Re: [asterisk-users] MeetMe echo problems with more than twoparticipants
Hi Danny, I will try as you suggested. Thanks Alessandro R. On Thu, Dec 11, 2008 at 9:51 PM, Danny Nicholas da...@debsinc.com wrote: If callers need to just listen, you could run meetme with the –l mode. Otherwise, you might try the –o mode (optimize, mute non-talker) or –m (set initially muted). -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Alessandro Russo *Sent:* Thursday, December 11, 2008 2:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] MeetMe echo problems with more than twoparticipants Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference http://sourceforge.net/projects/appconference/ ?? THX Bye Alessandro R. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables for dial plan
Use setvar=variablename=value
Eg: under [client1]
setvar=dialplan=NZ
Then just reference ${dialplan} in your extensions.conf
Cheers
Andy
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Michael
-- Sent: 15 December 2008 04:36
-- To: asterisk-users@lists.digium.com
-- Subject: [asterisk-users] Variables for dial plan
--
-- I want to have a arbitary named variable within the client's user
-- details in
-- sip.conf
--
-- [client1]
-- dialplan=NZ
-- ..
--
-- In extensions.conf (Logic expressed using PHP style)
--
-- if ($dialplan == NZ) {
-- $NAT = 0;
-- $INT = 00;
-- };
--
-- and in the [outgoing] section
--
-- ; Australia
-- exten = _${INT}61[278]NXX.,1,Set(CDR(UserField)=AUSTRALIA)
-- exten =
_${INT}61[278]NXX.,n,Dial(SIP/SIP_PROVIDER/0${EXTEN:4:9})
--
-- How can I implement this in Asterisk style?
--
-- Thanks,
--
-- Michael
--
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Re: [asterisk-users] MeetMe echo problems with more than two participants
Hi to all, Unfortunately echo is not due to speakerphone. Each participant calls a geographical number that is redirected from the PBX to a call manager which pass the flow to the asterisk machine which creates a meetme voice conference, so user calls via traditional either fixed or mobile phone. Therefore they cannot mute their phone while they aren't speak :( Moreover the echo problem occurs when we do tests within the same phone-cloud, in our organization phones are connected through some cisco call managers, so when a phone calls the internal number ABCD the flow arrives to the call manger which forward it to the asterisk, this is the path done: phone = call manager = asterisk and also in internal cloud we experienced echo problems with more than 2 participants, not all the conversation is affected by echo, sometimes there is echo and sometimes not. I performed the zttest and I obtained the following results: asterisk:~# zttest Opened pseudo zap interface, measuring accuracy... 99.966690% 99.971863% 99.936729% 99.967766% 99.936913% 99.968163% 99.967667% 99.936623% 99.969818% 99.937019% 99.967972% 99.937012% 99.968063% 99.967865% 99.936440% 99.967766% 99.935356% 99.967667% 99.937401% 99.968460% 99.967667% 99.936333% --- Results after 22 passes --- Best: 99.972 -- Worst: 99.935 -- Average: 99.955330, Difference: 99.992836 Any suggestions? Alessandro R. On Fri, Dec 12, 2008 at 7:39 PM, Matthew J. Roth mr...@imminc.com wrote: Alessandro Russo wrote: we are using Asterisk 1.4.18.1 http://1.4.18.1/ on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference http://sourceforge.net/projects/appconference/ ?? Alessandro, Are you certain that the echo isn't being introduced by someone on the conference using a speakerphone? This would cause what is known as acoustic echo http://en.wikipedia.org/wiki/Echo_cancellation#Acoustic_echo and it's always my first suspect in a situation like the one you are describing. This is not a problem that is specific to Asterisk and I'm fairly certain there is nothing that can be done within your configuration to correct it. Instructing the conference participants to mute their phones when they aren't speaking or to use their handsets should reduce acoustic echo. Some phones http://www.voip-info.org/wiki/view/Uni-Ta+Technology also claim to have a full-duplex speakerphone with advanced acoustic echo cancellation, but caveat emptor. That said, I'm not an expert on echo cancellation and I have an installation where the users are making similar complaints about echo during conference calls. I'd greatly appreciate it if anyone on the list corrected any misunderstandings that I might have on the subject. As an aside, how is the timing on your conference server. The MeetMe application relies on it to mix the audio in conferences. You should get at least 99.98% output from zttest (as shown below) or the audio quality will suffer. This is an overall quality issue and is not necessarily related to your echo problems. [r...@astconf ~]# zttest Opened pseudo zap interface, measuring accuracy... 99.999413% 99.995407% 99.995499% 99.998047% 99.996483% 99.997849% 99.999008% ... --- Results after 107 passes --- Best: 100.000 -- Worst: 99.995 -- Average: 99.997687, Difference: 99.997815 Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1: iax trunk needs dahdi timing ??
On 20:24, Sun 14 Dec 08, sean darcy wrote: starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? If you dont have any dahdi hardware installed and configured, make sure to load dahdi_dummy. That will provide you the timers. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables for dial plan
On Mon, 15 Dec 2008 21:31:56 you wrote:
Use setvar=variablename=value
Eg: under [client1]
setvar=dialplan=NZ
Then just reference ${dialplan} in your extensions.conf
Cheers
Andy
Thanks, now how do I achieve the following logic?
if ($dialplan == NZ) {
$NAT = 0;
$INT = 00;
};
Michael
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Re: [asterisk-users] Country numbering plan resources
For informational purposes many people find ITU's web site useful, although not always as detailed as one would probably want: http://www.itu.int/itu-t/inr/nnp/index.html It even has event dates of official numbering plan changes. Best regards, Vlasis Hatzistavrou Kinetix Tele.com International Inc. 306 Victoria House, Victoria, Mahe, Seychelles Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: vh...@kinetixtele.com http://www.kinetixtele.com Postal address: Monastiriou 9 Enotikon 54627 Thessaloniki Greece ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
On Monday 15 December 2008 00:57:08 Langdon Stevenson wrote: Hi Paul Thanks for the reply. I have removed and re-installed all of the Fedora Zaptel packages with Yum. I have the following installed: asterisk-zaptel 1.4.12.1-1.fc8 zaptel.i386 1.4.12.1-1.fc8 zaptel-devel.i386 1.4.12.1-1.fc8 zaptel-lib.i386 1.4.12.1-1.fc8 zaptel-utils.i386 1.4.12.1-1.fc8 The command: modprobe wctdm produces: FATAL: Module wctdm not found. This probably means that the modules were compiled for a kernel other than the one you have installed. You probably have multiple directories within /lib/modules, and the zaptel modules are in a directory other than what is listed with 'uname -r'. In this case, compiling from source is probably your best bet. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe echo problems with more than two participants
Alessandro Russo wrote: Unfortunately echo is not due to speakerphone. Each participant calls a geographical number that is redirected from the PBX to a call manager which pass the flow to the asterisk machine which creates a meetme voice conference, so user calls via traditional either fixed or mobile phone. Therefore they cannot mute their phone while they aren't speak :( Moreover the echo problem occurs when we do tests within the same phone-cloud, in our organization phones are connected through some cisco call managers, so when a phone calls the internal number ABCD the flow arrives to the call manger which forward it to the asterisk, this is the path done: phone = call manager = asterisk and also in internal cloud we experienced echo problems with more than 2 participants, not all the conversation is affected by echo, sometimes there is echo and sometimes not. I performed the zttest and I obtained the following results: asterisk:~# zttest Opened pseudo zap interface, measuring accuracy... 99.966690% 99.971863% 99.936729% 99.967766% 99.936913% 99.968163% 99.967667% 99.936623% 99.969818% 99.937019% 99.967972% 99.937012% 99.968063% 99.967865% 99.936440% 99.967766% 99.935356% 99.967667% 99.937401% 99.968460% 99.967667% 99.936333% --- Results after 22 passes --- Best: 99.972 -- Worst: 99.935 -- Average: 99.955330, Difference: 99.992836 Alessandro, I'm sorry to hear that your problem isn't acoustic echo. I'll be following this thread to see if anyone offers you any suggestions and I'll let you know if I discover anything that improves the echo problem in my installation. What is the timing source in the conference server? In general, it will be either a Zaptel/DAHDI hardware device or the ztdummy/dahdi-dummy module. See this page http://www.voip-info.org/wiki/view/Asterisk+timer for details. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
This appears to be the case. If someone else know how, please feel free to share. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, December 15, 2008 7:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 12, 2008 15:31 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking After some research, it seems that asterisk builds a dynamic context called [park-dial] and puts a callback for the parker into line 1, so this would have to be a patch/workaround. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, December 12, 2008 12:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking Danny, I've been starring at features.conf since yesterday AM, and I do realize there is an example that looks close to what I want, but the same thing typed in my own dialplan doesn't work. All I want, for the sake of discussion, is to Hangup() when the call gets out of parking after the 45 second timeout. As for show application park, this is not helping. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 12, 2008 9:26 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking You should try these steps 1. core show application park from the CLI interface 2. look at features.conf 3. one of these should offer the hint you seek _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 10:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Follow up on parking I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context. [internal-local-only-hamel] exten = s,1,Hangup include = parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 15 seconds Seems like this will work.until it doesn't. The s,1 extension is never executed, instead park-dial() is called. What am I missing? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables for dial plan
One of these methods will work:
exten = s,n,ExecIf($[${dialplan} = NZ]|Set|NAT=0)
exten = s,n,ExecIf($[${dialplan} = NZ]|Set|INT=00)
-or-
exten = s,n,GotoIf($[${dialplan} != NZ]?not-nz)
exten = s,n,Set(NAT=0)
exten = s,n,Set(INT=00)
exten = s,n(not-nz),more_dialplan_stuff
On Mon, Dec 15, 2008 at 3:26 AM, Michael mich...@networkstuff.co.nz wrote:
On Mon, 15 Dec 2008 21:31:56 you wrote:
Use setvar=variablename=value
Eg: under [client1]
setvar=dialplan=NZ
Then just reference ${dialplan} in your extensions.conf
Cheers
Andy
Thanks, now how do I achieve the following logic?
if ($dialplan == NZ) {
$NAT = 0;
$INT = 00;
};
Michael
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Re: [asterisk-users] Follow up on parking
Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 12, 2008 15:31 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking After some research, it seems that asterisk builds a dynamic context called [park-dial] and puts a callback for the parker into line 1, so this would have to be a patch/workaround. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, December 12, 2008 12:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking Danny, I've been starring at features.conf since yesterday AM, and I do realize there is an example that looks close to what I want, but the same thing typed in my own dialplan doesn't work. All I want, for the sake of discussion, is to Hangup() when the call gets out of parking after the 45 second timeout. As for show application park, this is not helping. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 12, 2008 9:26 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking You should try these steps 1. core show application park from the CLI interface 2. look at features.conf 3. one of these should offer the hint you seek _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 10:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Follow up on parking I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context [internal-local-only-hamel] exten = s,1,Hangup include = parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 15 seconds Seems like this will work until it doesn't. The s,1 extension is never executed, instead park-dial() is called. What am I missing? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
if you write you own [park-dial] context maybe asterisk dont over write it... David 2008/12/15 Danny Nicholas da...@debsinc.com This appears to be the case. If someone else know how, please feel free to share. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mike *Sent:* Monday, December 15, 2008 7:05 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Follow up on parking Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Mike *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Friday, December 12, 2008 15:31 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Follow up on parking After some research, it seems that asterisk builds a dynamic context called [park-dial] and puts a callback for the parker into line 1, so this would have to be a patch/workaround. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mike *Sent:* Friday, December 12, 2008 12:40 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Follow up on parking Danny, I've been starring at features.conf since yesterday AM, and I do realize there is an example that looks close to what I want, but the same thing typed in my own dialplan doesn't work. All I want, for the sake of discussion, is to Hangup() when the call gets out of parking after the 45 second timeout. As for show application park, this is not helping. Regards, Mike *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Friday, December 12, 2008 9:26 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Follow up on parking You should try these steps 1. core show application park from the CLI interface 2. look at features.conf 3. one of these should offer the hint you seek -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mike *Sent:* Thursday, December 11, 2008 10:46 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Follow up on parking I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context… [internal-local-only-hamel] exten = s,1,Hangup include = parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 15 seconds Seems like this will work…until it doesn't. The s,1 extension is never executed, instead park-dial() is called. What am I missing? Regards,** * * *Mike* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1: iax trunk needs dahdi timing ??
Michiel van Baak wrote: On 20:24, Sun 14 Dec 08, sean darcy wrote: starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? If you dont have any dahdi hardware installed and configured, make sure to load dahdi_dummy. That will provide you the timers. In 1.6.1, this should not be required. It's probalby a check in the code that shouldn't be there anymore. If you post this on bugs.digium.com, I'll remove it. -- Russell Bryant Digium, Inc. | Senior Software Engineer, Open Source Team Lead 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
On Dec 15, 2008, at 7:05 AM, Mike wrote: Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Yes. I'm working on an issue right now related to parking and noticed that Asterisk completely lies with the verbose statement saying that it will time back out to an extension. There is an if/else that checks a string that will always be set and therefore will never hit the else...which is where the code is that would time back out to an extension as opposed to trying to magically find the original caller and call the channel back. It is fairly complex code in there, so it may take a bit to fix...but I thought I'd let your know that I am working on it, anyway. Terry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] We think we are cpe but they think they are cpe too
Just for records. I have got red light on Rx loss on keymile pri modem
because
I did not make good pinout on my cable. Right pinout is 1 2 4 5 and 1 2 4 5
(4 wires)
on both Rj-45 conncectors not 1 2 4 5 and 4 5 1 2 as I made. After I made
cable
right red light disappeared.
Other problem was that telco did not route calls on my isdn number.After
they made
necessary changes I can receive calls. Thanks for your help.
I have few more questions for you:
1) Zap show status gave me IRQ 1 (irq misses 1) . Is it a problem ? TE121B
card is alone on
interupt now (I moved it to another PCI express slot)
2) After restart of asterisk (not always) from time to time I can see
following error message in log file:
ERROR chan_zap.c got S-frame while link down
3) Finnaly afeter restarting of zaptel (not always) I can see following
error after which I can only power of server.
BUG: soft lockup CPU#3 stuck for 11 s! [swapper:0]
[rmmod:5305]
soft lockup CPU#0 stuck for 11 s! [swapper:0]
[rmmod:5305]
status: {DRDY} ata4.0 revalidation failed errno=s
e mask 0x0 SAcT 0x0
SErr 0x0 action 0x2 frozen
After restart everything is fine again.
I hope this is not connected with known issue about this cards (shipped
before November 5th 2008 mine was shipped on September 15th) I've read on
Digium forum (freeze some Dell servers Digium called everyone who has this
issue to change card.Cause was voltage)
Thanks again,
Uros Djokic
--
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
2) To copy program
3) To modify source code
4) To redistribute modified program under condition that new user has all 4
freedoms.
Richard M. Stallman
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[asterisk-users] UDPTL setup
This setting here- ; UDPTL start and UDPTL end configure start and end addresses ; udptlstart=4000 udptlend=4999 Does this need to be allowed on incoming or outgoing firewall rulesets of the machine running Asterisk? Thanks in anticipation, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] D-channel errors and Channelbanks
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've been seeing some strange issues recently as it relates to our Asterisk phone system. Let me give a little background on things: We have a T1 provided by Knology. We recently had an outage of Internet and phones. The following problems (except for the log messages) seemed to have cropped up after the outage. I was prompted by a friend of mine to power cycle all the equipment detailed below (not sure if that's relevant.) T1 Ditech Echo Canceler Asterisk server asterisk1 Zhone Z-plex 10 Fax channel bank fax devices T1 Ditech Echo Canceler Zhone Z-plex 10 Voice channel bank voice devices. Now, let me explain the symptoms of the problem. * We have a Pitney Bowes Postage meter attached to the fax channel bank. It is not able to retrieve postage. It received a dial-tone, and attempts to dial. It does not connect. * A regular phone connected to the voice channel bank receives a dial tone, but no number tones are recognized. (when you attempt to dial a number, the dial tone stays constant) * A fax machine attached to the fax channel bank has problems sending and receiving faxes. These problems are intermittent. It is not related to the fax machine itself because the machine has been replaced with a new device. * Various errors in /var/log/asterisk/messages, as detailed below. Dec 15 03:38:11 NOTICE[5077] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 15 07:44:06 NOTICE[5077] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 15 08:21:41 NOTICE[5077] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 15 10:37:18 NOTICE[5077] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 There is a much longer history of these errors attached to this email, and the attachment includes a wider variety of messages. The file is the result of grep -i d-channel on an archive of asterisk message logs. # cat /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs fxoks=49-72 span=4,0,0,esf,b8zs fxoks=73-96 loadzone = us defaultzone = us # cat /proc/interrupts CPU0 CPU1 0: 245968510 245928679IO-APIC-edge timer 1:157142IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 11: 0 0 IO-APIC-level ohci_hcd 12: 62 4IO-APIC-edge i8042 14:22071972216054IO-APIC-edge ide2 177:1875788465 IO-APIC-level eth0 185: 37433273 0 IO-APIC-level eth1 193:11246691127450 IO-APIC-level ide0, ide1 201: 245939300 245915557 IO-APIC-level wct4xxp NMI: 0 0 LOC: 491977277 491977276 ERR: 24 MIS: 0 # lspci 00:00.0 Host bridge: Broadcom GCNB-LE Host Bridge (rev 32) 00:00.1 Host bridge: Broadcom GCNB-LE Host Bridge 00:03.0 Ethernet controller: Intel Corporation 82546EB Gigabit Ethernet Controller (Copper) (rev 01) 00:03.1 Ethernet controller: Intel Corporation 82546EB Gigabit Ethernet Controller (Copper) (rev 01) 00:04.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) 00:05.0 IDE interface: Silicon Image, Inc. PCI0680 Ultra ATA-133 Host Controller (rev 02) 00:0f.0 Host bridge: Broadcom CSB6 South Bridge (rev a0) 00:0f.1 IDE interface: Broadcom CSB6 RAID/IDE Controller (rev a0) 00:0f.2 USB Controller: Broadcom CSB6 OHCI USB Controller (rev 05) 00:0f.3 ISA bridge: Broadcom GCLE-2 Host Bridge 00:10.0 Host bridge: Broadcom CIOB-E I/O Bridge with Gigabit Ethernet (rev 12) 00:10.2 Host bridge: Broadcom CIOB-E I/O Bridge with Gigabit Ethernet (rev 12) 01:03.0 Communication controller: Xilinx Corporation Wildcard TE405P/TE410P (1st Gen) (rev 01) I've searched the mailing lists on these topics, and most messages say the d-channel errors relate to IRQ problems on the server. I'm not sure how to go about solving this issue. I'm also not sure if the d-channel problems are related to the channel bank problems we are having. I'm new to asterisk as a whole, and I've inherited this system sans documentation. Any assistance would be greatly appreciated. Please let me know if there is any additional information you might need. - -- Regards, Justin Phelps IT Director Eye Center of North Florida 850.522.7952 - Office 850.522.9829 - Fax 850.832.0249 - Cell www.eyecarenow.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAklGkNwACgkQfV4idVrZqT9pJQCglS1Ra/8yhv5OclWeQbCleBie iAkAnjAom9iJcRKcelRT78X9hgw6Q98J =3lPQ -END PGP SIGNATURE- 550 551 552 553 Nov 5 14:24:13 NOTICE[5067] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Nov 5 14:24:14 NOTICE[5067] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Nov 5 14:24:14
Re: [asterisk-users] Follow up on parking
Nope. If you write your own PD, asterisk just inserts it's own call back into line 1 and moves all of your code down Example Dialplan show park-dial Before park [ Context 'park-dial' created by 'pbx_config' ] 's' =1. Background(vm-goodbye) [pbx_config] 2. HangUp() [pbx_config] After park [ Context 'park-dial' created by 'pbx_config' ] 's' =1. Dial(SIP/XXX,XX) 2. Background(vm-goodbye) [pbx_config] 3. HangUp() [pbx_config] _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Monday, December 15, 2008 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Follow up on parking if you write you own [park-dial] context maybe asterisk dont over write it... David 2008/12/15 Danny Nicholas da...@debsinc.com This appears to be the case. If someone else know how, please feel free to share. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, December 15, 2008 7:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 12, 2008 15:31 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking After some research, it seems that asterisk builds a dynamic context called [park-dial] and puts a callback for the parker into line 1, so this would have to be a patch/workaround. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, December 12, 2008 12:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking Danny, I've been starring at features.conf since yesterday AM, and I do realize there is an example that looks close to what I want, but the same thing typed in my own dialplan doesn't work. All I want, for the sake of discussion, is to Hangup() when the call gets out of parking after the 45 second timeout. As for show application park, this is not helping. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 12, 2008 9:26 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Follow up on parking You should try these steps 1. core show application park from the CLI interface 2. look at features.conf 3. one of these should offer the hint you seek _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, December 11, 2008 10:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Follow up on parking I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context. [internal-local-only-hamel] exten = s,1,Hangup include = parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 1...@parkedcalls. Will timeout back to extension [internal-local-only-hamel] s, 1 in 15 seconds Seems like this will work.until it doesn't. The s,1 extension is never executed, instead park-dial() is called. What am I missing? Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
Tilghman Lesher wrote: On Monday 15 December 2008 00:57:08 Langdon Stevenson wrote: Hi Paul Thanks for the reply. I have removed and re-installed all of the Fedora Zaptel packages with Yum. I have the following installed: asterisk-zaptel 1.4.12.1-1.fc8 zaptel.i386 1.4.12.1-1.fc8 zaptel-devel.i386 1.4.12.1-1.fc8 zaptel-lib.i386 1.4.12.1-1.fc8 zaptel-utils.i386 1.4.12.1-1.fc8 The command: modprobe wctdm produces: FATAL: Module wctdm not found. This probably means that the modules were compiled for a kernel other than the one you have installed. You probably have multiple directories within /lib/modules, and the zaptel modules are in a directory other than what is listed with 'uname -r'. In this case, compiling from source is probably your best bet. This may be an obvious thing, but you didn't mention checking whether or not the card was still seated in the slot properly after the move. I know from experience that when you move offices, even if you take all the precautions possible, a card can get bumped just enough to jostle the connections loose. Even if the card appears to be seated correctly I'd take it out and re-seat it. Unfortunately it looks like you may have compounded the problem by removing and reinstalling the zaptel packages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
That information is very much appreciated. Thank you. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson Sent: Monday, December 15, 2008 11:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Follow up on parking On Dec 15, 2008, at 7:05 AM, Mike wrote: Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Yes. I'm working on an issue right now related to parking and noticed that Asterisk completely lies with the verbose statement saying that it will time back out to an extension. There is an if/else that checks a string that will always be set and therefore will never hit the else...which is where the code is that would time back out to an extension as opposed to trying to magically find the original caller and call the channel back. It is fairly complex code in there, so it may take a bit to fix...but I thought I'd let your know that I am working on it, anyway. Terry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dedicated Fax Line
Hello folks, I have a 20 channel fractional PRI and I would like to dedicate one of the lines for a Fax service (in and outbound). Is this possible with Asterisk and what conf would I need for that? Thanks, -JE ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ALG SIP
Hello everybody, I want to ask you if Asterisk can resolve problems of openning dynamically RTP port through firewall, or resolving NAT traversal for the protocol SIP. Thanks for information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dedicated Fax Line
2008/12/15 Johnny Edge je...@visafirst.com Hello folks, I have a 20 channel fractional PRI and I would like to dedicate one of the lines for a Fax service (in and outbound). Do you imply casual incoming calls not to be answered, to be replied a busy tone or to deflected elsewhere ? How do you expect inbound faxes to be treated ? Switched to an analog fax machine ? E-mailed ? Is this possible with Asterisk and what conf would I need for that? Thanks, -JE ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tcpdum
*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: *Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re : tcpdum
Hi, The TTL is the time to live of the IP paquet. But I think that what interests you is the SIP_TIMEOUT which is by default 3600 seconds. but you can modify it. De : michel freiha mich...@gmail.com À : Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; asterisk-users-boun...@lists.digium.com Envoyé le : Lundi, 15 Décembre 2008, 21h35mn 01s Objet : [asterisk-users] tcpdum Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
Dear Sir, There is no relation between TTL and the latency on asterisk server? Regards On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere j...@jeff.net wrote: TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: *Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
No. TTL in the header is about hop traversal. Each IP router that forwards the packet will reduce this number in the live packet until it reaches zero, when it will be dropped. I believe this is to eliminate route loops creating packet storms. FWIW this is how traceroute works - it sends out packets with continually increasing TTLs and the router that drops the packet will send back a notification, so you can trace each hop... What is it you are trying to do or measure? j On Mon, 15 Dec 2008, michel freiha wrote: Dear Sir, There is no relation between TTL and the latency on asterisk server? Regards On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere j...@jeff.net wrote: TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: *Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Question
Hi, In queues realtime, when the queue start and when it ends. I mean, for example to calculate service level, how many calls, etc. If I want to start the queue from with 0 calls, etc, how do I do this? And if I want to stop it, so I can start it again?? Thanks!! Regards, Sebastian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dedicated Fax Line
Sorry I didn't clarify more. I have one number for fax and 9 more for regulars calls, all of them terminated on the same 20 chan PRI. When there 20 active calls I can't send/recv faxes. Inbound faxes are sent to e-mail. I wish to make sure the fax line is separate and is not used for anything else but faxes. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 15 December 2008 22:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dedicated Fax Line 2008/12/15 Johnny Edge je...@visafirst.com Hello folks, I have a 20 channel fractional PRI and I would like to dedicate one of the lines for a Fax service (in and outbound). Do you imply casual incoming calls not to be answered, to be replied a busy tone or to deflected elsewhere ? How do you expect inbound faxes to be treated ? Switched to an analog fax machine ? E-mailed ? Is this possible with Asterisk and what conf would I need for that? Thanks, -JE ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dedicated Fax Line
Tell your PRI provider that you want one of those channels exclusively bound to your fax DID. Also, it should be removed from the normal hunt group where the rest of your calls come in. Then, the only way that PRI channel will be used is when someone calls your fax number/DID. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Johnny Edge je...@visafirst.com wrote: Sorry I didn't clarify more. I have one number for fax and 9 more for regulars calls, all of them terminated on the same 20 chan PRI. When there 20 active calls I can't send/recv faxes. Inbound faxes are sent to e-mail. I wish to make sure the fax line is separate and is not used for anything else but faxes. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 15 December 2008 22:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dedicated Fax Line 2008/12/15 Johnny Edge je...@visafirst.com Hello folks, I have a 20 channel fractional PRI and I would like to dedicate one of the lines for a Fax service (in and outbound). Do you imply casual incoming calls not to be answered, to be replied a busy tone or to deflected elsewhere ? How do you expect inbound faxes to be treated ? Switched to an analog fax machine ? E-mailed ? Is this possible with Asterisk and what conf would I need for that? Thanks, -JE ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
Terry Wilson wrote: On Dec 15, 2008, at 7:05 AM, Mike wrote: Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Yes. I'm working on an issue right now related to parking and noticed that Asterisk completely lies with the verbose statement saying that it will time back out to an extension. There is an if/else that checks a string that will always be set and therefore will never hit the else...which is where the code is that would time back out to an extension as opposed to trying to magically find the original caller and call the channel back. It is fairly complex code in there, so it may take a bit to fix...but I thought I'd let your know that I am working on it, anyway. I saw this in 1.2 as well. I don't know about 1.4, since my customers never used 1.4. Since all parked calls were supposed to be sent to the operator, it was not an issue for my customers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
Dear Sir, What I'm interested to is to know how much time the rtp packets takes from the time it access the asterisk server,to when it'll leave Is this function or variable exist anywhere? Regards On Mon, Dec 15, 2008 at 10:55 PM, Jeff LaCoursiere j...@jeff.net wrote: No. TTL in the header is about hop traversal. Each IP router that forwards the packet will reduce this number in the live packet until it reaches zero, when it will be dropped. I believe this is to eliminate route loops creating packet storms. FWIW this is how traceroute works - it sends out packets with continually increasing TTLs and the router that drops the packet will send back a notification, so you can trace each hop... What is it you are trying to do or measure? j On Mon, 15 Dec 2008, michel freiha wrote: Dear Sir, There is no relation between TTL and the latency on asterisk server? Regards On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere j...@jeff.net wrote: TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: *Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1: iax trunk needs dahdi timing ??
Russell Bryant wrote: Michiel van Baak wrote: On 20:24, Sun 14 Dec 08, sean darcy wrote: starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? If you dont have any dahdi hardware installed and configured, make sure to load dahdi_dummy. That will provide you the timers. In 1.6.1, this should not be required. It's probalby a check in the code that shouldn't be there anymore. If you post this on bugs.digium.com, I'll remove it. OK, it's http://bugs.digium.com/view.php?id=14082 BTW I do have a TDM400P with dahdi-2.1.0 installed and configured. So dahdi_dummy wouldn't help. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a problem on Ubuntu with Asterisk
Have a nice day, Scott Berry E-mail: n7...@northlc.com scott 6882 6797 0 15:39 pts/0 00:00:00 grep asterisk On Wed, 2008-12-10 at 09:11 -0600, Danny Nicholas wrote: You've checked that another asterisk is running (ps -ef|grep asterisk)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott Berry Sent: Wednesday, December 10, 2008 9:06 AM To: Asterisk Users Subject: [asterisk-users] a problem on Ubuntu with Asterisk Have a nice day, Scott Berry E-mail: n7...@northlc.com I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do asterisk -r and hook up to the asterisk CLI. I have checked to see that /var/run/asterisk/asterisk.ctl is available which it is. I have also set up the zaptel.conf, zapata.conf and also the extensions.conf as specified in the book. The error I get is: Unable to connect to asterisk remote (does /var/run/asterisk/asterisk.ctl exist? Yes it certainly does. Any help would be appreciated. if need be i would be happy to send my extensions.conf, zaptel.conf, and zapata.conf to the lisOne other question I think I am correct on this but not sure does zaptel.conf and zapata.conf go in to /etc? Thanks for all the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] D-channel errors and Channelbanks
was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081215/6c29a35e/attachment-0001.htm -- Message: 3 Date: Mon, 15 Dec 2008 07:16:06 -0600 From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com Subject: Re: [asterisk-users] Zaptel / TDM400P card stopped working To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 200812150716.06498.tilgh...@mail.jeffandtilghman.com Content-Type: text/plain; charset=iso-8859-1 On Monday 15 December 2008 00:57:08 Langdon Stevenson wrote: Hi Paul Thanks for the reply. I have removed and re-installed all of the Fedora Zaptel packages with Yum. I have the following installed: asterisk-zaptel 1.4.12.1-1.fc8 zaptel.i386 1.4.12.1-1.fc8 zaptel-devel.i386 1.4.12.1-1.fc8 zaptel-lib.i386 1.4.12.1-1.fc8 zaptel-utils.i386 1.4.12.1-1.fc8 The command: modprobe wctdm produces: FATAL: Module wctdm not found. This probably means that the modules were compiled for a kernel other than the one you have installed. You probably have multiple directories within /lib/modules, and the zaptel modules are in a directory other than what is listed with 'uname -r'. In this case, compiling from source is probably your best bet. I just realized I did something very silly by not giving an asterisk version or OS version. Connected to Asterisk 1.2.24 # cat /etc/issue CentOS release 4.4 (Final) Kernel \r on an \m - -- Regards, Justin Phelps IT Director Eye Center of North Florida 850.522.7952 - Office 850.522.9829 - Fax 850.832.0249 - Cell www.eyecarenow.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAklG0sIACgkQfV4idVrZqT/GVwCeMibhFlqb/+H/nlIpqiXu++Pn X8IAoJa0+sCDGwFjt/DjWvE4NqF9ReAH =Pz1z -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
That would help me, but I can't even do that (send all parked calls to anybody) because of the dynamic park-dial context. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric ManxPower Wieling Sent: Monday, December 15, 2008 16:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Follow up on parking Terry Wilson wrote: On Dec 15, 2008, at 7:05 AM, Mike wrote: Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Yes. I'm working on an issue right now related to parking and noticed that Asterisk completely lies with the verbose statement saying that it will time back out to an extension. There is an if/else that checks a string that will always be set and therefore will never hit the else...which is where the code is that would time back out to an extension as opposed to trying to magically find the original caller and call the channel back. It is fairly complex code in there, so it may take a bit to fix...but I thought I'd let your know that I am working on it, anyway. I saw this in 1.2 as well. I don't know about 1.4, since my customers never used 1.4. Since all parked calls were supposed to be sent to the operator, it was not an issue for my customers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
michel freiha wrote: Dear Sir, What I'm interested to is to know how much time the rtp packets takes from the time it access the asterisk server,to when it'll leave Is this function or variable exist anywhere? If you want statistics on RTP packets, then you should look into RTCP reporting. A simple facility for looking at this information would be the Asterisk CLI commands rtcp stats on and rtcp debug assuming that you are running Asterisk 1.4. If you are using Asterisk trunk, the commands are rtcp set stats on and rtcp set debug on. You may also be able to filter the RTCP packets in a program like wireshark and analyze them there as well. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Question
Sebastian wrote: Hi, In queues realtime, when the queue start and when it ends. I mean, for example to calculate service level, how many calls, etc. If I want to start the queue from with 0 calls, etc, how do I do this? And if I want to stop it, so I can start it again?? Thanks!! Regards, Sebastian Currently, there is not a way to do this with realtime queues. During a reload, realtime queues are not touched at all. I have a development branch set up which is supposed to help this as well as other rigidities present when it comes to reloading and resetting queues. The branch is located at the following URL if you wish to give it a test: http://svn.digium.com/svn/asterisk/team/mmichelson/queue-reset If you run the code there, you'll find that there is a command called queue reset stats which should do what you want. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
I'll assume that you suspect that asterisk is adding latency that you would like to tune. There is no simple variable that will affect latency as far as I know, but certainly one thing to look at is codec translation. Make sure your inbound and outbound paths are using the same codec, or latency will be added for sure. You can use tcpdump to measure the latency and the effect of anything you do to attempt tuning in a rough way - each packet has a timestamp at the beginning measured in ten thousandths (I think?) of a second. You should be able to see the RTP packet arrive and then leave again... just subtract the timestamps for your added latency. Cheers, j On Mon, 15 Dec 2008, michel freiha wrote: Dear Sir, What I'm interested to is to know how much time the rtp packets takes from the time it access the asterisk server,to when it'll leave Is this function or variable exist anywhere? Regards On Mon, Dec 15, 2008 at 10:55 PM, Jeff LaCoursiere j...@jeff.net wrote: No. TTL in the header is about hop traversal. Each IP router that forwards the packet will reduce this number in the live packet until it reaches zero, when it will be dropped. I believe this is to eliminate route loops creating packet storms. FWIW this is how traceroute works - it sends out packets with continually increasing TTLs and the router that drops the packet will send back a notification, so you can trace each hop... What is it you are trying to do or measure? j On Mon, 15 Dec 2008, michel freiha wrote: Dear Sir, There is no relation between TTL and the latency on asterisk server? Regards On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere j...@jeff.net wrote: TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: *Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] work in Chicago
Anyone know of any IT work in the Chicago area? I just moved up here and am finding the economy has really stifled things. Will do IT mgmt/Unix/Networking/VoIP/C for food... Cheers, j http://www.jeff.net/resume.pdf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
Dear Sir, I would like to ask please where I can find the results for RTCP debug on...It's saved on a file or it appears in the CDRs? Regards On Tue, Dec 16, 2008 at 12:06 AM, Mark Michelson mmichel...@digium.comwrote: michel freiha wrote: Dear Sir, What I'm interested to is to know how much time the rtp packets takes from the time it access the asterisk server,to when it'll leave Is this function or variable exist anywhere? If you want statistics on RTP packets, then you should look into RTCP reporting. A simple facility for looking at this information would be the Asterisk CLI commands rtcp stats on and rtcp debug assuming that you are running Asterisk 1.4. If you are using Asterisk trunk, the commands are rtcp set stats on and rtcp set debug on. You may also be able to filter the RTCP packets in a program like wireshark and analyze them there as well. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
You are right Jeff...Thanks a lot Regards On Tue, Dec 16, 2008 at 12:35 AM, Jeff LaCoursiere j...@jeff.net wrote: I'll assume that you suspect that asterisk is adding latency that you would like to tune. There is no simple variable that will affect latency as far as I know, but certainly one thing to look at is codec translation. Make sure your inbound and outbound paths are using the same codec, or latency will be added for sure. You can use tcpdump to measure the latency and the effect of anything you do to attempt tuning in a rough way - each packet has a timestamp at the beginning measured in ten thousandths (I think?) of a second. You should be able to see the RTP packet arrive and then leave again... just subtract the timestamps for your added latency. Cheers, j On Mon, 15 Dec 2008, michel freiha wrote: Dear Sir, What I'm interested to is to know how much time the rtp packets takes from the time it access the asterisk server,to when it'll leave Is this function or variable exist anywhere? Regards On Mon, Dec 15, 2008 at 10:55 PM, Jeff LaCoursiere j...@jeff.net wrote: No. TTL in the header is about hop traversal. Each IP router that forwards the packet will reduce this number in the live packet until it reaches zero, when it will be dropped. I believe this is to eliminate route loops creating packet storms. FWIW this is how traceroute works - it sends out packets with continually increasing TTLs and the router that drops the packet will send back a notification, so you can trace each hop... What is it you are trying to do or measure? j On Mon, 15 Dec 2008, michel freiha wrote: Dear Sir, There is no relation between TTL and the latency on asterisk server? Regards On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere j...@jeff.net wrote: TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: *Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Question
Is this going to be realeased in any 1.6 version son?? Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: lunes, 15 de diciembre de 2008 08:14 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Question Sebastian wrote: Hi, In queues realtime, when the queue start and when it ends. I mean, for example to calculate service level, how many calls, etc. If I want to start the queue from with 0 calls, etc, how do I do this? And if I want to stop it, so I can start it again?? Thanks!! Regards, Sebastian Currently, there is not a way to do this with realtime queues. During a reload, realtime queues are not touched at all. I have a development branch set up which is supposed to help this as well as other rigidities present when it comes to reloading and resetting queues. The branch is located at the following URL if you wish to give it a test: http://svn.digium.com/svn/asterisk/team/mmichelson/queue-reset If you run the code there, you'll find that there is a command called queue reset stats which should do what you want. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information from ESET Smart Security, version of virus signature database 3693 (20081215) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 3693 (20081215) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
[park-dial] ; app_park adds a priority 1 for us, but due to Asterisk oddities, we still need this Noop exten = _.,1,Noop exten = _.,n,Goto(corporate,3500,1) exten = h,1,Noop Mike wrote: That would help me, but I can't even do that (send all parked calls to anybody) because of the dynamic park-dial context. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric ManxPower Wieling Sent: Monday, December 15, 2008 16:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Follow up on parking Terry Wilson wrote: On Dec 15, 2008, at 7:05 AM, Mike wrote: Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Yes. I'm working on an issue right now related to parking and noticed that Asterisk completely lies with the verbose statement saying that it will time back out to an extension. There is an if/else that checks a string that will always be set and therefore will never hit the else...which is where the code is that would time back out to an extension as opposed to trying to magically find the original caller and call the channel back. It is fairly complex code in there, so it may take a bit to fix...but I thought I'd let your know that I am working on it, anyway. I saw this in 1.2 as well. I don't know about 1.4, since my customers never used 1.4. Since all parked calls were supposed to be sent to the operator, it was not an issue for my customers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tcpdum
Hi Michel, how's beirut's weather with ya! anyway, TTL stands for TIME To LIVE. it's encapsulated on layer three of the OSI layer to each packet going out that specific interface. by default routers has a 16 TTL that means each time the designated packet reaches a router (gets decapsulated) it gets a -1... this helps in preventing loops which would eventually lead to congestion. now latency wise, for VOIP to operate correctly it needs a latency of under 200 ms. (I currently have a microwave link , and unfortunately im not getting that a latency less than 280 to my SIP provider) if your asterisk server is hosted online, you could simply traceroute it and check the highest latency, point. and depending on where that bottle neck would be, youll troubleshoot from there.. mine were on my ISP's international link, after having a meeting with my account manager, I got my link routed through a different international path which drastically decreased my latency. now on a different approach, you absolutly have to talk to your ISP/network administrator to provide you QOS for that specific IP whether it's public or private. depending on your network's traffic QOS would surely help with no doubt.. this would decrease latency as well hope I've shed some light about this, if not well the more knowledge the betteR best, Roland From: michel freiha Sent: Monday, December 15, 2008 10:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users-boun...@lists.digium.com Subject: Re: [asterisk-users] tcpdum Dear Sir, There is no relation between TTL and the latency on asterisk server? Regards On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere j...@jeff.net wrote: TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: *Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean Regards* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Question
Is this going to be released in any 1.6 version soon?? Your branch (queue-reset) is supouse to be the same as trunk but with this functionality? Is this branch updated every time trunk is committed?? I checked the log and seems to have the latest commits of trunk, but I would like to be sure. Thanks Regards, Sebastian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Sent: lunes, 15 de diciembre de 2008 09:00 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queue Question Is this going to be realeased in any 1.6 version son?? Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: lunes, 15 de diciembre de 2008 08:14 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Question Sebastian wrote: Hi, In queues realtime, when the queue start and when it ends. I mean, for example to calculate service level, how many calls, etc. If I want to start the queue from with 0 calls, etc, how do I do this? And if I want to stop it, so I can start it again?? Thanks!! Regards, Sebastian Currently, there is not a way to do this with realtime queues. During a reload, realtime queues are not touched at all. I have a development branch set up which is supposed to help this as well as other rigidities present when it comes to reloading and resetting queues. The branch is located at the following URL if you wish to give it a test: http://svn.digium.com/svn/asterisk/team/mmichelson/queue-reset If you run the code there, you'll find that there is a command called queue reset stats which should do what you want. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information from ESET Smart Security, version of virus signature database 3693 (20081215) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 3693 (20081215) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information from ESET Smart Security, version of virus signature database 3693 (20081215) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 3693 (20081215) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
This may be an obvious thing, but you didn't mention checking whether or not the card was still seated in the slot properly after the move. I know from experience that when you move offices, even if you take all the precautions possible, a card can get bumped just enough to jostle the connections loose. Even if the card appears to be seated correctly I'd take it out and re-seat it. Unfortunately it looks like you may have compounded the problem by removing and reinstalling the zaptel packages. It looked like the card was still there - from memory the lspci command said it was. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
Paul Hales wrote: This may be an obvious thing, but you didn't mention checking whether or not the card was still seated in the slot properly after the move. I know from experience that when you move offices, even if you take all the precautions possible, a card can get bumped just enough to jostle the connections loose. Even if the card appears to be seated correctly I'd take it out and re-seat it. Unfortunately it looks like you may have compounded the problem by removing and reinstalling the zaptel packages. It looked like the card was still there - from memory the lspci command said it was. PaulH That is correct, lspci shows the card is there. I have also tried moving the card to a different slot to be sure. Langdon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow up on parking
I will definitely try this later todaythanks! Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric ManxPower Wieling Sent: Monday, December 15, 2008 18:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Follow up on parking [park-dial] ; app_park adds a priority 1 for us, but due to Asterisk oddities, we still need this Noop exten = _.,1,Noop exten = _.,n,Goto(corporate,3500,1) exten = h,1,Noop Mike wrote: That would help me, but I can't even do that (send all parked calls to anybody) because of the dynamic park-dial context. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric ManxPower Wieling Sent: Monday, December 15, 2008 16:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Follow up on parking Terry Wilson wrote: On Dec 15, 2008, at 7:05 AM, Mike wrote: Just so I'm clear: there is no way to do what I want short of playing with the underlying code, correct? Yes. I'm working on an issue right now related to parking and noticed that Asterisk completely lies with the verbose statement saying that it will time back out to an extension. There is an if/else that checks a string that will always be set and therefore will never hit the else...which is where the code is that would time back out to an extension as opposed to trying to magically find the original caller and call the channel back. It is fairly complex code in there, so it may take a bit to fix...but I thought I'd let your know that I am working on it, anyway. I saw this in 1.2 as well. I don't know about 1.4, since my customers never used 1.4. Since all parked calls were supposed to be sent to the operator, it was not an issue for my customers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
Langdon Stevenson wrote: Paul Hales wrote: It looked like the card was still there - from memory the lspci command said it was. PaulH That is correct, lspci shows the card is there. I have also tried moving the card to a different slot to be sure. Langdon So - the current state of play is: card = yes drivers = no As a stop gap, have you tried building the drivers from source? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
On Mon, Dec 15, 2008 at 05:57:08PM +1100, Langdon Stevenson wrote: Hi Paul Thanks for the reply. I have removed and re-installed all of the Fedora Zaptel packages with Yum. I have the following installed: asterisk-zaptel 1.4.12.1-1.fc8 zaptel.i386 1.4.12.1-1.fc8 zaptel-devel.i386 1.4.12.1-1.fc8 zaptel-lib.i386 1.4.12.1-1.fc8 zaptel-utils.i386 1.4.12.1-1.fc8 None of those packages contains kernel modules. A simple test: find /lib/modules/`uname -r` -name zaptel.ko Should not find anything. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Record CMD
I don't see a method to detect the success or failure for the Record CMD.
I'd like to know the reason why the recording ended
Am I wrong?
exten = recordmsg,1,Noop()
exten = recordmsg,n,Record(${NEWPHRASEID}:ulaw|4|180)
Bart___
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Re: [asterisk-users] Zaptel / TDM400P card stopped working
Paul Hales wrote: Langdon Stevenson wrote: Paul Hales wrote: It looked like the card was still there - from memory the lspci command said it was. PaulH That is correct, lspci shows the card is there. I have also tried moving the card to a different slot to be sure. Langdon So - the current state of play is: card = yes drivers = no As a stop gap, have you tried building the drivers from source? PaulH Yes, that is the current state of play and yes, it looks like I will have to build from source. I haven't done this before and am pretty busy at the moment, so it will take me a while. I will post back when I have done so. Thanks for the input (to all who have contributed), it is much appreciated. Regards, Langdon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
Langdon Stevenson wrote: Yes, that is the current state of play and yes, it looks like I will have to build from source. I haven't done this before and am pretty busy at the moment, so it will take me a while. I will post back when I have done so. Thanks for the input (to all who have contributed), it is much appreciated. Regards, Langdon Building the drivers from source will only take you 10 minutes - not a huge hassle. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
Hi Paul This is the thing that is confusing. When I set up Asterisk first time round I just used the Fedora packages, I didn't build from source. Is the asterisk-zaptel or the zaptel rpm supposed to provide the drivers? Does rpm -qf filename show the correct kernel version? My understanding is that yes, the Fedora packages provide everything. It may be relevant to mention that: uname -a returns: Linux switch 2.6.26.3-14.fc8 #1 SMP Wed Sep 3 03:40:05 EDT 2008 i686 i686 i386 GNU/Linux I then checked for the wctdm module: locate wtctdm returns: /lib/modules/2.6.23.15-80.fc7/misc/wctdm.ko /lib/modules/2.6.23.15-80.fc7/misc/wctdm24xxp /lib/modules/2.6.23.15-80.fc7/misc/wctdm24xxp/wctdm24xxp.ko /lib/modules/2.6.24.3-34.fc8/misc/wctdm.ko /lib/modules/2.6.24.3-34.fc8/misc/wctdm24xxp /lib/modules/2.6.24.3-34.fc8/misc/wctdm24xxp/wctdm24xxp.ko snip So it looks like the Zaptel packages (at least for wctdm) don't have a version to support the kernel that I have installed, which would explain things. Langdon Paul Hales wrote: h...I haven't used the RPM's before, so I can only guess that the RPM's are doing something not quite right. Is the asterisk-zaptel or the zaptel rpm supposed to provide the drivers? Does rpm -qf filename show the correct kernel version? If that fails, you could download the source files from the Asterisk site and build them yourself. PaulH Langdon Stevenson wrote: Hi Paul Thanks for the reply. I have removed and re-installed all of the Fedora Zaptel packages with Yum. I have the following installed: asterisk-zaptel 1.4.12.1-1.fc8 zaptel.i386 1.4.12.1-1.fc8 zaptel-devel.i386 1.4.12.1-1.fc8 zaptel-lib.i386 1.4.12.1-1.fc8 zaptel-utils.i386 1.4.12.1-1.fc8 The command: modprobe wctdm produces: FATAL: Module wctdm not found. The command: modprobe zaptel produces: FATAL: Module zaptel not found. Is there anything else that I should be doing? Regards, Langdon Paul Hales wrote: Have you tried loading the zaptel driver for your card manually? PaulH Langdon Stevenson wrote: Hi I have a Dell PE2300 with a Digium TDM400P line card in it (with one module to handle an inbound phone line). This is running on a Fedora 8 system with Asterisk 1.4.21.2-1.fc8 This system has been working nicely for about 12 months. After a recent move of office and relocation of the server Asterisk is back on line, but the TDM line card has stopped working. I have spent half a day working through Google search results, but no luck so far. The command: lspci -v produces: snip 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at e400 [size=256] Memory at f9ffd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel modules: hisax The IRQ is not in use by any other device, so there is no conflict (this seems to be a common problem). The card has always been detected as a Tiger3XX. What stands out here to me is: Kernal modules: hisax I don't believe that this was the case when I first installed the card (but it was over a year ago, so I may be wrong). The hisax driver is blacklisted in /etc/modprobe.d/blacklist. The command: lsmod produces: Module Size Used by xt_dscp 6465 0 rfcomm 32721 0 l2cap 21953 9 rfcomm bluetooth 47013 6 rfcomm,l2cap autofs420933 2 fuse 47837 1 tun12613 0 sunrpc154785 3 nf_conntrack_netbios_ns 6593 0 iptable_nat 8777 0 nf_nat 18393 1 iptable_nat iptable_mangle 6849 0 nf_conntrack_ipv4 11849 5 iptable_nat,nf_nat xt_state6209 2 nf_conntrack 51221 5 nf_conntrack_netbios_ns,iptable_nat,nf_nat,nf_conntrack_ipv4,xt_state ipt_REJECT 6977 2 ipt_LOG 9285 4 iptable_filter 6849 1 ip_tables 14033 3 iptable_nat,iptable_mangle,iptable_filter xt_tcpudp 6977 33 ip6t_REJECT 7617 2 ip6table_filter 6593 1 ip6_tables 15057 1 ip6table_filter x_tables 15557 9 xt_dscp,iptable_nat,xt_state,ipt_REJECT,ipt_LOG,ip_tables,xt_tcpudp,ip6t_REJECT,ip6_tables ipv6 238277 25 ip6t_REJECT dm_multipath 18505 0 parport_pc 26725 0 parport32173 1 parport_pc floppy 52229 0 i2c_piix4 11473 0 i2c_core 20949 1 i2c_piix4 pcspkr 6593 0 e100 33997 0 mii 8385 1 e100 dcdbas
Re: [asterisk-users] Country numbering plan resources
Laurent a écrit : Hello, HI I believe that one of the most comprehensive resources, in terms of numbering plans, is on the ITU website : http://www.itu.int/oth/T0202.aspx?parent=T0202 [...] As regards France [...] The document for France is out of date, eg +33 87x xxx xxx ar no more personal numbers and replaced by +33 9xx xxx xxx -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
Tilghman Lesher wrote: On Monday 15 December 2008 00:57:08 Langdon Stevenson wrote: Hi Paul Thanks for the reply. I have removed and re-installed all of the Fedora Zaptel packages with Yum. I have the following installed: asterisk-zaptel 1.4.12.1-1.fc8 zaptel.i386 1.4.12.1-1.fc8 zaptel-devel.i386 1.4.12.1-1.fc8 zaptel-lib.i386 1.4.12.1-1.fc8 zaptel-utils.i386 1.4.12.1-1.fc8 The command: modprobe wctdm produces: FATAL: Module wctdm not found. This probably means that the modules were compiled for a kernel other than the one you have installed. You probably have multiple directories within /lib/modules, and the zaptel modules are in a directory other than what is listed with 'uname -r'. In this case, compiling from source is probably your best bet. Hi Tilghman, I think that you are spot on. Sounds like I should try building from source. Regards, Langdon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6 upgrade issues
Greetings list,
Over the last few days I've been gearing up to replace a couple of our servers
with 1.6 as something of a testbed, but I'm encountering a few problems, and
wondering if anyone can help...
In extensions.conf, there are a number of contexts defined for each group of
users, along the lines of:
[groupa] [groupb] etc.
In each of those, there's a command include = outbound
[outbound] has entries similar to the following:
exten = _0[123],1,Macro(outbound,${EXTEN}, provider1, provider2)
the macro outbound is defined in extensions.ael as follows:
macro outbound (number, route1, route2) {
dosomestuff;
}
This has worked fine in 1.2 and 1.4, but seems to be choking on 1.6. I've
looked through the various changes.txt files, and have read mention of
replacing macro calls with Gosub(), but I'm not sure that's relevant to this
issue.
It looks like the dialplan parser is amalgamating the commands in macro
outbound with the context [outbound], which means of course the pattern match
in [outbound] can never execute the macro outbound.
Any thoughts?
Also seem to be getting some errors writing CDRs to a postgresql database. I'm
using the schema for pgsql from voip-info.org, which, again, has worked fine
logging 1.2 and 1.4. Have there been any schema changes in 1.6 one needs to be
aware of?
Thanks in advance!
Regards,
Chris
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Re: [asterisk-users] 1.6.1: iax trunk needs dahdi timing ??
Russell Bryant wrote: Michiel van Baak wrote: On 20:24, Sun 14 Dec 08, sean darcy wrote: starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? If you dont have any dahdi hardware installed and configured, make sure to load dahdi_dummy. That will provide you the timers. In 1.6.1, this should not be required. It's probalby a check in the code that shouldn't be there anymore. If you post this on bugs.digium.com, I'll remove it. Thanks for fixing this so promptly. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] devicestate / inuse issue with 1.4.21.1
Hi all, we do have a callcenter system running with 1.4.21.1 - the agents are connected used sip phones. SIP accounts are configured using realtime (sip buddies) - and are configured with call-limit=1. It is operating just fine - but from time to time it does happen that an agent with an active call (inbound or outbound) does start to get a second call offered. I have taken a look at the logging output and found the following [Dec 15 11:39:37] VERBOSE[10419] logger.c: -- Packet2Packet bridging SIP/tel01-b6b09b18 and SIP/spa941_0027-09047cf8 [Dec 15 11:40:45] DEBUG[31007] app_queue.c: Device 'SIP/spa941_0027' changed to state '3' (Busy) [Dec 15 11:41:40] DEBUG[10481] app_queue.c: SIP/spa941_0027 in use, can't receive call [Dec 15 11:42:43] DEBUG[31007] app_queue.c: Device 'SIP/spa941_0027' changed to state '3' (Busy) [Dec 15 11:45:18] DEBUG[31008] chan_sip.c: Destroying user object from memory: spa941_0027 [Dec 15 11:45:41] DEBUG[10619] app_queue.c: SIP/spa941_0027 in use, can't receive call [Dec 15 11:45:52] DEBUG[10626] app_queue.c: SIP/spa941_0027 in use, can't receive call [Dec 15 11:46:39] DEBUG[31008] chan_sip.c: Allocating new SIP dialog for 142376f5-f100a...@192.168.2.117 - REGISTER (No RTP) [Dec 15 11:46:39] DEBUG[31007] app_queue.c: Device 'SIP/spa941_0027' changed to state '1' (Not in use) As you can see - the agent with spa941_0027 does have an active call starting at 11:39:37 - it does get marked as busy (because of call limit) - thats correct. At 11:45:18 there was a sip reload - the user object gets destroyed - but the peer object not - so the busy level is still correct. Than at 11:46:39 the sip phone does reregister at the system - and the system does change the peer to be marked as not in use - from this point things are going wrong So i think the way to reproduce is - active call - sip reload, reregister, not in use state I have to verify this to be reproduceable - but wanted to ask here firstly if someone does already know this behaviour... I have seen bug http://bugs.digium.com/view.php?id=13525 - i think it is releated to it Here are the relevant sip settings Realtime SIP Settings: -- Realtime Peers: Yes Realtime Users: Yes Cache Friends: Yes Update: Yes Ignore Reg. Expire: No Save sys. name: Yes Auto Clear: 120 Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 360 secs regards, Wolfgang ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Netcomm V90s + Asterisk + conference
This might be a curly one- I have a Netcomm V90s VoIP phone that has 4 line function - L1 to L4. It appears the only way to use the conference function is to set up a 2nd (and any subsequent) VoIP account. Has anyone found away around this that does not involve setting up multiple SIP accounts? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
