Re: [asterisk-users] Asterisk crash - segmentation fault
--- On Tue, 3/23/10, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote:
--- On Tue, 3/23/10, Vieri rentor...@yahoo.com
wrote:
My Asterisk 1.2.40 process crashes
regularly in the is_zero_or_null function at:
return (*vp-u.s == 0 || (to_integer (vp)
vp-u.i == 0));
My gdb trace is at:
http://pastebin.com/raw.php?i=hmhzZxye
Other examples here:
http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html
Can anyone please help?
And my Asterisk log shows the following right before
the crash:
Mar 23 12:32:37 VERBOSE[9054] logger.c:
-- Executing
ExecIf(SIP/4070-09464648,
0|Set|REALCALLERIDNUM=4070) in new stac k
Mar 23 12:32:37 DEBUG[9054] app_macro.c: Executed
application: ExecIf
Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result
is '0'
Mar 23 12:32:37 DEBUG[9054] pbx.c: Function result is
'4070'
Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result
is '0'
Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result
is '1'
Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result
is '0'
Mar 23 12:32:37 WARNING[9054] ast_expr2.y: Conversion
of 0 to integer
under/overflowed!
What does this mean?
It's quite clearly a bug, but given that 1.2 is in security
maintenance mode,
it's not a bug that will ever be fixed in an official
release of Asterisk.
Your best bet is to bite the bullet and upgrade to 1.4.
Understood. However, 1.4 also has the same code for that function.
There's something I'd like to know about this logic:
errno = 0;
i = strtoll(vp-u.s, (char**)NULL, 10);
if (errno != 0) {
ast_log(LOG_WARNING,Conversion of %s to integer
under/overflowed!\n, vp-u.s);
free(vp-u.s);
vp-u.s = 0;
return(0);
}
Since my warning message is Conversion of 0 to integer under/overflowed! then
that means the string was set to 0 before the conversion.
0 is within the range LLONG_MIN - LLONG_MAX.
So what I don't understand is why strtoll is failing if vp-u.s is actually 0.
Wouldn't that fail in 1.4 too?
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[asterisk-users] G.729 Codec problem.
Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of called party, but he can't hear me. And also we have sip trunks from callcentric.com, but it is functioning as normal. Also the sip to sip local extension calls works fine. When I make a call through PSTN, the Asterisk log showing the following error: r 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 Please suggest a solution. Do we need additional licence? Thanking you in anticipation, * * *Arun Sasidhar* * * * * * * * * -- _ -- 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] permit/deny in sip.conf iax.conf
23 mar 2010 kl. 22.20 skrev Kevin P. Fleming: Steve Edwards wrote: It may not be as intended, but from a user standpoint, it seems logical and convenient to establish policy in [general] and make exceptions in the entities as needed. Right... for when you have one policy. When you have two policies, each that apply to a dozen or more entries in the config file, then it really doesn't help, it harms. Templates solve that problem completely, because each policy can be its own (named!) template, and they can be combined. Since templates are also very easy to use for the single policy case, they are a better solution to teach people (and they're also easier to implement in the configuration code of the module). In other modules created since chan_sip, we've intentionally avoided this problem, and you'll note that in nearly every other module, the [general] section is exactly that; general settings for the module, and not defaults. In my NACL work, I implemented a channel-wide NACL for blacklist purposes. /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] Safe_asterisk doesn't exists???
@Danny: How do you start your Asterisk ? -- Regards, Prince Singh Drishti-Soft Solutions Pvt Ltd On Wed, Mar 24, 2010 at 6:35 AM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 23 Mar 2010, Danny Dias wrote: This safe_asterisk could be the cause of my problems? how does it works? how can i activate it? safe_asterisk is a script that usually lives at /usr/sbin/safe_asterisk. The script runs in the background. If it detects that Asterisk died, it can send you an email before restarting Asterisk. safe_asterisk is not the problem, but it can be useful as a band-aid until you find the real problem. -- 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 -- _ -- 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] Hook playback or ControlPlayBack cmd
Dear all, I want playback or ControlPlayback cmd to trigger me when a DTMF key is pressed, so I can execute Monitor cmd or any thing I want. Anyone did this job before?. Please help me. Thanks in advance, Giang -- _ -- 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] Firewall audio : need a wide range to work !
Hello list ! I have the following problem at a customer : Their is a firewall in between the internal network (with IP-phones) and the public Asterisk-server. I see the following message when sip debug enabled : [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] --- (11 headers 11 lines) --- [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 101 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port 192.168.0.24:11772 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format PCMA for ID 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format telephone-event for ID 101 alaw) d - 0x1 (telephone-event) [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port 192.168.0.24:11772 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] list_route: hop: sip:ic...@192.168.0.24:5062 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: Parsing sip:ic...@192.168.0.24:5062 for address/port to send to [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: set destination to 192.168.0.24, port 5062 But when opening a range of ports on the firewall 11700 -- 11800, the audio is not coming through !! When opening the ports 11000 -- 11800, then the audio is coming through fine ! Can someone explain me why range 1 is not enough fot the RTP-traffic ?! Jonas. -- _ -- 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] Firewall audio : need a wide range to work !
Have a look at rtp.conf. On 03/24/2010 06:33 AM, jonas kellens wrote: Hello list ! I have the following problem at a customer : Their is a firewall in between the internal network (with IP-phones) and the public Asterisk-server. I see the following message when sip debug enabled : [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] --- (11 headers 11 lines) --- [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 101 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port *192.168.0.24:11772* [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format PCMA for ID 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format telephone-event for ID 101 alaw) d - 0x1 (telephone-event) [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port *192.168.0.24:11772* [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] list_route: hop: sip:ic...@192.168.0.24:5062 sip:itcza...@192.168.0.24:5062 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: Parsing sip:ic...@192.168.0.24:5062 for address/port to send to [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: set destination to 192.168.0.24, port 5062 But when opening a range of ports on the firewall 11700 -- 11800, the audio is not coming through !! When opening the ports 11000 -- 11800, then the audio is coming through fine ! Can someone explain me why range 1 is not enough fot the RTP-traffic ?! Jonas. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 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] Firewall audio : need a wide range to work !
In rtp.conf the audio port range for the public Asterisk server is defined. Why is this important for the firewall at client side ?? By the way the range defined is : rtpstart=11500 rtpend=11600 Do I then need to open up the same range on the firewall at my customer ?? This has nothing to do with incoming traffic on the firewall at my customer's site. Jonas. On Wed, 2010-03-24 at 06:39 -0400, Alex Balashov wrote: Have a look at rtp.conf. On 03/24/2010 06:33 AM, jonas kellens wrote: Hello list ! I have the following problem at a customer : Their is a firewall in between the internal network (with IP-phones) and the public Asterisk-server. I see the following message when sip debug enabled : [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] --- (11 headers 11 lines) --- [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 101 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port *192.168.0.24:11772* [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format PCMA for ID 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format telephone-event for ID 101 alaw) d - 0x1 (telephone-event) [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port *192.168.0.24:11772* [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] list_route: hop: sip:ic...@192.168.0.24:5062 sip:itcza...@192.168.0.24:5062 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: Parsing sip:ic...@192.168.0.24:5062 for address/port to send to [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: set destination to 192.168.0.24, port 5062 But when opening a range of ports on the firewall 11700 -- 11800, the audio is not coming through !! When opening the ports 11000 -- 11800, then the audio is coming through fine ! Can someone explain me why range 1 is not enough fot the RTP-traffic ?! Jonas. -- _ -- 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] Firewall audio : need a wide range to work !
You should be able to establish a very narrow range (4 ports per line) by monitoring the ports with netstat and adjusting accordingly. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Wednesday, March 24, 2010 6:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Firewall audio : need a wide range to work ! In rtp.conf the audio port range for the public Asterisk server is defined. Why is this important for the firewall at client side ?? By the way the range defined is : rtpstart=11500 rtpend=11600 Do I then need to open up the same range on the firewall at my customer ?? This has nothing to do with incoming traffic on the firewall at my customer's site. Jonas. On Wed, 2010-03-24 at 06:39 -0400, Alex Balashov wrote: Have a look at rtp.conf. On 03/24/2010 06:33 AM, jonas kellens wrote: Hello list ! I have the following problem at a customer : Their is a firewall in between the internal network (with IP-phones) and the public Asterisk-server. I see the following message when sip debug enabled : [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] --- (11 headers 11 lines) --- [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 101 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port *192.168.0.24:11772* [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format PCMA for ID 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format telephone-event for ID 101 alaw) d - 0x1 (telephone-event) [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port *192.168.0.24:11772* [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] list_route: hop: sip:ic...@192.168.0.24:5062 sip:itcza...@192.168.0.24:5062 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: Parsing sip:ic...@192.168.0.24:5062 for address/port to send to [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: set destination to 192.168.0.24, port 5062 But when opening a range of ports on the firewall 11700 -- 11800, the audio is not coming through !! When opening the ports 11000 -- 11800, then the audio is coming through fine ! Can someone explain me why range 1 is not enough fot the RTP-traffic ?! Jonas. -- _ -- 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] Using asterisk as avaya definity recordingserver
Moises Silva wrote: On Mon, Mar 22, 2010 at 7:56 PM, Rafael Prado Rocchi pr...@practis.com.brwrote: Hi, it's not that simple. It requires deep modification on asterisk and dahdi sources to work the way you want. Why? I must confess I still don't quite understand what he wants, from what I've read the legacy pbx will place a secondary call via ISDN ( did he mean PRI? ) therefore Asterisk will just Record(), what is it that is not so simple about that? Hi Moses Task: Recording phone calls Here is the scenario; - A legacy system is connected back to back to asterisk pbx with PRI connection and asterisk is connected to the telco via PRI Users(Analog/Digital) Legacy (PRI)-Asterisk---(PRI)---Telco - Telco to users (vise versa) need to be recorded on asterisk - Easily Done - Internal calls (extension to extension) on legacy need to be recording (currently is done via Nice) on asterisk - This's the problem Sam -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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 -- _ -- 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] pstn calls not picked up
Zaptel seems to be running. Channel status: Channel: 4 File Descriptor: 13 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: Zap/4-1 Real: Zap/4-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook Is this right ? Verbosity is at least 3 Channel: 1 File Descriptor: 12 Span: 1 Extension: Dialing: no Context: from-internal Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook Verbosity is at least 3 Hope the above helps in helping me. Thanks, -braman On Wed, Mar 24, 2010 at 1:45 AM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Hello, Please Confirm if the dahdi/Zaptel service is running . check your channels status. On Wed, Mar 24, 2010 at 9:29 AM, Balu Raman brama...@gmail.com wrote: I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls are not being picked up. I don't find anything unusual in asterisk log. I am clueless where I should look. I also find zapata-additional.conf empty. The trouble started when the system was accidentally shut down and rebooted. Any help ? How do I diagnose if the TDM400P is not fried ? Thanks, -braman -- _ -- 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 Shakeel Abbas -- _ -- 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] software version
what is the general view about the versions of the packages that are used with asterisk. lame asterisk asterisk-addons dahdi libpri i like to say on a version and not upgrade due to my experience with Linux and upgrading screwing up things. When it comes to Asterisk i have only one server build under my belt and I have had issue along the way. What do most people do with the software versions? In general if i have a stable system and there is no reason like security or new needed features i don't upgrade. Should i do new builds with the same config (old packages) that i know works? _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_3-- _ -- 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] software version
Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. Regards, Am -- _ -- 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] software version
Hi, Please don't hijack existing thread. Write a new message to the list rather than replying to an existing one if you have a new topic. On Wed, Mar 24, 2010 at 04:36:16PM +0100, Amine Mrichcha wrote: Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. What version of asterisk? How was it installed (from source? packages?) What distribution is it? Services are normally started with init.d scripts at boot time. You should not need to start them manually. -- 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] permit/deny in sip.conf iax.conf
Steve Edwards wrote: It may not be as intended, but from a user standpoint, it seems logical and convenient to establish policy in [general] and make exceptions in the entities as needed. Right... for when you have one policy. When you have two policies, each that apply to a dozen or more entries in the config file, then it really doesn't help, it harms. Templates solve that problem completely, because each policy can be its own (named!) template, and they can be combined. Since templates are also very easy to use for the single policy case, they are a better solution to teach people (and they're also easier to implement in the configuration code of the module). In other modules created since chan_sip, we've intentionally avoided this problem, and you'll note that in nearly every other module, the [general] section is exactly that; general settings for the module, and not defaults. In my NACL work, I implemented a channel-wide NACL for blacklist purposes. Can you talk more about this? Were your Named ACL's something other than templates? What was/were the specific 'pain point/s' you were trying to assuage? For example did you need something not currently offered in the existing frameworks, for example DNS-resolved hostnames for permitting/restricting registration/connection? Or were you just doing a clever/elaborate/well-implemented setup of the existing frameworks? I for one would love to hear your 10,000 foot concepts and any details you'd be willing to share. -Karl -- _ -- 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] Restarting Asterisk using a script - Thanks to all -
Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. Regards, Am -- _ -- 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] software version
Sorry about that. I will send another topic right away. On Wed, Mar 24, 2010 at 4:44 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: Hi, Please don't hijack existing thread. Write a new message to the list rather than replying to an existing one if you have a new topic. On Wed, Mar 24, 2010 at 04:36:16PM +0100, Amine Mrichcha wrote: Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. What version of asterisk? How was it installed (from source? packages?) What distribution is it? Services are normally started with init.d scripts at boot time. You should not need to start them manually. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- _ -- 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] software version
On Wed, 24 Mar 2010, Amine Mrichcha wrote: I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. On Wed, 24 Mar 2010, Steve Edwards wrote: 1) Not a -dev question. Try -users. 2) Choose a better subject. Are you asking how to start Asterisk on reboot or how to restart Asterisk because it is crashing? (If it is crashing, you should fix the problem, not apply a band-aid.) 3) Be more specific. By access do you mean login via ssh? 4) Don't hijack existing threads. -- 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
Re: [asterisk-users] Restarting Asterisk using a script - Thanks to all -
On Wed, 24 Mar 2010, Amine Mrichcha wrote: I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. Why do you need to restart Asterisk? Is it crashing (in which case you should fix the underlying cause, not just put a band-aid over it) or is it not starting when the system is started? If you are not going to access the server, what event is going to trigger the execution of your script? -- 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
Re: [asterisk-users] Restarting Asterisk using a script - Thanks to all -
Hi there, a. You could (maybe) use PHP and send some command via POST, and (after secure/validating the command) use 'exec()' function in php, or 'system()' function. Note: that would require to have a webserver with php installed on it. And allowing the user under which the webserver runs, to restart asterisk via sudoers file. b. You could use a shellscript that sends the command via SSH. In order to avoid password prompt, you could generate a RSA (or DSA) key pair on the machine that will send the command, and copy the rsa_key.pub content on your asterisk box 'authorized_keys'. That would allow you to execute the command remotely via SSH without having to insert the password manually. Note: you could consider using a very limited user on the asterisk box, and with sudoers file allowing it just to restart Asterisk. Regards, Amine Mrichcha wrote: Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. Regards, Am -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] software version
thanks for hijacking my thread. i have an idea don't help him/her so that people will help me! now i am going to re-post this. Date: Wed, 24 Mar 2010 09:08:02 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] software version On Wed, 24 Mar 2010, Amine Mrichcha wrote: I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. On Wed, 24 Mar 2010, Steve Edwards wrote: 1) Not a -dev question. Try -users. 2) Choose a better subject. Are you asking how to start Asterisk on reboot or how to restart Asterisk because it is crashing? (If it is crashing, you should fix the problem, not apply a band-aid.) 3) Be more specific. By access do you mean login via ssh? 4) Don't hijack existing threads. -- 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 _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_3-- _ -- 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] Restarting Asterisk using a script - Thanks to all -
Guess it's not a matter of asterisk as it is of Linux scripting. first check: http://linuxproblem.org/art_9.html then try something like: http://forums.digitalpoint.com/showthread.php?t=70926 but as Steve said, why you need to restart the asterisk service in the first place Fix what's wrong don't just use the band-aid 2010/3/24 Steve Edwards asterisk@sedwards.com On Wed, 24 Mar 2010, Amine Mrichcha wrote: I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. Why do you need to restart Asterisk? Is it crashing (in which case you should fix the underlying cause, not just put a band-aid over it) or is it not starting when the system is started? If you are not going to access the server, what event is going to trigger the execution of your script? -- 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 -- _ -- 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] chan_h323 and ToS
Hi all, I'm using asterisk 1.4.26.2. I need to set TOS on H.323 channel. Does chan_h323.conf support tos (or tos_audio) statement, as well as sip.conf and iax.conf ? Thanks, Daniel -- _ -- 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] Firewall audio : need a wide range to work !
Netstat is indeed a nice tip to view the RTP-connections between the public Asterisk-server and the firewall on location. On Wed, 2010-03-24 at 08:33 -0500, Danny Nicholas wrote: You should be able to establish a very narrow range (4 ports per line) by monitoring the ports with netstat and adjusting accordingly. -- _ -- 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] Restarting Asterisk using a script - Thanks to all -
Hi Steve, Thanks. Yes it is crashing but as a workaround we restart it and we are investigating about the root cause. My question was if it is possible to create a patch script or somthing like this that we can launch from a PC without directly access the server. Regards, On Wed, Mar 24, 2010 at 5:11 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 24 Mar 2010, Amine Mrichcha wrote: I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. Why do you need to restart Asterisk? Is it crashing (in which case you should fix the underlying cause, not just put a band-aid over it) or is it not starting when the system is started? If you are not going to access the server, what event is going to trigger the execution of your script? -- 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 -- _ -- 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] software version (lets try it again)
what is the general view about the versions of the packages that are used with asterisk. lame asterisk asterisk-addons dahdi libpri i like to say on a version and not upgrade due to my experience with Linux and upgrading screwing up things. When it comes to Asterisk i have only one server build under my belt and I have had issue along the way. What do most people do with the software versions? In general if i have a stable system and there is no reason like security or new needed features i don't upgrade. Should i do new builds with the same config (old packages) that i know works? _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/210850552/direct/01/-- _ -- 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] software version (lets try it again)
If it is only a version move, say from asterisk 1.6.1 to 1.6.1.X it's generally ok, but be careful since there are some changes that might hit you from 1.6.0 to 1.6.2 need to have a read into the change log before changing versions, same for the other packages you mention. Alyed 2010/3/24 Ott Rose sixfourimp...@hotmail.com what is the general view about the versions of the packages that are used with asterisk. lame asterisk asterisk-addons dahdi libpri i like to say on a version and not upgrade due to my experience with Linux and upgrading screwing up things. When it comes to Asterisk i have only one server build under my belt and I have had issue along the way. What do most people do with the software versions? In general if i have a stable system and there is no reason like security or new needed features i don't upgrade. Should i do new builds with the same config (old packages) that i know works? -- Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. http://clk.atdmt.com/GBL/go/210850552/direct/01/ -- _ -- 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] software version
On Wed, Mar 24, 2010 at 03:28:52PM +, Ott Rose wrote: what is the general view about the versions of the packages that are used with asterisk. lame asterisk asterisk-addons dahdi libpri Generally I'd say that you should ask a more specific question. i like to say on a version and not upgrade due to my experience with Linux and upgrading screwing up things. When it comes to Asterisk i have only one server build under my belt and I have had issue along the way. What do you currently use? Any specific issues you encounter? What Linux distribution is it? -- 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] Restarting Asterisk using a script - Thanks to all -
Hi, Have you tested safe_asterisk ? Amine Mrichcha wrote: Hi Steve, Thanks. Yes it is crashing but as a workaround we restart it and we are investigating about the root cause. My question was if it is possible to create a patch script or somthing like this that we can launch from a PC without directly access the server. Regards, On Wed, Mar 24, 2010 at 5:11 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Wed, 24 Mar 2010, Amine Mrichcha wrote: I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. Why do you need to restart Asterisk? Is it crashing (in which case you should fix the underlying cause, not just put a band-aid over it) or is it not starting when the system is started? If you are not going to access the server, what event is going to trigger the execution of your script? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com mailto: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 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 -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] Restarting Asterisk using a script - Thanks to all -
On Wed, 24 Mar 2010, Amine Mrichcha wrote: Thanks. Yes it is crashing but as a workaround we restart it and we are investigating about the root cause. My question was if it is possible to create a patch script or somthing like this that we can launch from a PC without directly access the server. The safe_asterisk script (which usually lives at /usr/sbin/safe_asterisk) can be a useful band-aid or safety-net. -- 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
Re: [asterisk-users] software version
Don't be so hard in him/her we all make mistakes, let's just learn from them and move on. Alyed 2010/3/24 Ott Rose sixfourimp...@hotmail.com thanks for hijacking my thread. i have an idea don't help him/her so that people will help me! now i am going to re-post this. Date: Wed, 24 Mar 2010 09:08:02 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] software version On Wed, 24 Mar 2010, Amine Mrichcha wrote: I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. On Wed, 24 Mar 2010, Steve Edwards wrote: 1) Not a -dev question. Try -users. 2) Choose a better subject. Are you asking how to start Asterisk on reboot or how to restart Asterisk because it is crashing? (If it is crashing, you should fix the problem, not apply a band-aid.) 3) Be more specific. By access do you mean login via ssh? 4) Don't hijack existing threads. -- 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 -- The New Busy is not the old busy. Search, chat and e-mail from your inbox. Get started.http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_3 -- _ -- 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] software version
you are correct. just sorta went on a rant. Date: Wed, 24 Mar 2010 10:39:14 -0600 From: al...@vivoxie.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] software version Don't be so hard in him/her we all make mistakes, let's just learn from them and move on. Alyed 2010/3/24 Ott Rose sixfourimp...@hotmail.com thanks for hijacking my thread. i have an idea don't help him/her so that people will help me! now i am going to re-post this. Date: Wed, 24 Mar 2010 09:08:02 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] software version On Wed, 24 Mar 2010, Amine Mrichcha wrote: I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service asterisk restart Any help would be appreciated, sorry if it is a newbie question. On Wed, 24 Mar 2010, Steve Edwards wrote: 1) Not a -dev question. Try -users. 2) Choose a better subject. Are you asking how to start Asterisk on reboot or how to restart Asterisk because it is crashing? (If it is crashing, you should fix the problem, not apply a band-aid.) 3) Be more specific. By access do you mean login via ssh? 4) Don't hijack existing threads. -- 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 The New Busy is not the old busy. Search, chat and e-mail from your inbox. Get started. -- _ -- 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 _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_3-- _ -- 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] Safe_asterisk doesn't exists???
Thanks for all the answers... Asterisk starts at boot time, but if i stop now this is how i will make it up again: asterisk asterisk -vvvrc The weird think is that safe_Asterisk doesn't appear on my process, take a look: [r...@mypbx ~]# ps -A | grep asterisk 14605 ?00:00:02 asterisk 14704 pts/000:00:00 asterisk [r...@mypbx ~]# ps aux | grep asterisk root 14605 0.1 0.2 26396 10256 ? Sl 12:16 0:02 /usr/sbin/asterisk -f -vvvg -c root 14704 0.0 0.0 4216 1276 pts/0S+ 12:17 0:00 rasterisk r root 14819 0.0 0.0 4716 640 pts/3S+ 12:44 0:00 grep asterisk Sometimes my asterisk got frozen...i mean, stop now does not work, asterisk still running but nothing works, not even internall or outgoing calls, my only way out isto kill the process and start again... Thanks in advance for all your help! -- Message: 2 Date: Wed, 24 Mar 2010 14:39:06 +0530 From: Prince Singh pri...@drishti-soft.com Subject: Re: [asterisk-users] Safe_asterisk doesn't exists??? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 626964fc1003240209i6a48bd5cj93fbacd25e2cf...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 @Danny: How do you start your Asterisk ? -- Regards, Prince Singh Drishti-Soft Solutions Pvt Ltd On Wed, Mar 24, 2010 at 6:35 AM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 23 Mar 2010, Danny Dias wrote: This safe_asterisk could be the cause of my problems? how does it works? how can i activate it? safe_asterisk is a script that usually lives at /usr/sbin/safe_asterisk. The script runs in the background. If it detects that Asterisk died, it can send you an email before restarting Asterisk. safe_asterisk is not the problem, but it can be useful as a band-aid until you find the real problem. -- 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 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100324/f2b441a4/attachment-0001.htm -- -- _ -- 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] Safe_asterisk doesn't exists???
turn color=no in the init.d script... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Mar 24, 2010 at 1:23 PM, Danny Dias ing.diasda...@gmail.com wrote: Thanks for all the answers... Asterisk starts at boot time, but if i stop now this is how i will make it up again: asterisk asterisk -vvvrc The weird think is that safe_Asterisk doesn't appear on my process, take a look: [r...@mypbx ~]# ps -A | grep asterisk 14605 ? 00:00:02 asterisk 14704 pts/0 00:00:00 asterisk [r...@mypbx ~]# ps aux | grep asterisk root 14605 0.1 0.2 26396 10256 ? Sl 12:16 0:02 /usr/sbin/asterisk -f -vvvg -c root 14704 0.0 0.0 4216 1276 pts/0 S+ 12:17 0:00 rasterisk r root 14819 0.0 0.0 4716 640 pts/3 S+ 12:44 0:00 grep asterisk Sometimes my asterisk got frozen...i mean, stop now does not work, asterisk still running but nothing works, not even internall or outgoing calls, my only way out isto kill the process and start again... Thanks in advance for all your help! -- Message: 2 Date: Wed, 24 Mar 2010 14:39:06 +0530 From: Prince Singh pri...@drishti-soft.com Subject: Re: [asterisk-users] Safe_asterisk doesn't exists??? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 626964fc1003240209i6a48bd5cj93fbacd25e2cf...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 @Danny: How do you start your Asterisk ? -- Regards, Prince Singh Drishti-Soft Solutions Pvt Ltd On Wed, Mar 24, 2010 at 6:35 AM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 23 Mar 2010, Danny Dias wrote: This safe_asterisk could be the cause of my problems? how does it works? how can i activate it? safe_asterisk is a script that usually lives at /usr/sbin/safe_asterisk. The script runs in the background. If it detects that Asterisk died, it can send you an email before restarting Asterisk. safe_asterisk is not the problem, but it can be useful as a band-aid until you find the real problem. -- 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 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100324/f2b441a4/attachment-0001.htm -- -- _ -- 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] AstLinux 0.7.1 released
The AstLinux Team is happy to announce the release of AstLinux 0.7.1. This is a bugfix release which includes updates to Asterisk (1.4.30), Dahdi and several other items as detailed in the Changelog. http://www.astlinux.org/release/071 Existing users can upgrade from the web interface or from the CLI. From the CLI execute the following: upgrade-run-image check http://mirror.astlinux.org/firmware (should report that Newest available version is astlinux-0.7.1) then do the upgrade: upgrade-run-image upgrade http://mirror.astlinux.org/firmware Reboot After rebooted, you'll need to check two more items: Upgrade firewall plugins: upgrade-arno-firewall check upgrade-arno-firewall upgrade Install sound files: **NOTE sound files are not installed by default starting with 0.7.1** upgrade-asterisk-sounds upgrade core en ulaw Enjoy! Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Safe_asterisk doesn't exists???
https://issues.asterisk.org/view.php?id=16887 do a make update ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Mar 23, 2010 at 7:16 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [r...@mypbx ~]# ps -A | grep asterisk 9118 ? 00:01:30 asterisk [r...@dreampbx ~]# ps aux | grep asterisk root 9118 0.1 0.3 29668 12520 ? Sl Mar22 1:30 /usr/sbin/asterisk -f -vvvg -c root 12096 0.0 0.0 4140 640 pts/1 S+ 18:40 0:00 grep asterisk I have another asterisk servers working and the commands above always shows safe _asterisk as a process... This safe_asterisk could be the cause of my problems? how does it works? how can i activate it? Thanks in advance for your valuable help! DD -- _ -- 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] Asterisk 1.6 and OpenVPN RTP problem
Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn tunnel. When attempting to make a call between the clients, the siganling part of the call goes well. But, when the call is set up, some RTP packets are exchanged at the beginning and then the RTP flow stops (no RTP is exchangd). Wireshark demonstrates no problem with SIP signaling. I am using OpenVPN 2.1.1. Has anyone had such a problem. Please help. -- *Please discover scientific miracles of CORAN* http://www.55a.net/ -- _ -- 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 1.6 and OpenVPN RTP problem
mosbah.abdelkader wrote: Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. Just a guess, set canreinvite=no in the sip.conf for each of the end points Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] installing dahdi card
i have this card installed Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) following the steps below found on freepbx site cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0 make make install make config /sbin/ztcfg echo /sbin/ztcfg /etc/rc.d/rc.local cd /usr/src/libpri-1.4.10.2 make clean make make install when i run make config i do not get any errors but i don't have /sbin/ztcfg. Not sure what to do next. I don't know if need to create the ztcfg file or if i missed a step? I have never had to deal with the cards before. _ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_2-- _ -- 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] This is a test, hijack this
Hello Asterisk, This is only a test, because I can't start new thread in this list... -- Best regards, Gergo mailto:csi...@gmail.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] installing dahdi card
On Wed, Mar 24, 2010 at 07:18:52PM +, Ott Rose wrote: i have this card installed Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) following the steps below found on freepbx site cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0 That's not the latest. Please use latest dahdi (currently 2.2.1). make make install make config /sbin/ztcfg Nope. dahdi_cfg . But actually: dahdi_genconf modules /etc/init.d/dahdi start dahdi_genconf /etc/init.d/dahdi start This should provide you with an initial configuration. echo /sbin/ztcfg /etc/rc.d/rc.local Nope . 'make config' installs the dahdi init.d script. If it doesn't, it's a bug that should be fixed (and just copy it manually) cd /usr/src/libpri-1.4.10.2 make clean make make install This looks reasonable. -- 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
[asterisk-users] Dahdi-linux Dahdi-tools 2.2.1.1 Release Announcement
The Asterisk Development Team is pleased to announce the release of DAHDI-Linux and DAHDI-Tools version 2.2.1.1. DAHDI-Linux 2.2.1.1, DAHDI-Tools 2.2.1.1, and DAHDI-Linux-Complete are available for immediate download at http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete These releases contain wct4xxp driver support for Lantiq's updated Quadfalc 3.1 T1/E1 framer chip. For a full list of changes in these releases, please see the ChangeLogs at http://svn.asterisk.org/svn/dahdi/linux/tags/2.2.1.1/ChangeLog and http://svn.asterisk.org/svn/dahdi/tools/tags/2.2.1.1/ChangeLog Issues found in these releases can be reported at http://issues.asterisk.org Thank you for your continued support of Asterisk!-- _ -- 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] This is a test, hijack this
Gergo Csibra escribió: Hello Asterisk, This is only a test, because I can't start new thread in this list... If you can send an email, you can start a new thread on this list. What's the point of all this? -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] installing dahdi card
Date: Wed, 24 Mar 2010 21:42:09 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] installing dahdi card On Wed, Mar 24, 2010 at 07:18:52PM +, Ott Rose wrote: i have this card installed Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) following the steps below found on freepbx site cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0 That's not the latest. Please use latest dahdi (currently 2.2.1). sorry i did get 2.2.1 i copied that form old doc i had. make make install make config /sbin/ztcfg Nope. dahdi_cfg . i have dahdi_cfg in /usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools not sure what i need to do with it. can you give me some more details. i have never had to build asterisk with a card before. But actually: dahdi_genconf modules /etc/init.d/dahdi start dahdi_genconf /etc/init.d/dahdi start This should provide you with an initial configuration. echo /sbin/ztcfg /etc/rc.d/rc.local Nope . 'make config' installs the dahdi init.d script. If it doesn't, it's a bug that should be fixed (and just copy it manually) chkconfig --list dahdi 0:off 1:off 2:on3:on4:on5:on6:off i don't fully understand how this is setup or how to set it up. sorry if i am not providing enough info. cd /usr/src/libpri-1.4.10.2 make clean make make install This looks reasonable. -- 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 _ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_2-- _ -- 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] installing dahdi card
On Wed, Mar 24, 2010 at 07:59:56PM +, Ott Rose wrote: Date: Wed, 24 Mar 2010 21:42:09 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] installing dahdi card On Wed, Mar 24, 2010 at 07:18:52PM +, Ott Rose wrote: i have this card installed Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) following the steps below found on freepbx site cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0 That's not the latest. Please use latest dahdi (currently 2.2.1). sorry i did get 2.2.1 i copied that form old doc i had. make make install make config /sbin/ztcfg Nope. dahdi_cfg . i have dahdi_cfg in /usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools not sure what i need to do with it. can you give me some more details. i have never had to build asterisk with a card before. But actually: dahdi_genconf modules /etc/init.d/dahdi start dahdi_genconf /etc/init.d/dahdi start This should provide you with an initial configuration. echo /sbin/ztcfg /etc/rc.d/rc.local Nope . 'make config' installs the dahdi init.d script. If it doesn't, it's a bug that should be fixed (and just copy it manually) Have you tried running those commands? See also http://docs.tzafrir.org.il/dahdi-tools/ chkconfig --list dahdi 0:off 1:off 2:on3:on4:on5:on6:off init.d script was installed, then. -- 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] installing dahdi card
Date: Wed, 24 Mar 2010 22:11:28 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] installing dahdi card On Wed, Mar 24, 2010 at 07:59:56PM +, Ott Rose wrote: Date: Wed, 24 Mar 2010 21:42:09 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] installing dahdi card On Wed, Mar 24, 2010 at 07:18:52PM +, Ott Rose wrote: i have this card installed Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) following the steps below found on freepbx site cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0 That's not the latest. Please use latest dahdi (currently 2.2.1). sorry i did get 2.2.1 i copied that form old doc i had. make make install make config /sbin/ztcfg Nope. dahdi_cfg . i have dahdi_cfg in /usr/src/dahdi-linux-complete-2.2.1+2.2.1/tools not sure what i need to do with it. can you give me some more details. i have never had to build asterisk with a card before. But actually: dahdi_genconf modules /etc/init.d/dahdi start dahdi_genconf /etc/init.d/dahdi start This should provide you with an initial configuration. echo /sbin/ztcfg /etc/rc.d/rc.local Nope . 'make config' installs the dahdi init.d script. If it doesn't, it's a bug that should be fixed (and just copy it manually) Have you tried running those commands? See also http://docs.tzafrir.org.il/dahdi-tools/ thanks for the link. i think its working. i ran a couple of the commands. # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.988% 99.991% 99.992% 99.991% 99.992% 99.991% 99.991% chkconfig --list dahdi 0:off 1:off 2:on3:on4:on5:on6:off init.d script was installed, then. -- 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 _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/210850552/direct/01/-- _ -- 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] This is a test, hijack this
On 03/24/2010 03:56 PM, Miguel Molina wrote: Gergo Csibra escribió: Hello Asterisk, This is only a test, because I can't start new thread in this list... If you can send an email, you can start a new thread on this list. What's the point of all this? He was probably having the same problem I've had where I can reply to exiting threads fine but any time I send a fresh email to start a new thread it never goes through. Murphy's law being what it is this email that he suspected would never go through... did. -Dave -- _ -- 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] This is a test, hijack this
Wednesday, March 24, 2010, 9:42:52 PM, Dave wrote: On 03/24/2010 03:56 PM, Miguel Molina wrote: Gergo Csibra escribió: This is only a test, because I can't start new thread in this list... If you can send an email, you can start a new thread on this list. What's the point of all this? He was probably having the same problem I've had where I can reply to exiting threads fine but any time I send a fresh email to start a new thread it never goes through. Murphy's law being what it is this email that he suspected would never go through... did. Your diagnosis is perfect :) Normally I read lists in digest mode, with mime-digest I can reply to individual messages. I use another e-mail address to do this. But I can't start new thread with this (other) e-mail address. With this I can (as we seen :)). -- Best regards, Gergomailto:csi...@gmail.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] This is a test, hijack this
Gergo is using Gmail, He is also using an IMAP client to author his messages (as I do). The messages (new threads) he authors ARE of course 'going through' (that is to say, he CAN start a new thread) but because of the way the Gmail 'label' paradigm is 'shoe-horned' into the IMAP folder paradigm (read: less than perfectly), the Gmail server logic ONLY applies the SENT MAIL label, but not (also) the INBOX label therefore they don't show up in his IMAP inbox (I'm using inbox synonymously with whatever folder that he has created server-side rules for). THEREFORE the perception is that they don't 'go through' because he can't see them in his inbox and assumes therefore that nobody else can see them either. However, as we know they DO go through. As a workaround he should grab the message from his IMAP Sent Mail folder and copy them into his inbox. I put quotes around the folder names because of course there are no such folders in reality--only labels, but those actions he can trigger the IMAP server to apply the right Gmail labels 'behind-the-scenes I haven't tried to think through why the nuances are such that this characteristic doesn't express itself when he replies to others' threads, but I suspect it has something to do with Gmail's 'threading' logic. I hope this helps. -Karl [experience, the] Original Message - From: Dave Fullerton dfullertaster...@shorelinecontainer.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 24, 2010 3:42 PM Subject: Re: [asterisk-users] This is a test, hijack this On 03/24/2010 03:56 PM, Miguel Molina wrote: Gergo Csibra escribió: Hello Asterisk, This is only a test, because I can't start new thread in this list... If you can send an email, you can start a new thread on this list. What's the point of all this? He was probably having the same problem I've had where I can reply to exiting threads fine but any time I send a fresh email to start a new thread it never goes through. Murphy's law being what it is this email that he suspected would never go through... did. -Dave -- _ -- 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] Which folder for sounds?
Tzafrir Cohen wrote: On Mon, Mar 22, 2010 at 09:38:26PM -0400, sean darcy wrote: 1.6.2: -- Executing [...@incoming-pstn-line:4] VoiceMail(DAHDI/4-1, 1...@default,u) in new stack -- DAHDI/4-1 Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]: file.c:953 ast_streamfile: Unable to open vm-intro (format 0x4 (ulaw)): No such file or directory But: locate vm-intro /var/lib/asterisk/sounds/en/vm-intro.gsm /var/lib/asterisk/sounds/en/vm-intro.ulaw /var/lib/asterisk/sounds/en/vm-intro.wav head -12 /etc/asterisk/asterisk.conf [directories](!) ; remove the (!) to enable this As long as it is not enabled, the compile-time defaults are used. astetcdir = /etc/asterisk astmoddir = /usr/lib64/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk So in which folder are these sounds supposed to be? Unless you specifid the full path explicitly, the sound files are looked for in the following pathes (in the following order). Suppose you wanted the sound file called: 'somewhere/soundname' DATADIR/sounds/LANG_FULL/somewhere/soundname DATADIR/sounds/LANG/somewhere/soundname DATADIR/sounds/somewhere/soundname DATADIR/sounds/DEFAULTLANG/soundname DATADIR defaults to /var/lib/asterisk (unless you use the Debian/Ubuntu packages...) and can be set in asterisk.conf otherwise. LANG_FULL is the complete value of LANGUAGE, if it is set. LANG is the value of LANGUAGE sliced after the first '_'. For instance, if LANG_FULL was 'en_US_Whatever', LANG will be 'en'. DEFAULTLANG defaults to 'en' and can be set to a default value in asterisk.conf. Normally people don't change it. Hmm... 'core show settings' does not show datadir. Should it? So the sounds are in /var/lib/asterisk/sounds/en which should be DATADIR/sounds/DEFAULTLANG/ since: DATADIR defaults to /var/lib/asterisk, unless set in asterisk.conf which it isn't (as pointed out by one helpful poster!) DEFAULTLANG defaults to en unless changed in asterisk.conf. # grep DEFAULTLANG /etc/asterisk/asterisk.conf # But 1.6.2 can't find them. I've found that if I make the folders explicit, by removing the (!) in asterisk.conf it does find them. Which means that somehow the default datadir is NOT /var/lib/asterisk. Odd. A long winded way of saying yes, it would be helpful if 'core show settings' showed datadir. sean -- _ -- 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] This is a test, hijack this
Wednesday, March 24, 2010, 10:35:20 PM, Karl wrote: Gergo is using Gmail, He is also using an IMAP client to author his messages (as I do). No. I don't use any IMAP thing :) I use The Bat! but only to download messages with POP3, and I send messages through my ISP's SMTP server. -- Best regards, Gergomailto:csi...@gmail.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] AMD reporting NOTSURE most of the time
On 24/03/10 3:06 PM, Steve Moran wrote: I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering lots of asterisk calls, since the AMDSTATUS always reports things such as:- AMDSTATUS:NOTSURE-AMDCAUSE:TOOLONG-5500 Looks like it's missing the first word - some VoIP providers take a while to pass audio - might be that there is a delay in your dialplan or that the first words of audio are simply not transmitted. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] AMD reporting NOTSURE most of the time
On 24/03/10 3:06 PM, Steve Moran wrote: I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering lots of asterisk calls, since the AMDSTATUS always reports things such as:- AMDSTATUS:NOTSURE-AMDCAUSE:TOOLONG-5500 Alternatively your threshold might be too high - do a few tests to your own phone and make sure it recognizes the individual words. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] new server install errors starting asterisk
here is the issue phones freepbx-2.7.0]# ./start_asterisk start STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. mpg123: no process killed - Asterisk could not start! Use 'tail /full' to find out why. i the only place i know where logs are at is /var/log/asterisk i don't have the log full there. here are the versions of everything i have os: centos 5.4 asterisk: 1.6.2.6 asterisk-addons: 1.6.2.0 dahdi-linux-complete: 2.2.1+2.2.1 freepbx: 2.7.0 lame: 3.98.4 libpri: 1.4.10.2 mysql: 5.0.77 httpd: 2.2.3 _ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_2-- _ -- 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] new server install errors starting asterisk
Just try running: asterisk -vcd And you'll see the error. Alternatively you can edit /etc/asterisk/logger.conf to allow you to have a full log. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] new server install errors starting asterisk
On Wed, 24 Mar 2010, Ott Rose wrote:
here is the issue
phones freepbx-2.7.0]# ./start_asterisk start
Yep. That's the issue :)
STARTING ASTERISK
Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.
Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.
mpg123: no process killed
I don't know where your configuration stashes files, but you can start
Asterisk by hand using a command similar to:
sudo -u whatever-user-you-run-asterisk-as\
/usr/sbin/asterisk\
-C /etc/asterisk/{$PROJECT}/asterisk.conf\
-c\
-f\
-g\
-n\
-p\
-d -d -d -d -d -d\
-v -v -v -v -v -v\
${END_OF_LIST}
This should give you a clue to your issue.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] Asterisk 1.6 and OpenVPN RTP problem
Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn tunnel. When attempting to make a call between the clients, the siganling part of the call goes well. But, when the call is set up, some RTP packets are exchanged at the beginning and then the RTP flow stops (no RTP is exchangd). Wireshark demonstrates no problem with SIP signaling. I am using OpenVPN 2.1.1. Has anyone had such a problem. I had a vaguely-similar problem, getting a Nokia N810's Telepathy- based SIP client to talk to Asterisk over an OpenVPN connection. The problem in that case turned out to be the fact that the Nokia was sending all of the packets to the Asterisk server, using its primary-network (WiFi) IP address, rather than the address to which its end of the OpenVPN tunnel was bound. The SIP packets from the Asterisk server had no way to get back to the client. The fix for this was to stick a couple of scripts into the Nokia, to be executed when OpenVPN started or stopped the VPN tunnel. The up script changes the SIP configuration, setting its local IP address parameter to that of the Nokia end of the tunnel, while the down script clears this override. Works fine. That doesn't sound like exactly the problem you're having, though, since you're getting SIP through the tunnel OK. The problem sounds more as if the RTP packets from one client are either not being send through the tunnel at all, or are being dropped prior to getting to the other. There may be a couple of ways to fix this: (1) As another poster suggested, specify canreinvite=no (or, in 1.6, directmedia=no) for each of your SIP clients. This will prevent them from trying to send the RTP directly to one another, instead sending it to Asterisk for forwarding. This is probably the most reliable approach. It's also probably the only one which will allow reliable connections between these clients, and SIP endpoints which aren't part of your own local IP-address space. (2) If you really do want to try to allow directmedia connections between the clients, you'll need to make certain of two things: [A] Your OpenVPN setup, for each client, must install a route on each client which directs the client to send all packets for any address on the entire VPN back to the VPN server. Without such a route being installed, it's likely that the OpenVPN-installed routing would only channel packets for the OpenVPN server itself into the tunnel. Packets for other IP addresses in the OpenVPN range would end up being sent out through the client's normal IP route, and probably lost forever in the grand stew of the Intertube. [B] Make sure that your OpenVPN setup allows direct client-to- client communications. There's a parameter which can disable this, and permits only client-to-server packets to survive... make sure you haven't set this. (3) You may need to make sure that your iptables (or similar) configuration isn't accidentally NAT'ing packets which are trying to come in through the OpenVPN tunnel and then go back out through another OpenVPN tunnel. -- _ -- 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] Aastra weirds IP 169.x.x.x
Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP, V2.5.3.18, January 2010 , English , ZIP , 2,849 KB on the site: http://www.aastra.com/cps/rde/xchg/SID-3D8CCB6A-FE6670DF/04/hs.xsl/19705.htm Could this make problems? We are receving very weirds ip on the server, see here: mypbx*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 10.4.1.130 (None) b7dd744679a 00101/14365 0x0 (nothing)No Rx: REGISTER 10.4.1.151 308 8cbe459c33e 00102/16076 0x0 (nothing)No Tx: NOTIFY 10.4.1.144 368 607d5af86cd 00102/25625 0x0 (nothing)No Tx: NOTIFY 169.254.236.26 308 4f407ce65eb 00102/15097 0x0 (nothing)No Tx: NOTIFY 169.254.21.164 309 f1e31e48e10 00102/23948 0x0 (nothing)No Tx: NOTIFY 5 active SIP channels Can you see the 169.x.x.x? whats the meaning of this? is this a serious firmware problem? dhcp problem? Thanks in advance DD -- _ -- 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] Aastra weirds IP 169.x.x.x
You should be safe to upgrade to the latest Aastra firmware - i've never had problems with any of the Aastra upgrades - that being said, thats probably better asked to Aastra support instead of here. As for the 169.254.0.0/16 address... theres nothing weird about these addresses... 169.254.0.0/16 is a block designated as the link-local address block, operating systems can assign an IP in this range to themselves automatically when there is either no statically configured address or they are unable to contact a DHCP server. Its the same with the FE80::/10 block in IPv6. Nearly all of the time you see these addresses its a problem where your devices are unable to talk to your DHCP server for some reason. I'm actually a little surprised that you are able to have those phones connect to your Asterisk server at all... do you have a 169.254.0.0/16address assigned on your server or a router between the phones and Asterisk server? -- Matt On Wed, Mar 24, 2010 at 7:07 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP, V2.5.3.18, January 2010 , English , ZIP , 2,849 KB on the site: http://www.aastra.com/cps/rde/xchg/SID-3D8CCB6A-FE6670DF/04/hs.xsl/19705.htm Could this make problems? We are receving very weirds ip on the server, see here: mypbx*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 10.4.1.130 (None) b7dd744679a 00101/14365 0x0 (nothing)No Rx: REGISTER 10.4.1.151 308 8cbe459c33e 00102/16076 0x0 (nothing)No Tx: NOTIFY 10.4.1.144 368 607d5af86cd 00102/25625 0x0 (nothing)No Tx: NOTIFY 169.254.236.26 308 4f407ce65eb 00102/15097 0x0 (nothing)No Tx: NOTIFY 169.254.21.164 309 f1e31e48e10 00102/23948 0x0 (nothing)No Tx: NOTIFY 5 active SIP channels Can you see the 169.x.x.x? whats the meaning of this? is this a serious firmware problem? dhcp problem? Thanks in advance DD -- _ -- 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] Maximum number of PRI calls on 1 asterisk box (no HW echo)
Hi, Does anyone have any good empirical data suggesting what the maximum number of PRI calls (incoming and outgoing) without hardware echo cancellation can be handled on a single box is? I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). I've looked at the number of simultaneous calls at each of these points, and each time the span seems to have around 21-23 calls, and the total number of calls ranges between 47 and 53. I'm trying to figure out if this is a load issue or an issue on the provider side, though my provider says they do not see any errors on any of the T1s. Could this be some sort of hardware interrupt problem? If so, how can I check? The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT) 4GB memory. Running asterisk 1.4.26.3 (32-bit) with libpri-1.4.7 and zaptel-1.4.12.9 Thanks. -- James Please CC me on responses. [Mar 22 09:45:00] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 down [Mar 22 09:45:00] WARNING[8887] chan_dahdi.c: No D-channels available! Using Primary channel 48 as D-channel anyway! [Mar 22 09:45:00] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 up [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:44:36] NOTICE[] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 [Mar 22 10:45:44] NOTICE[8886] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Mar 22 10:59:33] NOTICE[8887] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Mar 22 11:30:53] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 11:30:53] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 11:30:53] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 15:34:28] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 down [Mar 22 15:34:28] WARNING[8887] chan_dahdi.c: No D-channels available! Using Primary channel 48 as D-channel anyway! [Mar 22 15:34:28] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 up -- _ -- 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] Polycom DHCP Option 66 FTP provisioning
This works for me using DNSMasq: dhcp-host=00:04:f2:*:*:*,net:polycom # creates a 'polycom' group for all equipment with MAC prefix of 0004f2 dhcp-range=net:polycom,192.168.1.151,192.168.1.180 # dhcp range for 'polycom' group dhcp-option=net:polycom,66,http://pbxserver/gui/phoneprov; # polycom bootserver HTTP URL is asterisk provisioning URL. I assume an FTP URL can work fine. -- _ -- 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] Mobile phone shut down, but Queue() Ring as usual
2010/3/24 Alyed al...@vivoxie.com: Try the same as in http://lists.digium.com/pipermail/asterisk-users/2010-March/246316.html just make sure to add this in the [channels] context ;) Thanks for your answer! but i am not use zaptel device card to connect to PSTN but use Mediant 2000 - AudioCodes to connect to PSTN. so i can't change Zapata.conf file to add something like busydetect=yes busycount=3 In this situation how should i do? Hope it helps. Alyed 2010/3/23 Zhang Shukun bit...@gmail.com hi, all i use Queue() to call a Mobile phone, there is only one mobile phone in the queue. even if the mobile phone shut down, Queue() is ring in the cli verbose as mobile phone is normally working. what i want to see is if the mobile phone is shut down. queue() will end immediately to tell on one in the queue. is there any method to do this ? -- Best regards, Sucan -- _ -- 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 -- Best regards, Sucan -- _ -- 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] Maximum number of PRI calls on 1 asterisk box (no HW echo)
In theory asterisk can handle 8 spans, which means 192 concurrent calls if all spans are T1. In reality after 6 spans it starts giving problems. I recently shipped a server to a client in Taiwan, and before shipment it was put under a load of 100 plus concurrent calls (zap channels) for two whole weeks without any break. It had 6 spans, Rhino T1, FXO and FXS cards all with on board EC. The server's processors combined were hardly 7% in use and no call quality issues. For the same client another server with 9 spans were causing asterisk to freeze once there are more than 96 calls, i.e. it going to 1st span of the second T1 card, which made it 9th span considering FXO and FXS spans, but server load remained low as usual. So I think it is not your T1 card but some software/driver issue. Did you upgrade anything recently on this server? -- Zeeshan A Zakaria On 2010-03-24 8:49 PM, James Lamanna jlama...@gmail.com wrote: Hi, Does anyone have any good empirical data suggesting what the maximum number of PRI calls (incoming and outgoing) without hardware echo cancellation can be handled on a single box is? I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). I've looked at the number of simultaneous calls at each of these points, and each time the span seems to have around 21-23 calls, and the total number of calls ranges between 47 and 53. I'm trying to figure out if this is a load issue or an issue on the provider side, though my provider says they do not see any errors on any of the T1s. Could this be some sort of hardware interrupt problem? If so, how can I check? The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT) 4GB memory. Running asterisk 1.4.26.3 (32-bit) with libpri-1.4.7 and zaptel-1.4.12.9 Thanks. -- James Please CC me on responses. [Mar 22 09:45:00] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 down [Mar 22 09:45:00] WARNING[8887] chan_dahdi.c: No D-channels available! Using Primary channel 48 as D-channel anyway! [Mar 22 09:45:00] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 up [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:44:36] NOTICE[] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 [Mar 22 10:45:44] NOTICE[8886] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Mar 22 10:59:33] NOTICE[8887] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Mar 22 11:30:53] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 11:30:53] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 11:30:53] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 15:34:28] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 down [Mar 22 15:34:28] WARNING[8887] chan_dahdi.c: No D-channels available! Using Primary channel 48 as D-channel anyway! [Mar 22 15:34:28] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 up -- _ -- 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] Maximum number of PRI calls on 1 asterisk box (no HW echo)
I'm running 2 Sangoma A104D cards and a Sangoma A102D card for a total of 10 PRIs. Right now I have 9 PRIs configured as 3 DAHDI groups. 3 PRIs are in a group for local calls, 2 are grouped for long distance, and the last 4 connect to our Toshiba PBX for users still on digital phones. These cards are in a Dell R710 with two quad core CPU's. System load hovers just below 1 with 30-40 concurrent calls. Half the calls are to SIP phones connected over OpenVPN running on the server. The other half are DAHDI to DAHDI bridged to the PBX. Software wise I am running CentOS 5.4 x86_64, Asterisk 1.6.1.18, DAHDI 2.2.1, libPRI 1.4.10.2, and Sangoma wanpipe driver 3.5.6. I had some issues with newer wanpipe drivers and kernel soft locks. I also had a PCI dma timeout issue which required a Sangoma firmware update. Since then it has been rock solid since with 22 days of uptime. Ryan On Wed, Mar 24, 2010 at 9:53 PM, Zeeshan Zakaria zisha...@gmail.com wrote: In theory asterisk can handle 8 spans, which means 192 concurrent calls if all spans are T1. In reality after 6 spans it starts giving problems. I recently shipped a server to a client in Taiwan, and before shipment it was put under a load of 100 plus concurrent calls (zap channels) for two whole weeks without any break. It had 6 spans, Rhino T1, FXO and FXS cards all with on board EC. The server's processors combined were hardly 7% in use and no call quality issues. For the same client another server with 9 spans were causing asterisk to freeze once there are more than 96 calls, i.e. it going to 1st span of the second T1 card, which made it 9th span considering FXO and FXS spans, but server load remained low as usual. So I think it is not your T1 card but some software/driver issue. Did you upgrade anything recently on this server? -- Zeeshan A Zakaria On 2010-03-24 8:49 PM, James Lamanna jlama...@gmail.com wrote: Hi, Does anyone have any good empirical data suggesting what the maximum number of PRI calls (incoming and outgoing) without hardware echo cancellation can be handled on a single box is? I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). I've looked at the number of simultaneous calls at each of these points, and each time the span seems to have around 21-23 calls, and the total number of calls ranges between 47 and 53. I'm trying to figure out if this is a load issue or an issue on the provider side, though my provider says they do not see any errors on any of the T1s. Could this be some sort of hardware interrupt problem? If so, how can I check? The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT) 4GB memory. Running asterisk 1.4.26.3 (32-bit) with libpri-1.4.7 and zaptel-1.4.12.9 Thanks. -- James Please CC me on responses. [Mar 22 09:45:00] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 down [Mar 22 09:45:00] WARNING[8887] chan_dahdi.c: No D-channels available! Using Primary channel 48 as D-channel anyway! [Mar 22 09:45:00] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 up [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 09:59:23] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 09:59:23] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 10:36:11] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 10:36:11] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 10:44:36] NOTICE[] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 [Mar 22 10:45:44] NOTICE[8886] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Mar 22 10:59:33] NOTICE[8887] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Mar 22 11:30:53] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 down [Mar 22 11:30:53] WARNING[8886] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Mar 22 11:30:53] VERBOSE[8886] logger.c: == Primary D-Channel on span 1 up [Mar 22 15:34:28] VERBOSE[8887] logger.c: == Primary D-Channel on span 2 down [Mar 22 15:34:28]
Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)
A good idea is to call sangoma tech support or email them with your config but as per my knowledge and experience and as per Rhino's tech support and Digium techsupport suggestions which I got when dealing with a lot of spans, you can't have 10 PRIs running under one asterisk. You'll need to physically remove one 4xPRI card, and set it up on a separate server, and configure your dialplan accordingly. On 2010-03-24 11:14 PM, Ryan Wagoner rswago...@gmail.com wrote: I'm running 2 Sangoma A104D cards and a Sangoma A102D card for a total of 10 PRIs. Right now I have 9 PRIs configured as 3 DAHDI groups. 3 PRIs are in a group for local calls, 2 are grouped for long distance, and the last 4 connect to our Toshiba PBX for users still on digital phones. These cards are in a Dell R710 with two quad core CPU's. System load hovers just below 1 with 30-40 concurrent calls. Half the calls are to SIP phones connected over OpenVPN running on the server. The other half are DAHDI to DAHDI bridged to the PBX. Software wise I am running CentOS 5.4 x86_64, Asterisk 1.6.1.18, DAHDI 2.2.1, libPRI 1.4.10.2, and Sangoma wanpipe driver 3.5.6. I had some issues with newer wanpipe drivers and kernel soft locks. I also had a PCI dma timeout issue which required a Sangoma firmware update. Since then it has been rock solid since with 22 days of uptime. Ryan On Wed, Mar 24, 2010 at 9:53 PM, Zeeshan Zakaria zisha...@gmail.com wrote: In theory asterisk c... -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] Music class default requested but no musiconhold loaded
Hi i have the problem that music on hold is not working at all. on the CLI i get the message Music class default requested but no musiconhold loaded thanks lance 2010-03-25 675842709 -- _ -- 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] How to get Sip response codes in Dialplan?
hi ,all when a Dial or Queue excutes, a sip response code will return. like == Using SIP RTP CoS mark 5 -- Got SIP response 502 Bad Gateway back from 211.150.119.32 -- SIP/95040-004a is circuit-busy -- Nobody picked up in 2000 ms My quesion is how to get the response code in the dial plan immediatelly in order to do different thing according the returned codes? for example: a queue response code is busy now i will queue another number immediately not let the user waiting for the timeout. Thanks! -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Sip response codes in Dialplan?
Try using DIALSTATUS. --Mensaje original-- De: Zhang Shukun Remitente: asterisk-users-boun...@lists.digium.com Para: Asterisk Users Mailing List - Non-Commercial Discussion Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] How to get Sip response codes in Dialplan? Enviado: 24 Mar, 2010 23:29 hi ,all when a Dial or Queue excutes, a sip response code will return. like == Using SIP RTP CoS mark 5 -- Got SIP response 502 Bad Gateway back from 211.150.119.32 -- SIP/95040-004a is circuit-busy -- Nobody picked up in 2000 ms My quesion is how to get the response code in the dial plan immediatelly in order to do different thing according the returned codes? for example: a queue response code is busy now i will queue another number immediately not let the user waiting for the timeout. Thanks! -- Best regards, Sucan -- _ -- 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 Saludos, Juan E. Rodríguez -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Sip response codes in Dialplan?
2010/3/25 Juan E. Rodríguez jerdg...@gmail.com: Try using DIALSTATUS. Thank you! but DIALSTATUS IS used for Dial. not for queue --Mensaje original-- De: Zhang Shukun Remitente: asterisk-users-boun...@lists.digium.com Para: Asterisk Users Mailing List - Non-Commercial Discussion Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] How to get Sip response codes in Dialplan? Enviado: 24 Mar, 2010 23:29 hi ,all when a Dial or Queue excutes, a sip response code will return. like == Using SIP RTP CoS mark 5 -- Got SIP response 502 Bad Gateway back from 211.150.119.32 -- SIP/95040-004a is circuit-busy -- Nobody picked up in 2000 ms My quesion is how to get the response code in the dial plan immediatelly in order to do different thing according the returned codes? for example: a queue response code is busy now i will queue another number immediately not let the user waiting for the timeout. Thanks! -- Best regards, Sucan -- _ -- 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 Saludos, Juan E. Rodríguez -- _ -- 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, Sucan -- _ -- 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
