Re: [asterisk-users] BRI confiugration error
On Tue, May 31, 2011 at 11:21:48AM +0530, mahesh katta wrote:
On Tue, May 31, 2011 at 11:08 AM, Shaun Ruffell sruff...@digium.com wrote:
On Tue, May 31, 2011 at 10:48:08AM +0530, mahesh katta wrote:
Hi sir,
I was installed Goautodial server and I have b410p BRI card. BRI card
showing OK with dahdi_tool, this NT mode. whenever I am dialing from
server i am not able to connect the call . in Cli below mention
warning is comming . please what is the mistake with me . help me
snip
-- Executing [0559566768@default:2] Dial(Console/dsp,
Dahdi/g0/0559566768|55|tTo) in new
stack
[May 31 01:10:09] WARNING[30356]: channel.c:3443 ast_request: No channel
type registered for
'Dahdi'
[May 31 01:10:09] WARNING[30356]: app_dial.c:1296 dial_exec_full: Unable to
create channel of type 'Dahdi' (cause 66 - Channel not implemented)
It appears that you do not have chan_dahdi in Asterisk configured and/or
loaded properly.
sir this chan_dahdi.conf file
[channels]
#include dahdi-channels.conf
language=en
context=default
usecallerid=yes
hidecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
relaxdtmf=yes
rxgain=0.0
txgain=0.0
;group=1
;callgroup=1
;pickupgroup=1
busydetect=yes
busycount=6
immediate=no
resetinterval=never
switchtype=euroisdn
signalling=bri_cpe
pridialplan=unknown
prilocaldialplan=unknown
group=0
channel = 1-2
dahdi-channels.conf file
; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
context = default
/etc/dahdi/system.conf
# Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
span=1,1,0,ccs,ami
# termtype: te
bchan=1-2
hardhdlc=3
echocanceller=mg2,1-2
/etc/asterisk/extensions.conf
[globals]
TRUNK=Dahdi/g0
[Default]
exten = _0X,1,Answer()
exten = _0X,2,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,3,Hangup()
shortly this is my configuration.
What is the output of dahdi show channels on the CLI? The No channel
type 'Dahdi' error message you originally reported still suggests that
something is preventing chan_dahdi from loading.
Also, you probably want to put the #include dahdi-channels.conf at the
end of your chan_dahdi.conf file, and drop the group=0 / channel = 1-2
from that file as well since they are already defined in
dahdi-channels.conf that you are including.
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] BRI confiugration error
On Tue, May 31, 2011 at 11:38 AM, Shaun Ruffell sruff...@digium.com wrote:
On Tue, May 31, 2011 at 11:21:48AM +0530, mahesh katta wrote:
On Tue, May 31, 2011 at 11:08 AM, Shaun Ruffell sruff...@digium.com
wrote:
On Tue, May 31, 2011 at 10:48:08AM +0530, mahesh katta wrote:
Hi sir,
I was installed Goautodial server and I have b410p BRI card. BRI card
showing OK with dahdi_tool, this NT mode. whenever I am dialing from
server i am not able to connect the call . in Cli below mention
warning is comming . please what is the mistake with me . help me
snip
-- Executing [0559566768@default:2] Dial(Console/dsp,
Dahdi/g0/0559566768|55|tTo) in new
stack
[May 31 01:10:09] WARNING[30356]: channel.c:3443 ast_request: No
channel
type registered for
'Dahdi'
[May 31 01:10:09] WARNING[30356]: app_dial.c:1296 dial_exec_full:
Unable to
create channel of type 'Dahdi' (cause 66 - Channel not implemented)
It appears that you do not have chan_dahdi in Asterisk configured and/or
loaded properly.
sir this chan_dahdi.conf file
[channels]
#include dahdi-channels.conf
language=en
context=default
usecallerid=yes
hidecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
relaxdtmf=yes
rxgain=0.0
txgain=0.0
;group=1
;callgroup=1
;pickupgroup=1
busydetect=yes
busycount=6
immediate=no
resetinterval=never
switchtype=euroisdn
signalling=bri_cpe
pridialplan=unknown
prilocaldialplan=unknown
group=0
channel = 1-2
dahdi-channels.conf file
; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
context = default
/etc/dahdi/system.conf
# Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
span=1,1,0,ccs,ami
# termtype: te
bchan=1-2
hardhdlc=3
echocanceller=mg2,1-2
/etc/asterisk/extensions.conf
[globals]
TRUNK=Dahdi/g0
[Default]
exten = _0X,1,Answer()
exten = _0X,2,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,3,Hangup()
shortly this is my configuration.
What is the output of dahdi show channels on the CLI? The No channel
type 'Dahdi' error message you originally reported still suggests that
something is preventing chan_dahdi from loading.
sir this command is not working like
go*CLI dahdi show
channels
No such command 'dahdi show channels' (type 'help dahdi show' for other
possible commands)
Also, you probably want to put the #include dahdi-channels.conf at the
end of your chan_dahdi.conf file, and drop the group=0 / channel = 1-2
from that file as well since they are already defined in
dahdi-channels.conf that you are including.
sir I put this line #include dahdi-channels.conf at the end of
chan_dahdi.conf file. till same .
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] BRI confiugration error
On Tue, May 31, 2011 at 11:47:58AM +0530, mahesh katta wrote: sir this command is not working like go*CLI dahdi show channels No such command 'dahdi show channels' (type 'help dahdi show' for other possible commands) Also, you probably want to put the #include dahdi-channels.conf at the end of your chan_dahdi.conf file, and drop the group=0 / channel = 1-2 from that file as well since they are already defined in dahdi-channels.conf that you are including. sir I put this line #include dahdi-channels.conf at the end of chan_dahdi.conf file. till same . Did you delete the duplicate groups / and channels also? Does the output from module load chan_dahdi entered on the console give you the clues you need to fix your configuration file? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AJAM XML output not valid xml
- Original Message - On Mon, 2011-05-23 at 15:41 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 and have been implementing AJAM. I've noticed the final '' is missing from every response I've had so far. Here is an example ajax-response response type='object' id='unknown'generic response='Success' message='Authentication accepted' //response /ajax-response Has anyone else noticed this? Is it a bug in the code or possibly a config setting I've missed? Thanks Ish Can someone else please verify that this is happening to them as well and if so I'll raise the issue... Try the latest code from the 1.8 branch. This sounds very familiar. I think it has already been fixed. $ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8 asterisk-1.8 -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI confiugration error
On Tue, May 31, 2011 at 11:53 AM, Shaun Ruffell sruff...@digium.com wrote: On Tue, May 31, 2011 at 11:47:58AM +0530, mahesh katta wrote: sir this command is not working like go*CLI dahdi show channels No such command 'dahdi show channels' (type 'help dahdi show' for other possible commands) Also, you probably want to put the #include dahdi-channels.conf at the end of your chan_dahdi.conf file, and drop the group=0 / channel = 1-2 from that file as well since they are already defined in dahdi-channels.conf that you are including. sir I put this line #include dahdi-channels.conf at the end of chan_dahdi.conf file. till same . Did you delete the duplicate groups / and channels also? No sir i did't delete the groups in dahdi-channels.conf , but channels are deleted in chan_dahdi.conf file Does the output from module load chan_dahdi entered on the console give you the clues you need to fix your configuration file? this below error is comming in cli module load chan_dahdi [May 31 02:30:49] WARNING[10771]: pbx.c:2965 ast_register_application: Already have an application 'DAHDISendKeypadFacility' [May 31 02:30:49] WARNING[10771]: pbx.c:2965 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/chan_dahdi.conf': Found == Parsing '/etc/asterisk/dahdi-channels.conf': Found [May 31 02:30:49] ERROR[10771]: chan_dahdi.c:11730 process_dahdi: Unknown signalling method 'bri_cp' [May 31 02:30:49] ERROR[10771]: chan_dahdi.c:11280 build_channels: Signalling must be specified before any channels are. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI confiugration error
On Tue, May 31, 2011 at 12:03:13PM +0530, mahesh katta wrote: On Tue, May 31, 2011 at 11:53 AM, Shaun Ruffell sruff...@digium.com wrote: Does the output from module load chan_dahdi entered on the console give you the clues you need to fix your configuration file? this below error is comming in cli module load chan_dahdi [May 31 02:30:49] WARNING[10771]: pbx.c:2965 ast_register_application: Already have an application 'DAHDISendKeypadFacility' [May 31 02:30:49] WARNING[10771]: pbx.c:2965 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/chan_dahdi.conf': Found == Parsing '/etc/asterisk/dahdi-channels.conf': Found [May 31 02:30:49] ERROR[10771]: chan_dahdi.c:11730 process_dahdi: Unknown signalling method 'bri_cp' [May 31 02:30:49] ERROR[10771]: chan_dahdi.c:11280 build_channels: Signalling must be specified before any channels are. Hopefully those two error messages from parsing chan_dahdi.conf and dahdi-channels.conf provide the clues you need to fix your configuration problem? Also, the original configuration files you posted didn't have 'bri_cp' set as the signalling. That makes me wonder if the files you posted are really the ones chan_dahdi is attempting to parse. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI confiugration error
On Tue, May 31, 2011 at 12:17 PM, Shaun Ruffell sruff...@digium.com wrote: On Tue, May 31, 2011 at 12:03:13PM +0530, mahesh katta wrote: On Tue, May 31, 2011 at 11:53 AM, Shaun Ruffell sruff...@digium.com wrote: Does the output from module load chan_dahdi entered on the console give you the clues you need to fix your configuration file? this below error is comming in cli module load chan_dahdi [May 31 02:30:49] WARNING[10771]: pbx.c:2965 ast_register_application: Already have an application 'DAHDISendKeypadFacility' [May 31 02:30:49] WARNING[10771]: pbx.c:2965 ast_register_application: Already have an application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/chan_dahdi.conf': Found == Parsing '/etc/asterisk/dahdi-channels.conf': Found [May 31 02:30:49] ERROR[10771]: chan_dahdi.c:11730 process_dahdi: Unknown signalling method 'bri_cp' [May 31 02:30:49] ERROR[10771]: chan_dahdi.c:11280 build_channels: Signalling must be specified before any channels are. Hopefully those two error messages from parsing chan_dahdi.conf and dahdi-channels.conf provide the clues you need to fix your configuration problem? sir can you help me sir i am trying but its not going fix. how can i do sir. Also, the original configuration files you posted didn't have 'bri_cp' set as the signalling. That makes me wonder if the files you posted are really the ones chan_dahdi is attempting to parse. yes sir , its not original configuration i am changing the signalling .i tried with bri_cp signalling thats why its in cli. sir please help if you want remote my server i can give you . -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf Caller-id detection before first ring
We don't have polarity reversal before first ring I think so. Not confirmed.
I able to see and hear dtmf and ring tones while playing recorded
wave file using audacity. As per the instruction given by Mr. Pezhman
Lai. I found somthing
while googling . The bug instructions as follows
https://issues.asterisk.org/view.php?id=9096nbn=2.
I am using 8 port digium tdm card. So I decided edit
wctdm24xxp/base.c as per the instruction written in above issue. As
of now I didn't touch asterisk chan_dahdi.c.
I modified base.c code as follows,
static void wctdm_dtmfcheck_fakepolarity(struct wctdm *wc, int card, u8 sample8)
{
u32 sample16;
struct fxo *const fxo = (wc-mods[card].fxo);
/* only look for sound on the line if dtmf flag is on, it is an fxo
* card and line is onhook */
if (!dtmf || !(wc-cardflag (1 card)) ||
(wc-modtype[card] != MOD_TYPE_FXO) || fxo-offhook) {
return;
}
/* don't look for noise if we're already processing it, or there is a
* ringing tone */
if (!fxo-readcid !fxo-wasringing
wc-intcount fxo-cidtimer + 400) {
sample16 = DAHDI_XLAW(sample8, wc-chans[card]);
if (sample16 2000 || sample16 -2000) {
fxo-readcid = 1;
fxo-cidtimer = wc-intcount;
if (debug ( card == 2 )) {
printk(KERN_DEBUG DTMF CLIP on
%i %X\n,
card + 1,sample16);
}
// dahdi_qevent_lock(wc-chans[card],
//DAHDI_EVENT_POLARITY);
}
} else if (fxo-readcid wc-intcount fxo-cidtimer + 2000) {
/* reset flags if it's been a while */
fxo-cidtimer = wc-intcount;
fxo-readcid = 0;
}
}
After compilation of above, I just restarted dahdi and monitored
kernel message. I got following messages before receiving call
DTMF CLIP on 3 68
DTMF CLIP on 3 84
DTMF CLIP on 3 58
DTMF CLIP on 3 78
DTMF CLIP on 3 60
DTMF CLIP on 3 48
DTMF CLIP on 3 60
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on 3 84
DTMF CLIP on 3 78
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on 3 68
DTMF CLIP on 3 68
DTMF CLIP on 3 70
DTMF CLIP on 3 A4
DTMF CLIP on 3 70
But some calls, I am getting long hexadecimall value as follows,
DTMF CLIP on 3 48
DTMF CLIP on 3 B64
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on 3 84
DTMF CLIP on 3 78
RING on 1/3!
NO RING on 1/3!
Can u guide me on right Mr.Lali and Cohen.. ?
Thanks Regards,
Ashik Ali
On Sun, May 29, 2011 at 12:13 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Sat, May 28, 2011 at 02:34:36PM +0300, Ashik Ali wrote:
Hi dears,
I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and
Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) .
I am facing problem with detecting caller id before first ring.I
recorded the dahdi channel using dahdi_monitor command. Where I am
able to see and hear caller-id dtmf tones.
Is there a polarity reversal before the caller ID string is sent?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Configuring ISDN PRI using DAHDI
Hi, "pri show version" should show you something like that: libpri version: 1.4.11.4 loadzone should be "de". Better you configure your system with the help of "./Setup dahdi" (which can be found in your wanpipe-source-directory). Or after installation you can change config files (as root) with "wancfg_dahdi" (should be found in /usr/sbin). Best regards, -Thorsten- Am 31.05.2011 00:29, schrieb bilal ghayyad: Hi All; >From the CLI, if I typed pri then I can find the command and the relative commands for it .. does this mean that the libpri is installed well? How can I be sure that Asterisk took the libpri and it is functioning? Now, regarding to the PRI configurations: The provider is using: ISDN PRI E1, hdb3 and they disable the CRC4 At the /etc/dahdi/system.conf I added the following: span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 I have a question, as I am using E1 so it is a wrong to use a loadzone us? Because us is using T1 and not E1? At the /etc/asterisk/chan_dahdi.conf I added the following: context=Incoming group=0 signaling=pri_net (and really I do not know if I have to use pri_cpe) switchtype=euroisdn channel=1-15,17-31 Well, Do I miss any thing? What commands that can help me to know if I have hardware failure in the card, if the E1 is UP or not? Your kindly help is highly appreciated. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI confiugration error
On Tue, May 31, 2011 at 12:29:36PM +0530, mahesh katta wrote: On Tue, May 31, 2011 at 12:17 PM, Shaun Ruffell sruff...@digium.com wrote: On Tue, May 31, 2011 at 12:03:13PM +0530, mahesh katta wrote: On Tue, May 31, 2011 at 11:53 AM, Shaun Ruffell sruff...@digium.com Also, the original configuration files you posted didn't have 'bri_cp' set as the signalling. That makes me wonder if the files you posted are really the ones chan_dahdi is attempting to parse. yes sir , its not original configuration i am changing the signalling .i tried with bri_cp signalling thats why its in cli. It's either bri_cpe (for CPE PtP) or bri_cpe_ptmp (for CPE PtMP). Please make sure you have a valid signalling name. sir please help if you want remote my server i can give you . If it's still not working (after 'module unload chan_dahdi.so' and 'module load chan_dahdi.so', could you please please provide both the error message and the current chan_dahdi.conf (with #include-d files)? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] please help
Hello
after remove the _ and put the number like that 0678922645,1, the issue has
been solved
thank you so much :)
2011/5/31 mahesh katta maheshka...@flexydial.com
Remove the _ in front of your dialplan,like
exten = 0678922645,1,--
On Mon, May 30, 2011 at 11:00 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten = _0678922645.,1,Set(CALLERID(number)=520460587)
exten = _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0678922645
.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten = _067892264*5*,2,Hangup()
i can not call my number but when i delet the last number '5' i can call
without any issue
i want to put all the number please any hel to solve this issue
thanks and regards
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Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
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Web http://www.buzzworks.com
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Re: [asterisk-users] asterisk fails when DNS or internet fails
On Mon, 2011-05-30 at 23:15 -0400, Jeff LaCoursiere wrote: On Mon, 30 May 2011, Sherwood McGowan wrote: True, but with all due respect, if the cache's TTL expires and the OP's PBX cannot reach an external DNS server, they have bigger problems ;-) Slainte all! The Mick I couldn't disagree more. In fact I think this problem is more serious than it is getting credit for, when asterisk is in use in places where Internet connectivity is far from stable. I have several hotels that have gone without Internet connectivity for days, and somewhere between one and three days down they can only spottily call within the system, and can't make outbound calls on their voice T1. Its certainly true that they were suffering without Internet access, but it is very hard to explain to the owners why they can't use their phones. In fact the symptoms are very strange - inbound calls on the T1 get the auto-attendant, but internal transfers fail. No one can call outbound, and only *sometimes* do internal extension to extension calls fail. I still scratch my head about what exactly asterisk is trying to lookup that keeps it from being able to place internal SIP calls from extension to extension, and sadly the few times this has occurred I wasn't around to debug. Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? What kind of info is it about? If it is the hostname of _local_ machines/clients, you should be authoritive. That should keep asterisk happy. If it is about remote nodes, well if your isp-connection is lost, you can not contact them anyway ;-( So run locally your bind-server, authoritive for your own addresses, and caching for external ones. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
I our setup we don't have DNS or Internet connectivity but we are good no issue so far. -- Sent from my iPhone On May 31, 2011, at 7:24 AM, Hans Witvliet h...@a-domani.nl wrote: On Mon, 2011-05-30 at 23:15 -0400, Jeff LaCoursiere wrote: On Mon, 30 May 2011, Sherwood McGowan wrote: True, but with all due respect, if the cache's TTL expires and the OP's PBX cannot reach an external DNS server, they have bigger problems ;-) Slainte all! The Mick I couldn't disagree more. In fact I think this problem is more serious than it is getting credit for, when asterisk is in use in places where Internet connectivity is far from stable. I have several hotels that have gone without Internet connectivity for days, and somewhere between one and three days down they can only spottily call within the system, and can't make outbound calls on their voice T1. Its certainly true that they were suffering without Internet access, but it is very hard to explain to the owners why they can't use their phones. In fact the symptoms are very strange - inbound calls on the T1 get the auto-attendant, but internal transfers fail. No one can call outbound, and only *sometimes* do internal extension to extension calls fail. I still scratch my head about what exactly asterisk is trying to lookup that keeps it from being able to place internal SIP calls from extension to extension, and sadly the few times this has occurred I wasn't around to debug. Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? What kind of info is it about? If it is the hostname of _local_ machines/clients, you should be authoritive. That should keep asterisk happy. If it is about remote nodes, well if your isp-connection is lost, you can not contact them anyway ;-( So run locally your bind-server, authoritive for your own addresses, and caching for external ones. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] To know if the ISDN PRI E1 is UP?
Hi All; This is the output of the pri show status, so I appreciate if to know if that means the E1 is UP? What does it means that the status us (Status: In Alarm, Down, Active)? What in the below result give an indication that it is UP? CC*CLI pri show span 1 Primary D-channel: 16 Status: In Alarm, Down, Active Switchtype: EuroISDN Type: Network Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To know if the ISDN PRI E1 is UP?
your PRI is not up. You can see this Status: In Alarm, Down, Active it means you have some error. Some parameter is not correct with the configuration and with your line. If you are using Sangoma card then please check this link http://wiki.sangoma.com/wanpipe-linux-asterisk-debugging On Tue, May 31, 2011 at 6:02 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; This is the output of the pri show status, so I appreciate if to know if that means the E1 is UP? What does it means that the status us (Status: In Alarm, Down, Active)? What in the below result give an indication that it is UP? CC*CLI pri show span 1 Primary D-channel: 16 Status: In Alarm, Down, Active Switchtype: EuroISDN Type: Network Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
Jeff LaCoursiere j...@sunfone.com writes: Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? If your recursive DNS server returns errors quickly rather than actually trying to look up the names, Asterisk works fine. It is not a particularly nice workaround, but it does work... As long as Asterisk does not actually NEED the DNS information, but that can be most worked around with static configuration of IP addresses in sip.conf. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
It seems to me: That every Asterisk system that is being used for PBX or other internal as well as external use should have a local DNS, run either on the same box or on an adjacent box. A simple BIND installation is low overhead. If remote phones use it for DNS then if they are on the net, they have all info they need to make a call anywhere, and commonly called locations are either in DNS or cached. Then there are no DNS based failures unless the Net is down or the Asterisk box is down. Well, now, I should do as I preach and install Bind on our Asterisk box. However since we do have multiple DNS locally, we are effectively following this advice. C. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
31 maj 2011 kl. 14.49 skrev Benny Amorsen: Jeff LaCoursiere j...@sunfone.com writes: Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? If your recursive DNS server returns errors quickly rather than actually trying to look up the names, Asterisk works fine. It is not a particularly nice workaround, but it does work... As long as Asterisk does not actually NEED the DNS information, but that can be most worked around with static configuration of IP addresses in sip.conf. Longterm we should really integrate an Asynchronus DNS library, like C-Ares. I've been wanting to do that for years. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
may i know what domain is asterisk specifically looking for? coz i don't use domains on the ip phones, i configure them to register to the IP e.g. 10.10.10.1. forgot to mention i am using freepbx as a GUI, does freepbx tells asterisk to look for a specific domain? TIA. Regards, Ron On Tue, May 31, 2011 at 9:17 PM, Olle E Johansson o...@edvina.net wrote: 31 maj 2011 kl. 14.49 skrev Benny Amorsen: Jeff LaCoursiere j...@sunfone.com writes: Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? If your recursive DNS server returns errors quickly rather than actually trying to look up the names, Asterisk works fine. It is not a particularly nice workaround, but it does work... As long as Asterisk does not actually NEED the DNS information, but that can be most worked around with static configuration of IP addresses in sip.conf. Longterm we should really integrate an Asynchronus DNS library, like C-Ares. I've been wanting to do that for years. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queuemetrics with 1.8 queue_log
Hi Guys! We were using queuemetrics since long time with asterisk 1.2 but recently we have install 1.8 asterisk and but there is a big different in queue_log its saying SIP/ instead of Agent/ that is obvious behaviors. so do i need to change Agent/ to SIP/ in queuemetrics ? or is there any workaround to keep business running same like it was before. -S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queuemetrics with 1.8 queue_log
On 05/31/2011 10:21 AM, satish patel wrote: We were using queuemetrics since long time with asterisk 1.2 but recently we have install 1.8 asterisk and but there is a big different in queue_log its saying SIP/ instead of Agent/ that is obvious behaviors. so do i need to change Agent/ to SIP/ in queuemetrics ? or is there any workaround to keep business running same like it was before. I am confident that if you addressed this question directly to Lenz Emilitri and/or Loway Research you would get a more speedy, relevant and precise answer. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
As far as I can tell it is trying to do a reverse lookup on the IPs configured on the system. With the internet down, does the command host 10.10.10.1 (or whatever IPs you have on the system) take a while to come back? Unless you can do a reverse lookup of all the IPs on the system don't expect Asterisk to be able to. If your /etc/hosts is set up correct, you should be able to look up any IP configured on any interface on the system without delay. I'm sure there are other places Asterisk tries to do DNS lookups, but the above info has solved this issue for me in the past. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nhadie ramos Sent: Tuesday, May 31, 2011 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk fails when DNS or internet fails may i know what domain is asterisk specifically looking for? coz i don't use domains on the ip phones, i configure them to register to the IP e.g. 10.10.10.1. forgot to mention i am using freepbx as a GUI, does freepbx tells asterisk to look for a specific domain? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
thank you eric. i have setup a reverse dns for my internal IP, hopefully that works. thanks again! regards Ron On Tue, May 31, 2011 at 10:29 PM, Eric Wieling ewiel...@nyigc.com wrote: As far as I can tell it is trying to do a reverse lookup on the IPs configured on the system. With the internet down, does the command host 10.10.10.1 (or whatever IPs you have on the system) take a while to come back? Unless you can do a reverse lookup of all the IPs on the system don't expect Asterisk to be able to. If your /etc/hosts is set up correct, you should be able to look up any IP configured on any interface on the system without delay. I'm sure there are other places Asterisk tries to do DNS lookups, but the above info has solved this issue for me in the past. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nhadie ramos Sent: Tuesday, May 31, 2011 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk fails when DNS or internet fails may i know what domain is asterisk specifically looking for? coz i don't use domains on the ip phones, i configure them to register to the IP e.g. 10.10.10.1. forgot to mention i am using freepbx as a GUI, does freepbx tells asterisk to look for a specific domain? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping incompatible voice frame on DAHDI/i1/xxxxxxx of format slin since our native format has changed to 0x4 (ulaw)
Hey, Sometime i am getting following messaged on asterisk CLI console just wondering what these messages are look like some codec related. [May 31 12:26:14] NOTICE[7349]: channel.c:4074 __ast_read: Dropping incompatible voice frame on DAHDI/i1/2031444389-28e of format slin since our native format has changed to 0x4 (ulaw) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI buffering event output?
Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple locations, all without this problem. Additionally I have tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent across these versions. manager.conf is identical across all these installations. The problem presents with a php script that opens a socket directly to AMI, a telnet client from the local machine to the AMI, but not when I telnet from the machine to a remote machine running AMI. Does anyone have any input as to what I can try? Best regards, Örn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI buffering event output?
On 05/31/2011 01:38 PM, Örn Arnarson wrote: Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple locations, all without this problem. Additionally I have tried version 1.6.2.11 and 1.8.4.1, and the problem is consistent across these versions. manager.conf is identical across all these installations. The problem presents with a php script that opens a socket directly to AMI, a telnet client from the local machine to the AMI, but not when I telnet from the machine to a remote machine running AMI. Does anyone have any input as to what I can try? Best regards, Örn This is because of some blocking and/or read-write order issue in your PHP script. PHP is a web language, not for standalone scripts doing network I/O. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI buffering event output?
On 11-05-31 01:38 PM, Örn Arnarson wrote: Does anyone have any input as to what I can try? We use Python and StarPy extensively for the TestSuite with no buffering issues. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ControlPlayback's options
On 2011-05-30 14:32, virendra bhati wrote:
Hi List,
Asterisk 's *ControlPlayback* will used for play any recorded file as
an audio player. Is it possible that we can use it for multiple
forward and rewind ?
ex:-
original: ControlPlayback(filename,skipms,ff,rew,stop,pause)
expected
ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
:
Yes, you can use the CPLAYBACKSTATUS, CPLAYBACKOFFSET and
CPLAYBACKSTOPKEY variables to get this behavior.
All you have to do is to list the additional keys and stop keys and
implement this in your dialplan...
I've attached some ael I use for this to implement 1 and 3 as 1 minute
rewind/forward. 4 and 6 as 5 minutes rewind/forward and 7 and 9 as 15
minutes.
5 I use as the pause key, and */# to switch recording.
Greetings,
Johan Wilfer
context conference_play_recordings_conference_connect {
playrec_intro = {
Set(position=0);
goto play,1;
}
play = {
while (true) {
if (${position}==-1) { goto recording_end,1; }
//rewind 5 seconds after every action (so the user doesn't feel
lost...)
Set(position=$[${position}-5000]);
if (${position} 0) { Set(position=0); }
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}));
Set(position=${CPLAYBACKOFFSET});
if (${CPLAYBACKSTATUS}==ERROR) {
Playback(pbx_error_500);
Playback(pbx_endcall);
Wait(2);
Hangup();
}
//If stopped by user
if (${CPLAYBACKSTATUS}==USERSTOPPED) {
if (!${ISNULL(${CPLAYBACKSTOPKEY})}) { goto
${CPLAYBACKSTOPKEY},1; }
}
}
}
recording_end = {
//The end of the recording is reached
Set(position=0);
Background(pbx_endcall);
WaitExten(2);
Hangup();
}
1 = {
Set(position=$[${position}-6]); //Rewind 1 minute
goto play,1;
}
2 = {
//Instructions, that could be aborted with 2.
//1 and 3 could be used to forward/rewind 0 ms effectivly
disabling the defalut..
ControlPlayback(conf_playrec_instructions_full,0,1,3,2);
Wait(1);
goto play,1;
}
3 = {
Set(position=$[${position}+6]); //Forward 1 minute
goto play,1;
}
4 = {
Set(position=$[${position}-30]); //Rewind 5 minutes
goto play,1;
}
5 = {
goto conference_play_recordings_conference_paused, announce, 1;
//Pause
}
6 = {
Set(position=$[${position}+30]); //Forward 5 minutes
goto play,1;
}
7 = {
Set(position=$[${position}-90]); //Rewind 15 minutes
goto play,1;
}
9 = {
Set(position=$[${position}+90]); //Forward 15 minutes
goto play,1;
}
0 = {
goto playrec_intro,1; //Restart playback of the current recording
}
* = {
//Previous recording
//If no recording found, resume playback
goto play,1;
}
# = {
//Next recording
//If no recording found, resume playback
goto play,1;
}
i = {
goto play,1;
}
}
context conference_play_recordings_conference_paused {
announce = {
Set(time=$[${epoch_start}+${position}/1000]);
while(true) {
WaitExten(1);
Background(conf_playrec_pause_part1);
SayUnixTime(${time},${tz},kM);
Background(conf_playrec_pause_part2);
SayUnixTime(${time},${tz},d 'digits/of' B);
Background(conf_playrec_pause_part3);
WaitExten(5);
}
}
i = {
//Every inputs goes here
Wait(1);
goto conference_play_recordings_conference_connect,play,1;
}
}
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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Med vänlig hälsning
Johan Wilfer email: jo...@jttech.se
JT Tech | Utvecklare webb: http://jttech.se
direkt: +46 31 380 91 01 support: +46 31 380 91 00
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Re: [asterisk-users] asterisk fails when DNS or internet fails
On Tue, 2011-05-31 at 10:29 -0400, Eric Wieling wrote: As far as I can tell it is trying to do a reverse lookup on the IPs configured on the system. With the internet down, does the command host 10.10.10.1 (or whatever IPs you have on the system) take a while to come back? Unless you can do a reverse lookup of all the IPs on the system don't expect Asterisk to be able to. If your /etc/hosts is set up correct, you should be able to look up any IP configured on any interface on the system without delay. I'm sure there are other places Asterisk tries to do DNS lookups, but the above info has solved this issue for me in the past. I'm not sure if that's all is true. Sure, if you add a line in /etc/hosts, that works for most applications, as not all commands follow /etc/resolv.conf i just tried, adding a line to /etc/hosts. ping hostname works, but host hostname fails, just as host ip-address. So even when you only put ip-addresses (brrr) into your config files, the reversed-lookup will still spoil the party. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mitel PBX caller id format?
I'm setting up an asterisk server to extend several extensions from a mitel pbx. I'd like to display the caller id that I receive from t he mitel pbx on the sip phone. The mitel PBX person has setup the PBX to send be callerid, but I don't see it. I've set chan_dahdi up with usecallerid=yes cidstart=ring cidsignalling=bell callerid = asreceived cid_rxgain = 0.0 Are there specific settings for receiving the callerid from a mitel pbx, or does something need to be changed on the PBX to send the callerid in the appropriate format? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
On Tue, 31 May 2011, Hans Witvliet wrote: Sure, if you add a line in /etc/hosts, that works for most applications, as not all commands follow /etc/resolv.conf i just tried, adding a line to /etc/hosts. ping hostname works, but host hostname fails, just as host ip-address. So even when you only put ip-addresses (brrr) into your config files, the reversed-lookup will still spoil the party. It depends on the settings in /etc/nsswitch.conf and the resolver library the application uses. Compare ping and ssh referencing a host that is only in /etc/hosts. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Register DOS attack
Hi List Recently i have noticed this attack on couple of servers, usually a foreign IP starts sending tons of register request without any answer to authentication, if you type sip show channels in cli you will see tons of these: 1.2.3.4 (None) 2389603298 00101/1 0x0 (nothing)No Rx: REGISTER since there is no authentication in place, asterisk does not see any failed register attempt, so there wont be anything added to log file as failed attempt. thus fail2ban wont see any activity and wont block the IP. it simply brings down the internet link and the box due to too many sip channels. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI buffering event output?
On 1/06/11 5:38 AM, Örn Arnarson wrote: The problem presents with a php script that opens a socket directly to AMI, a telnet client from the local machine to the AMI, but not when I telnet from the machine to a remote machine running AMI. Are you 100% sure it's happening when you use telnet on the local machine? Are you passing two carriage returns after logging in? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Tuesday, May 31, 2011 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk fails when DNS or internet fails On Tue, 2011-05-31 at 10:29 -0400, Eric Wieling wrote: As far as I can tell it is trying to do a reverse lookup on the IPs configured on the system. With the internet down, does the command host 10.10.10.1 (or whatever IPs you have on the system) take a while to come back? Unless you can do a reverse lookup of all the IPs on the system don't expect Asterisk to be able to. If your /etc/hosts is set up correct, you should be able to look up any IP configured on any interface on the system without delay. I'm sure there are other places Asterisk tries to do DNS lookups, but the above info has solved this issue for me in the past. I'm not sure if that's all is true. Sure, if you add a line in /etc/hosts, that works for most applications, as not all commands follow /etc/resolv.conf i just tried, adding a line to /etc/hosts. ping hostname works, but host hostname fails, just as host ip-address. So even when you only put ip-addresses (brrr) into your config files, the reversed-lookup will still spoil the party. Check /etc/host.conf and make sure hosts is before bind. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
- Original Message - From: Olle E Johansson o...@edvina.net To: Benny Amorsen benny+use...@amorsen.dk Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 31, 2011 16:17 Subject: Re: [asterisk-users] asterisk fails when DNS or internet fails 31 maj 2011 kl. 14.49 skrev Benny Amorsen: Jeff LaCoursiere j...@sunfone.com writes: Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? If your recursive DNS server returns errors quickly rather than actually trying to look up the names, Asterisk works fine. It is not a particularly nice workaround, but it does work... As long as Asterisk does not actually NEED the DNS information, but that can be most worked around with static configuration of IP addresses in sip.conf. Longterm we should really integrate an Asynchronus DNS library, like C-Ares. I've been wanting to do that for years. /O Ole, So what are you waiting for ;) ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
