuffer=yes maxjitterbuffer=500 maxexcessbuffer=100 dropcount=5 register = djohnson:[EMAIL PROTECTED] tos=lowdelay #include /etc/asterisk/users/iax/iax_users [iaxfwd] type=user context=IAX_FWD deny=0.0.0.0/0.0.0.0 permit=65.39.205.0/255.255.255.0 Thanks for the help! dj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX to IAX call with really bad echo
On Friday 02 July 2004 16:37, Darrin Johnson wrote: What happens is that when I make an IAX call to another IAX client the caller receives a really bad echo. All of the documentation I found around using -DAGRESSIVE_SUPPRESSION and MARK2 or one of those options seems to be around echo when going out an FXO port through a zaptel channel. It wasn't obvious to me if this same process could be done on the IAX channels and if so how? I suspect that it is not the IAX channel that is causing echo but rather the far end's interface to the end-user's ear and mic. i.e. the far-end hybrid. Is it all IAX calls or just ones to that particular endpoint? -A. I am using the most recent FireFly client -- 1.9.3 -- and it doesn't matter where I call I still get the echo. What I will do though is try another client and/or an earlier version of FireFly. Thanks, dj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-addons unable to compile
Hello, I know that there was a previous posting regarding this in which the addition of: CFLAGS+=-I../asterisk/include fixed the problem, but I have tried that and am still getting the following when trying to run make: ./mkdep -fPIC -I../asterisk/include/asterisk -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk/include/asterisk -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon _mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 My Makefile -- the beginning -- looks like this: # This program is free software, distributed under the terms of # the GNU General Public License # MODS= CFLAGS+=-fPIC CFLAGS+=-I../asterisk/include/asterisk CFLAGS+=-I../asterisk CFLAGS+=-D_GNU_SOURCE INSTALL=install INSTALL_PREFIX= ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk MODULES_DIR=$(ASTLIBDIR)/modules Does anyone have any other suggestions of what to do to fix the error? Thanks! dj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound ZAP calls
Hello, I have read through the Wiki extensively and have not been able to determine how to fix this problem. When making a call out through a ZAP channel to the PSTN my FireFly client will ring twice before ringing the ZAP channel picks it up and rings the number. I have usecallerid=no and immediate=yes in my Zapata.conf file. I also have made sure that there are no wait(x) statements in the context(s) for the outbound calls. Any ideas how to get to go straight to dial the phone number without the initial two rings? Thanks! Dj
[Asterisk-Users] Rate Engine Application
Hello, Have done some research on the Wiki and via Google, but have not found anything that describes what the tables and columns in the rate engine database really mean. Does anyone have any documentation on those? Thanks, Darrin Johnson Systems Engineer IS Domain Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD SIP Asterisk IAX Firefly
Hello, In my sip.conf I have: ;Register and forward FWD numbers to internal extensions register = FWDNUMBER:[EMAIL PROTECTED]/9500 Which should register Asterisk at FWD and then when any calls are made to FWDNUMBER those calls should be forwarded to extension 9500 as specified in the extensions.conf. What I am getting is it is trying to dial the 9500 (IAX Firefly) client twice when a call is made to FWDNUMBER. The output from the CLI from Asterisk is this: -- Registered '9500' (AUTHENTICATED) at 10.10.20.5:4569 -- Executing Macro(SIP/-080fbd10, stdExt|IAX2|9500) in new stack -- Executing Dial(SIP/-080fbd10, IAX2/9500|15) in new stack -- Called 9500 -- Call accepted by 10.10.20.5 (format GSM) -- Format for call is GSM -- IAX2[9500]/3 is ringing -- Executing Macro(SIP/-0814f210, stdExt|IAX2|9500) in new stack -- Executing Dial(SIP/-0814f210, IAX2/9500|15) in new stack -- Called 9500 Apr 21 11:14:37 WARNING[1158883648]: chan_iax2.c:4898 socket_read: Call rejected by 10.10.20.5: In call -- Hungup 'IAX2[9500]/4' == No one is available to answer at this time -- Executing VoiceMail(SIP/-0814f210, u9500) in new stack -- Playing 'vm-theperson' (language 'en') -- Hungup 'IAX2[9500]/3' == Spawn extension (macro-stdExt, s, 1) exited non-zero on 'SIP/-080fbd10' in macro 'stdExt' == Spawn extension (default, 9500, 1) exited non-zero on 'SIP/-080fbd10' -- Playing 'digits/9' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') Apr 21 11:14:49 WARNING[1142106688]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call 56da75f25ec [EMAIL PROTECTED] for seqno 104 (Response) == Spawn extension (macro-stdExt, s, 2) exited non-zero on 'SIP/-0814f210' in macro 'stdExt' == Spawn extension (default, 9500, 1) exited non-zero on 'SIP/-0814f210' Does anyone know why it is doing this? Have any suggestions? It worked yesterday well and then today I have been getting the above where when it dials the second time it goes to voicemail because the 9500 extension is in use. Thanks for the help! Darrin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zapateller issues
Or you could specify the answer in the Zapateller line like: ... exten = s,2,Zapateller(answer|nocallerid) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Monday, April 12, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Zapateller issues If I remember correctly (and I could be wrong) I think you have to answer the line first... exten = s,1,Answer exten = s,2,Zapateller(nocallerid) exten = s,3,Privacymanager exten = s,4,Dial(a bunch of SIP extensions) -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Monday, April 12, 2004 2:47 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zapateller issues Hi All, In theory if I do this; exten = s,1,Zapateller(nocallerid) exten = s,2,Privacymanager exten = s,3,Dial(a bunch of SIP extensions) My callers should only hear the anti-telemarketing tones if they call from a line that has no caller*ID and then get offered an opportunity to enter it, right? What I'm finding is that in the event of no CID the caller gets dumped into the Privacymanager which then passes the call onto the next step regardless of their input of details. At no time do I hear the tones. I tried, exten = s,1,Zapateller(answer|nocallerid) exten = s,2,Privacymanager exten = s,3,Dial(a bunch of SIP extensions) But then every call was answered regardless of CID and the tones were heard. Any ideas? G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX phone for Pocket PC
Hello all, Does anyone know of a good IAX softphone for Pocket PC's? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hang-ups when using IAX
I have two Asterisk systems running in my environment. In between the two there is a router running NAT. One server services extensions 90XX and the other extensions 95XX. Both boxes are running Red Hat 9 with version 0.7.2 Asterisk. I am running IAX and registering an IAX softphone to each server so two IAX clients with one registered as a 90XX number to the 90XX server and one registered to the 95XX server with a 95XX number. A call is initiated from the client registered to the 90XX server to the client registered on the 95XX server. The call is completed successfully but then after about 30 seconds to a minute the initiating client complains that the remote user (95XX client) hung-up. The 95XX client has the connection still open and live until the hang-up button is manually clicked. The debug in Asterisk shows that the 90XX server records a remote hang-up, but the 95XX server does not record anything until the hang-up button is pushed from the 95XX client. Does anyone have any ideas as to why I would be getting a hang-up after about 30 seconds when using IAX in this type of scenario? I have tried multiple clients with the same result which is implying there must be a problem server-to-server. Thanks much for your help! Darrin Johnson Systems Engineer IS Domain Inc.
RE: [Asterisk-Users] Hang-ups when using IAX
Excellent! That did the trick! Thanks for the tip! Darrin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hadar Pedhazur Sent: Friday, March 12, 2004 3:13 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Hang-ups when using IAX Darrin, I had a similar (though not identical) problem. The solution in my case was to add notransfer=yes in the iax.conf context for the IAX softphone. It's possible that the hand off to attempt a native transfer for you is failing because one of the servers is behind a NAT router. Anyway, it's worth a quick test. Darrin Johnson wrote: I have two Asterisk systems running in my environment. In between the two there is a router running NAT. One server services extensions 90XX and the other extensions 95XX. Both boxes are running Red Hat 9 with version 0.7.2 Asterisk. I am running IAX and registering an IAX softphone to each server - so two IAX clients with one registered as a 90XX number to the 90XX server and one registered to the 95XX server with a 95XX number. A call is initiated from the client registered to the 90XX server to the client registered on the 95XX server. The call is completed successfully but then after about 30 seconds to a minute the initiating client complains that the remote user (95XX client) hung-up. The 95XX client has the connection still open and live until the hang-up button is manually clicked. The debug in Asterisk shows that the 90XX server records a remote hang-up, but the 95XX server does not record anything until the hang-up button is pushed from the 95XX client. Does anyone have any ideas as to why I would be getting a hang-up after about 30 seconds when using IAX in this type of scenario? I have tried multiple clients with the same result which is implying there must be a problem server-to-server. Thanks much for your help! Darrin Johnson Systems Engineer IS Domain Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
