Re: [Asterisk-Users] pickup problem
Hi! Fabio == Fabio [EMAIL PROTECTED] writes: Fabio are you using canreinvite=yes on your SIP endpoints definition Fabio ? No, I'm using canreinvite=no. Fabio also check your features.conf, do you have pickupexten = *8 ? Yes it is: canopus*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 ... Thanks! -- DSS5-RIPE DSS-RIPN 2:550/[EMAIL PROTECTED] 2:550/[EMAIL PROTECTED] xmpp:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://neva.vlink.ru/~dsh/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QuadBri card
It's mark on some documentations... Where do i laucnh qozap ?? Best regards, Olivier S. Tzafrir Cohen a écrit : On Wed, Jun 07, 2006 at 05:44:16PM +0200, Olivier Saulnier wrote: Hello, I install the latest release of Asterisk, QuadBri driver. I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for launch qozap... Bad place. rc.local is just about the last place in the init sequence to be run. After Asterisk is started. -- Olivier Saulnier STEGANUX 1er étage Diamecans Bel Air 03410 St Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: SV: [Asterisk-Users] I can hear only one way when I use nokiae-60withX-lite
Thats just the thing, and it sucks, because the VoIP implementation actually works very good. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use nokiae-60withX-lite Sounds like they crippled the phone for cellulars sake. On Jun 7, 2006, at 10:35 AM, Jon Schøpzinsky wrote: Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both from cellular network and IP network, and actuatly works quite well, both for cellular and IP traffic. But you cant do seamless handover, for example when you walk out of the building. You have two different numbers, your mobile number and your IP number And these cant automaticly be transferred. Hope this answeres your question Regards Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] På vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Schøpzinsky [EMAIL PROTECTED]: Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon What do you mean by users has to have some local equipment from the telco ? Do you think Nokia E60, E61 and E70 are appropriate for Fixed Mobile Convergence (each mobile phone being reachable at the same time from inhouse PBX and Telco's mobile network without any handover or roaming between both networks) ? Regards -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.2/357 - Release Date: 06-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QuadBri card
On Thu, Jun 08, 2006 at 08:56:52AM +0200, Olivier Saulnier wrote: It's mark on some documentations... Where do i laucnh qozap ?? qozap is not a program that you loanch. It is a kernel module that you load. Stick the command 'modprobe qozap' somewhere in your init scripts. Actually, there is already a zaptel init scripts. set MODULES=qozap in /etc/sysconfig/zaptel . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latest SVN with downloaded sounds.
I'm getting this error when compiling:- make[1]: Entering directory `/usr/src/asterisk.svn/sounds' --09:22:12-- http://ftp.digium.com/pub/telephony/sounds/releases/asterisk-core-sounds-en-wav-1.4.0.tar.gz = `asterisk-core-sounds-en-wav-1.4.0.tar.gz' Resolving ftp.digium.com... 216.27.40.102, 69.16.138.164 Connecting to ftp.digium.com|216.27.40.102|:80... connected. HTTP request sent, awaiting response... 404 Not Found 09:22:14 ERROR 404: Not Found. make[1]: *** [/var/lib/asterisk/sounds/.asterisk-core-sounds-en-wav-1.4.0] Error 1 make[1]: Leaving directory `/usr/src/asterisk.svn/sounds' make: *** [datafiles] Error 2 but the file is definitely there. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QuadBri card
Olivier Saulnier wrote: It's mark on some documentations... Where do i laucnh qozap ?? Best regards, Olivier S. Tzafrir Cohen a écrit : On Wed, Jun 07, 2006 at 05:44:16PM +0200, Olivier Saulnier wrote: Hello, I install the latest release of Asterisk, QuadBri driver. I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for launch qozap... Bad place. rc.local is just about the last place in the init sequence to be run. After Asterisk is started. what i have done at some clients sites, is actually putting an entry into the /etc/init.d/zaptel file. search for the modprobe command and put your qozap line in at the bottom. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Latest SVN with downloaded sounds. Update
On Thu, 2006-06-08 at 09:26 +0200, Dave Cotton wrote: I'm getting this error when compiling:- make[1]: Entering directory `/usr/src/asterisk.svn/sounds' --09:22:12-- http://ftp.digium.com/pub/telephony/sounds/releases/asterisk-core-sounds-en-wav-1.4.0.tar.gz = `asterisk-core-sounds-en-wav-1.4.0.tar.gz' Resolving ftp.digium.com... 216.27.40.102, 69.16.138.164 Connecting to ftp.digium.com|216.27.40.102|:80... connected. HTTP request sent, awaiting response... 404 Not Found 09:22:14 ERROR 404: Not Found. make[1]: *** [/var/lib/asterisk/sounds/.asterisk-core-sounds-en-wav-1.4.0] Error 1 make[1]: Leaving directory `/usr/src/asterisk.svn/sounds' make: *** [datafiles] Error 2 but the file is definitely there. Update after another look it isn't, there is only a gsm version. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Query
Hi, Can anybody tell me Does Asterisk has a TAPI Interface sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] query
Hi, Can anybody tell me that does asterisk have TAPI interface sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Set(CDR(userfield)) Trouble
check cdr_mysql.conf for userfield=1 turby @ www.canistec.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tristan Sent: Wednesday, June 07, 2006 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Set(CDR(userfield)) Trouble Hi, I have troubles setting the userfield in mysql ( using asterisk 1.2.8 / addons 1.2.3 ) I use this in my dialplan: exten = s,n,SetCDRUserField(SOMEVALUE) I tried also: exten = s,n,Set(CDR(userfield)=SOMEVALUE) But everytime i look at the cdr database the userfield is still empty Does anyone has a clue on how to get things working ? Thanks in advance ! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to delete a key from database in extensions.conf
I can set a family/key=value just fine, but how can i delete it?
exten = _200,1,AgentCallbackLogin(||[EMAIL PROTECTED])
exten =
_200,2,Set(DB(AgentsMAP/${CALLERIDNUM})=${AGENTBYCALLERID_${CALLERIDNUM}})
exten = _200,3,Hangup
exten = _201,1,AgentCallbackLogin(||)
exten = _201,2,Set(DB(AgentsMAP/${CALLERIDNUM})=)
exten = _201,3,Hangup
--
~Shaun
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI (I.430) over ethernet
2006/6/8, Paul Hales [EMAIL PROTECTED]: Another option would be to see if the provider will provide 2 BRI linesthat are tied together in some way.Most of the providers in Australia will do similar things with PRI.PaulH Do you mean receiving traffic on 2 BRI lines (2 channels spread on 2 separate ports) connected to 2 differents boxes so that one line or box failure wouldn't affect incoming calls ?If positive, do these providers price this service (2 ports - 2 channels) at an intermediate level between simple capacity (1 port -2 channels) and double capacity (2 ports - 4 channels) ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: GXP-2000 (steer clear)
good to known. I played with the idea to buy one of these. Unacceptably bad voice quality. Point. You would suggest GrandStream then? Surely better in my experience. DV ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Quality of Asterisk
The file are there: http://thdei.info/results.zip and http://thdei.info/mos_6_MOS-USA_Test-114_20060605-042551cut-PESQ.png
because, last time I put them in attachment and the
mail was waiting for approvement and I never see it anmore
.
From: Deillon Thomas-WTD008 Sent: 05
June 2006 14:32To:
'asterisk-users@lists.digium.com'Subject: Quality of
Asterisk
Hi,
I have a problem
with the quality test. So if you have a idea for me
We test here,
Motorola phone with Asterisk. Asterisk play sample to the mobile phone which
recordthis and the inverse.
We have to be sure
that Asterisknot make distortion itself.
To do this, I tried
to play 7 longs files (20 minutes) in parallel (It go out from on zap line and
come back on a other line) like this:
#i=0
#while i
7:
#os.system("make the
call number 7%s"%i)
#
time.sleep(80)//80 sec = 1 column on graphs
#i+=1
And, what I see isthat when I launch
thesample 77, a delay appear on sample 71,73,74,75,76. Around 40
ms.
So, next, I try to
make 6 calls which play but no record and only one that record.The result was
just one little gap of 1 ms on one try and nogap on others.
Then, I launch 1
call and I make "hdparm -tT /dev/sda find / /tmp/tmp" and
make a graph of the result (file Test-114)
The HD is
a WDC WD400BD-75JM
hdparm result:
--8-
systemtest:/proc/scsi# hdparm -tT /dev/sda
/dev/sda:Timing cached reads: 4260 MB in 2.00
seconds = 2129.87 MB/secHDIO_DRIVE_CMD(null) (wait for flush complete)
failed: Inappropriate ioctl for deviceTiming buffered disk
reads: 170 MB in 3.02 seconds = 56.29
MB/secHDIO_DRIVE_CMD(null) (wait for flush complete) failed: Inappropriate
ioctl for
device--8--
server: 1Go
Ram, Intel(R) Pentium(R) 4 CPU 3.20GHz
cat
/proc/interrupts:
systemtest:/proc/scsi# cat /proc/interrupts
---8-
CPU0 0: 68356288 IO-APIC-edge
timer 1:
8 IO-APIC-edge i8042
9: 0
IO-APIC-level acpi12:
101 IO-APIC-edge
i804214: 486356
IO-APIC-edge
libata15:
4 IO-APIC-edge libata169:
845934 IO-APIC-level eth0177: 273396437
IO-APIC-level
wct4xxpNMI:
0LOC:
68356565ERR:
0MIS:
08--
I though it was the Hard-disk and my boss had the idea
to make a ramdisk and store the files ona
ramdisk.
So, then, the results were perfect but ifwe make
a "hdparm -T" on the disk whilewe make records, there are a lot of
gaps on files.
It's wherewe are. It's surely a IRQ problem :
http://www.asteriskguru.com/tutorials/hdlc_bad_fcs.html
Do youthink I am in the wrong way or do you know
ainterresting website orsomething like that that can help me
?
Thanks for your
help,
Thomas
DEILLON
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Simple Speeddial AGI
Hi, I am looking for a simple php agi script that locates a speeddial number in a MySQL database and then dials that number. ie. exten = 01,1,Noop(speeddial 01) exten = 01,2,Agi(do database lookup on 01 and finds $NUMBERTODIAL) exten = 01,3,Goto($NUMBERTODIAL,1) Anybody know if something like this exists or should I start from scratch. tx M -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astricon No More...
Friends in the community, I've received many mails saying I'll meet you at Astricon Europe. The sad answer is no, you will not. I have nothing to do with Astricon any more. After some arguments, Steve decided that Astricon, trainings, the business we had built together - everything belonged to him and he threw me out. I can't accept this behaviour and haven't gotten an offer for compensation that I feel I can accept. So please don't expect to meet me at any Astricon in the future until this issue is resolved. You can however, expect to meet me at other events, like the upcoming VON Fall in Boston, my trainings and other events that I will work on. The only reason I haven't gone public with this information before, is Mark. He asked me if he could help resolve this since he felt that Astricon belonged to the community and all of us should be able to work together. Well, it did not work out and Astricon now is in the hands of Steve Sokol. If not before, I'll see you at Von Boston! And by all means, continue testing Asterisk svn trunk - we need your feedback. Regards, /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] query
[EMAIL PROTECTED] wrote: Hi, Can anybody tell me that does asterisk have TAPI interface sanchal No, if you're a windows user, there is asttapi which uses the management interface though. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to check NAT behaviour before installing Asterisk
Hi,Installing Asterisk involve tuning NAT and network settings.So, before installing an Asterisk server, I would like to check my network settings.My setup is : IP Telephony Provider - ISP -- Home router-firewall -- Home LAN --- IPBX and IP Phones What and how would you check your settings ?More precisely, I would like to check outgoing and incoming calls capabilities.What is the safest test to run to be sure that an outgoing RTP flow is not blocked by my firewall (no logs in it) ? Is there some ping-echo-like command I could type-in from a home PC to check, independantly from any Asterisk setting, that a RTP packet would successfully come in ?Ping would be perfect (you can choose IP address and ports and you can access web sites from which you can issue a ping as if you were outside your home network) but, as far as I know, it relies on ICMP which is possibly treated specifically by firewalls.What do you think ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware to connect analog and ISDN fax devices
Hi all, I've read a lot of problems with faxing over asterisk. Most of them referred to Fax over Internet, but I want to connect analog and ISDN fax devices to asterisk to send and receive faxes over PRI: +-+ +--+++ | | | || ISDN Fax | | PRI |-| Asterisk |++ | | | || Analog Fax | +-+ +--+++ Can this be done without problems and can I use Digium cards to accomplish this? Or do I need other hardware devices? Regards, Jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHP UnixODBC MS SQl 2000
I used the configure option '--with-mssql' after freeTDS is installed. http://uk.php.net/manual/en/ref.mssql.php Fatal error: Call to undefined function: odbc_connect() in /var/www/html/odbctest.php on line 3 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to delete a key from database in extensions.conf
Shaun wrote: I can set a family/key=value just fine, but how can i delete it? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DBdel -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN master clock issue ?
Hi, Just a little question about digium/sangoma difference of behaviour... I need to setup 3 E1 connections to 3 different ISDN clock provider ... Can the TE411P handle this per span or do I have to buy a Sangoma one ? Thanks in advance ! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to delete a key from database in extensions.conf
show application DBdel on the CLI. OK this is deprecated but it still works. Maybe asterisk gives you hints what do use now. Doug Lytle schrieb: Shaun wrote: I can set a family/key=value just fine, but how can i delete it? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DBdel -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Christophorus Laube n:Laube;Christophorus org:SemanticEdge GmbH adr:;;Kaiserin-Augusta-Allee 10-11;Berlin;;10553;Deutschland email;internet:[EMAIL PROTECTED] title:Systemadministrator tel;work:+49-30-34507758 url:http://www.semanticedge.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nokia N80 and asterisk?
Recent posts indicate people have been having luck with the nokia E60/E7x phones and asterisk. I was wondering though if anyone had had any luck with the N80? I've got the N80 to register with asterisk, and that works just fine. However, it gives a 486 when I try to place SIP calls to it (either to the register username, or to the phone number). Oh, and I can't figure out how to make sip calls either. Has anyone got any further with the N80 and asterisk? Cheers Nick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using regcontext
Hello List Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip peers to have the regexten _[0-9]., so that I can capture all registrations in a single extension. But when they register, I can see that the dynamic extension is created, but none of the rest of the code is executed, priority 2-4. Can anyone explain how I should use the regcontext parameter, etc. am I using it wrongly? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/358 - Release Date: 07-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using regcontext
8 jun 2006 kl. 11.57 skrev Jon Schøpzinsky: Hello List Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip peers to have the regexten _[0-9]., so that I can capture all registrations in a single extension. But when they register, I can see that the dynamic extension is created, but none of the rest of the code is executed, priority 2-4. Can anyone explain how I should use the regcontext parameter, etc. am I using it wrongly? You can't set aq regexten= setting to a wildcard. Regexten does not capture registrations, it adds an execution step to an exact extension. Regards, /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Using regcontext
Hello Thanks for the answer... Just realized it myself, as your mail arrived :) Could be a nice feature though. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Olle E Johansson Sendt: 8. juni 2006 12:09 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Using regcontext 8 jun 2006 kl. 11.57 skrev Jon Schøpzinsky: Hello List Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip peers to have the regexten _[0-9]., so that I can capture all registrations in a single extension. But when they register, I can see that the dynamic extension is created, but none of the rest of the code is executed, priority 2-4. Can anyone explain how I should use the regcontext parameter, etc. am I using it wrongly? You can't set aq regexten= setting to a wildcard. Regexten does not capture registrations, it adds an execution step to an exact extension. Regards, /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/358 - Release Date: 07-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/358 - Release Date: 07-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What does RELAXDTMF do?
I have an issue with DTMF. DTMF is being partly recognised by some external IVR systems (banks, billing, etc), other IVR systems have intermittent issues. Call our VSP directly and using their IVR system without issue, and our internal IVR works just fine. Currently i have all voip devices using RFC2833, which is what is recommended, and thus the sip.conf file has dtmfmode=rfc2833 and relaxdtmf=yes. I have not seen any information that clearly defines the purpose of the relaxdtmf parameter in the sip.conf file, and wondering of flicking it from yes to no will have an impact, and if so what sort of impact will it have? Redhat FC4 + updates Asterisk v1.2.9.1 SNOM v6.0.3 beta SPA3000 v3.1.10d ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions problem
Dear If I have an extention 111 and 112 on my system but when the user 111 call the 112 call it through trunk not through local to perform a billing How can I solve it Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zap calls drop suddenly + tremendous noise when answering a call
We have an asterisk box with the following configuration: - AMD Athlon XP 2400+ - 512 MB RAM - SUSE Linux 10.1 - a Digium card TDM400P with 3 FXO - another Digium card TDM400P with 4 FXS - asterisk 1.2.7.1 - zaptel 1.2.4 I already checked that those cards aren't sharing interrupts (by cat /proc/interrupts): 0: 14119786 XT-PIC timer 1: 10 XT-PIC i8042 2: 0 XT-PIC cascade 8: 2 XT-PIC rtc 9: 0 XT-PIC acpi 10: 56469896 XT-PIC wctdm 11: 17172 XT-PIC eth0 12: 56474221 XT-PIC wctdm 14: 74633 XT-PIC ide0 15: 499385 XT-PIC ide1 This box is connected to 3 analog lines (PSTN), one of these lines is our ADSL line and 4 analog phones in our office (as you can see, we are a small business). Once or twice a day (yesterday it happens four times in a period of two hours) while we are in a call, it drops suddenly. No matter who is calling, I mean, it happens when we call from inside the office and when somebody calls us. Down here is the output from one of these phone calls (I replaced the phone number with some Xs). Somebody calls us from the line that's connected to zap channel 2. Then, our 4 phones ring and I pick up the call on phone connected to zap channel 7. Ten minutes later that event fires up: Jun 7 17:53:09 DEBUG[9015] chan_zap.c: Got event On hook(1) on channel 2 (index 0). After the output I attached my zapata.conf. Anyone has had this problem before? Is there something in my zapata.conf that's not correct? Any help would be very appreciated. I don't know if it's related, but, one or twice a day, when our phones ring and we answer, there's a tremendous noise and we can't do anything (for example, trying to park or transfer the call doesn't work). Then we hang up, our client calls again, we pick up the phone and the call goes well. I've searched on the web and found some messages talking about shared interrupts but this is not the case. thanks, enrico. Jun 7 17:43:33 VERBOSE[9015] logger.c: -- Starting simple switch on 'Zap/2-1' Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Executing System(Zap/2-1, /usr/local/bin/sendcallerid X ) in new stack Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Executing Dial(Zap/2-1, Zap/5Zap/6Zap/7Zap/8|30|rt) in new stack Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Called 5 Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Called 6 Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Called 7 Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Called 8 Jun 7 17:43:34 DEBUG[9015] chan_zap.c: Requested indication 3 on channel Zap/2-1 Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Zap/5-1 is ringing Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Zap/6-1 is ringing Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Zap/7-1 is ringing Jun 7 17:43:34 VERBOSE[9015] logger.c: -- Zap/8-1 is ringing Jun 7 17:43:35 DEBUG[9015] chan_zap.c: Exception on 17, channel 5 Jun 7 17:43:35 DEBUG[9015] chan_zap.c: Got event Ringer Off(11) on channel 5 (index 0) Jun 7 17:43:35 VERBOSE[9015] logger.c: -- Zap/5-1 is ringing Jun 7 17:43:35 DEBUG[9015] chan_zap.c: Exception on 18, channel 6 Jun 7 17:43:35 DEBUG[9015] chan_zap.c: Got event Ringer Off(11) on channel 6 (index 0) Jun 7 17:43:35 VERBOSE[9015] logger.c: -- Zap/6-1 is ringing Jun 7 17:43:35 DEBUG[9015] chan_zap.c: Exception on 19, channel 7 Jun 7 17:43:35 DEBUG[9015] chan_zap.c: Got event Ringer Off(11) on channel 7 (index 0) Jun 7 17:43:35 VERBOSE[9015] logger.c: -- Zap/7-1 is ringing Jun 7 17:43:35 DEBUG[9015] chan_zap.c: Exception on 20, channel 8 Jun 7 17:43:35 DEBUG[9015] chan_zap.c: Got event Ringer Off(11) on channel 8 (index 0) Jun 7 17:43:35 VERBOSE[9015] logger.c: -- Zap/8-1 is ringing Jun 7 17:43:36 DEBUG[9015] chan_zap.c: Exception on 15, channel 2 Jun 7 17:43:36 DEBUG[9015] chan_zap.c: Got event Ring Begin(18) on channel 2 (index 0) Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Exception on 15, channel 2 Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Got event Ring/Answered(2) on channel 2 (index 0) Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Setting IDLE polarity due to ring. Old polarity was 0 Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Exception on 17, channel 5 Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Got event Ringer On(10) on channel 5 (index 0) Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Exception on 18, channel 6 Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Got event Ringer On(10) on channel 6 (index 0) Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Exception on 19, channel 7 Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Got event Ringer On(10) on channel 7 (index 0) Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Exception on 20, channel 8 Jun 7 17:43:38 DEBUG[9015] chan_zap.c: Got event Ringer On(10) on channel 8 (index 0) Jun 7 17:43:40 DEBUG[9015] chan_zap.c: Exception on 17, channel 5 Jun 7 17:43:40 DEBUG[9015]
[Asterisk-Users] I can hear them but they can't hear me with VoipBuster
Hi; When connecting via VoipBuster or VoipStunt, I can hear them but they can't hear me . This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. I tried with different codecs: gsm, alaw and ulaw but no change. So, now I suppose VoipBuster must be blocking something. My tests were made with calls to several landlines in Portugal, which are currently free calls. I am using full cone nat at my PIX. Can anyone give me an explanation of what may justify it or a possible solution my sip.conf [vpb] type=peer secret= username=x fromuser=x host=sip1.voipbuster.com fromdomain= sip1.voipbuster.com insecure=very canredirect=no disallow=all allow=gsm allow=ulaw nat=yes qualify=no context=internal externip=my.public.ip.address localnet=my.local.ip.address /my.local.subnet.mask Many thanks! Rafael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to delete a key from
Christophorus Laube wrote: show application DBdel on the CLI. OK this is deprecated but it still works. Maybe asterisk gives you hints what do use now. As far as I know, dbdel is not depreciated. There is no function for dbdel yet, at least not that I've read about. dbput and dbget are depreciated. I'm sure I'll be corrected, if this isn't the case. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe - Annouce user join/leave without recording the name
Hi all, I an using MeetMe and I would like to use the -i function to annouce the join/leave of the user. However, this require that users record their names. Is there anyway to remove this? I just want MeetMe to annouce somethig like A new user has joined the conference and that need not to record user's name. Is there a way to do this?? Pim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Plainvoip problem.
When calling through Plainvoip from my Asterisk at Home box I get the following log entries. Jun 8 01:27:25 VERBOSE[5550] logger.c: -- Called plainvoip/1insert # Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Call accepted by 66.199.240.2 (format g729) Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729 Jun 8 01:27:26 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 is ringing Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 stopped sounds Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 answered SIP/503-6d4c Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256 What I hear on the phone is one ring and then nothing. This has only been in the past few days. Has anybody else had a problem like this? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe - Annouce user join/leave without recording the name
In article [EMAIL PROTECTED], Pimjai Wesnarat [EMAIL PROTECTED] wrote: Hi all, I an using MeetMe and I would like to use the -i function to annouce the join/leave of the user. However, this require that users record their names. Is there anyway to remove this? I just want MeetMe to annouce somethig like A new user has joined the conference and that need not to record user's name. Is there a way to do this?? Yes, but you will to modify the source code of app_meetme.c and recompile. Look for the places where the variable namerecloc is used. Remove the code that records, stores or uses the name file, but retain the code that plays the has joined or has left files. Change those file names to something of your choosing, and record the messages you desire into those files. Hope this helps Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay on calls
On 6/7/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: I have a 1.2.4 * box with two HFC modems using chan_modem_i4l and several SIP phones and ATA's. We have a terrible delay on calls between the PSTN (isdn BRI) and the SIP phones. All internal calls are fine. My first thought was that the transcoding could cause the delay but all of the SIP phones default to ulaw so there should not be any transcoding needed. I also checked the load on the server and it is well below 10% cpu utilisation and load average of below 1. Is there some form of jitter buffering going on? Perhaps the echo cancelling (I assume there is some) is adding a significant delay? Perhaps try turning it off temporarily. It seems likely to be related to chan_modem_i4l if your internal calls are all okay. Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/2.0 484 Address Incomplete
Hi all I have downloaded from openser and iam trying to integrate voice mail with asterisk I have read all the docs in the document site after config, and people recomendation iam able to run the openser successfully and able to fix the problem calling out side but when the local user not available, iam sending to asterisk voice mail and i get error SIP/2.0 484 Address Incomplete in my x-lite client any suggestions ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] increase the volume ?
Hi, Is it possible de tell asterisk to increase the volume? When we place or recieve a call the volume is very low, using a smartphone or a hardphone. Thanks for advance ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set quietmp3 but what about when using native? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
Turby, Thanks for your replay, but does this mean that * can't play mp3s? I was hoping not to have convert the MOH mp3s that came with * and are supposed to be the native default that * is supposed to be able to play -- not to mention that I don't have sox installed with whick to convert them. Does this mean that 1.2.7.1 has a bug? If so can someone tell me if I should, and how I would go about reporting it. Thanks again for the reply.turby [EMAIL PROTECTED] wrote: convert the moh sounfile to pcm or sln save the file to /var/lib/asterisk/moh/default set the musiconhold.conf [default]mode=filesdirectory=/var/lib/asterisk/moh/default turby@ www.canistec.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard ReinaSent: Wednesday, June 07, 2006 5:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install Thank you very much for your relply. No I did not install mpg123 as the instructions at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conffor version 1.2 say the mpg123 is no longer needed.| Rurouni Alucard | [EMAIL PROTECTED] wrote: Did you check your mpg123 version ?, asteriskneeds a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install I have followed the instructions provided at:http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincluding installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.However, no sound is heard and I get this message from the CLI when accessing MOH:-- Started music on hold, class 'default', on channel 'Zap/19-1'-- Stoped music on hold on Zap/19-1This happens whether it's a parked call or whether I access MOH directly via:exten = 800,1,Answerexten = 800,2,MusicOnHold()Any help would be greatly appreciated.Thank you very much.Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidthand Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update optionsvisit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: asterisk and nortel meredian option 11c
Hi I want to connect asterisk 1.0.9 ( kernel 2.6.8-2 debian )with TE110P and Nortel meridian option 11c release 25.40 with NTBK50AA card which is An E1 card. But the main problem is the first stage that no sync occurs the * card never syncs with meridian card I am using euroisdn, ccs , hdb3, crc4 , pri_net on asterisk And I am assuming that meridian is using same as it is connected to Nortel passport mvpe card which is an e1 isdn card and using the same config as astresk but the card never see each other. On the contrary when I connect the asterisk with the Nortel passport mvpe card it does detect the mvpe card but the d chan flaps btwn up and down and the hell of HDLC BAD FCS messages appears on the cli of * . I have also tried yellow alarm on the span but not of any help . Can any one tell me the config of meridian option 11c and asterisk and what I am doing wrong. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: meetme public
On Thu, Jun 08, 2006 at 12:03:32AM +0100, Marco Mouta wrote:
Marco. i solve this creating adding the meetme extension in the default
context. this extension now is valid for any user.
Hi,
Please check you [general] section in sip.conf
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying The number you have dialed is not in service. Please check
the
; number and try again.
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
It could be happening that your public sip call is arriving @
asterisk, and seems unknow, so it is sent to from-sip-external
context.
In your extensions.conf look for section called [from-sip-external],
there you need to paste your code to route the call to your meetme
room.
Hope it helps,
Best regards,
Marco Mouta
Ps. Please give me some feeback if it solved.
On 6/7/06, Pablo Allietti [EMAIL PROTECTED] wrote:
hi all i have an asterisk working and i need to add a mettme public
service.
for example i need to download a soft (sjphone) and without any
configuration call to [EMAIL PROTECTED] (meetme) and
join a conference but when i do that i
received an error saying nomber do not exist. but if i call a
extension
is work propperly.
in the extensions.conf have
exten = 411,1,Answer
exten = 411,2,Wait(1)
exten =
411,3,SetVar(CALLFILENAME=/var/spool/asterisk/monitor/${TIMESTAMP})
exten = 411,4,Monitor(wav,${TIMESTAMP},m)
exten = 411,5,Meetme(4001,qM)
exten = 411,6,Hangup
4001 is the room number
in the mmetme conf have
conf = 4001
any comments?
--
___
--Bandwidth and Colocation provided by [3]Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
[4]http://lists.digium.com/mailman/listinfo/asterisk-users
--
Com os melhores cumprimentos,
Marco Mouta
References
1. mailto:[EMAIL PROTECTED]
2. mailto:[EMAIL PROTECTED]
3. http://Easynews.com/
4. http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
--
.-
Pablo Allietti
E-mail: [EMAIL PROTECTED] | LACNIC
Phone : +598 2 604 | http://LACNIC.NET
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
If you have to use it, make sure you only use the mpg123 bundled with the asterisk distribution. mpg123 from any other source (yes, evem the developer's website) will yield major issues. On 8-Jun-06, at 8:12 AM, Matt wrote: I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set quietmp3 but what about when using native? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN master clock issue ?
Tristan wrote: Hi, Just a little question about digium/sangoma difference of behaviour... I need to setup 3 E1 connections to 3 different ISDN clock provider ... Can the TE411P handle this per span or do I have to buy a Sangoma one ? The digium card has a single on-board clock and you choose which of the three PRI's you want to sync to. The Sangoma card has an on-board clock for each PRI, therefore you can sync each clock to its respective PRI. However, there is seldom a need to truly sync to all three at the same time. In very general terms, all PRI providers sync their equipment to a higher level clock (their upstream provider), negating the need for you to sync to each. Is there some specific implementation that you're thinking of that requires each E1 port to sync to their respective provider? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hangup don't realease analog line
hi all (again). i have this problem. when a people call to meetme and join a conference when this people leave and hangup your phone asterisk can't detect the hangup. all people use analog lines to connect the meetme is any way to tell asterisk to hook when these people leave? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN master clock issue ?
The fact is that I have 2 different E1 (euroisdn) providers and an E1 (euroisdn) connection to a Matra PBX... The PBX needs to be master and as far as I know the PSTN providers needs it too... So I want to be sure that the quad E1 card I'll buy will work without troubles in this kind of setup... Can I only put the synchro to the PBX and forget about the PRI providers ? Rich Adamson a écrit : Tristan wrote: Hi, Just a little question about digium/sangoma difference of behaviour... I need to setup 3 E1 connections to 3 different ISDN clock provider ... Can the TE411P handle this per span or do I have to buy a Sangoma one ? The digium card has a single on-board clock and you choose which of the three PRI's you want to sync to. The Sangoma card has an on-board clock for each PRI, therefore you can sync each clock to its respective PRI. However, there is seldom a need to truly sync to all three at the same time. In very general terms, all PRI providers sync their equipment to a higher level clock (their upstream provider), negating the need for you to sync to each. Is there some specific implementation that you're thinking of that requires each E1 port to sync to their respective provider? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
I'm not using mpg123... I'm using NATIVE MOH! On 6/8/06, Jason Lixfeld [EMAIL PROTECTED] wrote: If you have to use it, make sure you only use the mpg123 bundled with the asterisk distribution. mpg123 from any other source (yes, evem the developer's website) will yield major issues. On 8-Jun-06, at 8:12 AM, Matt wrote: I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set quietmp3 but what about when using native? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gsm file
Hi, I'm newby here, reading the handbook and starting playing with *. What are the audio .gsm files in /var/lib/asterisk/sounds ? Playback command can only play .gsm ? how do i convert from .wav to .gsm ? Thanks a lot Victor ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN master clock issue ?
Tristan wrote: The fact is that I have 2 different E1 (euroisdn) providers and an E1 (euroisdn) connection to a Matra PBX... The PBX needs to be master and as far as I know the PSTN providers needs it too... So I want to be sure that the quad E1 card I'll buy will work without troubles in this kind of setup... Can I only put the synchro to the PBX and forget about the PRI providers ? No. It is highly unlikely the PBX truly needs to be master. If that were an actual requirement, the vendor would never be able to sell their PBX into any environment where they connected to a E1 provider. The telco providers never slave their equipment from a customer-owned PBX. You will need to find the option in the Matra PBX to define it as syncing from the E1. (In fact, I'd bet a small amount of money the default implementation in the PBX is to sync from any attached E1.) There is a 99.99% probability the two E1 providers obtain their clock sync from a higher level (hierarchical source), and are already in sync with each other. If you use a digium card, you select one of the providers as your first choice sync source, and the second provider as your second choice sync source when the first choice provider's E1 is down. You definitely want your digium/sangoma card to support the hierarchical design of the digital network, and that well known design requires you to sync from your upstream provider, and pass that sync along to your downstream PBX. If you don't do that, calls originating from the PBX and passing through the digium/sangoma card to the PRI network will incur clock slips (out of sync). If the clock slips are too great, you will experience clicks, etc, during a call. Also, if all the components are not in sync, any use of modems (eg, faxes or pc modems) will be significantly degraded if not impossible to use. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: help required plzzzzzzzzzz
Sir Thanks so much but I have done lots and lots of googling around and I also had a grip on this file earlier. I have already tried this but this is for the T1 scenerio. I am looking for the ISDN PRI over E1 and it is not doing any good to me. The exact card on the Nortel 11c is NTBK50AA. Which is an E1/PRI card. It seems to me that u have taken help from that file I have seen the mailing list archives and seems that people are using 5ess instead of euroisdn. In my case the physical interface does not go up and te110p indicates red led all the time some times goes yellow. I have confirmed the cable it work on other links. Is underneath the only change u made to the described config of meridian and what was the value u put in dch under in ld 96. ENL SERV dch Best Regards Mohammad Zeeshan Latif Sr. WAN Engineer NETWORK DIRECTORATE 0092-51-90391020,0092-321-5181157 From: Greg Camp [mailto:[EMAIL PROTECTED] Sent: Thursday, June 08, 2006 5:55 PM To: Muhammad Zeeshan Latif Subject: RE: help required plzz Good luck. Thanks, Greg From: Muhammad Zeeshan Latif [mailto:[EMAIL PROTECTED] Sent: Thursday, June 08, 2006 2:41 AM To: [EMAIL PROTECTED] Subject: help required plzz Hi sir I need ur help as I read ur post to google group asterisk-users. Which is as under Update: SUCCESS!!! There were two subtle items that allowed our Opt81C to talk PRI to * using a TE110P: 1) On the 81C in LD 96 we had to ENL SERV dch for the d-channel. 2) It appears that the TE110P needs a decent refresh time for the b-channels to come up cleanly. For example, on the Nortel if you disable the T1 or d-channel and re-enable it quickly (specifically, anything shorter than about 45 seconds) the TE110P won't come up clean. The Nortel will show all the b-channels as MBSY or FE MBSY. However, if you wait 45s - 1min then the d-channel and b-channels will come up clean every time. Many thanks to all who offered suggestions and worked with us on this! I am trying to connect * 1.0.9 with a TE110P card to Nortel option 11c 25.40 release and having very serious issues with both. First and last of all I was never able to bring the channels up on asterisk nor meridian option 11c. Can u plzz mail me the configuration of meridian and asterisk I will be very greatful for that. I request u once again plzz send me the config I will be very greatful. Best Regards Mohammad Zeeshan Latif Sr. WAN Engineer NETWORK DIRECTORATE 0092-51-90391020,0092-321-5181157 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gsm file
Hi Victor, 1) you can find sounds.txt file inside asterisk tar file containing the text of all asterisk sounds and relative filenames 2) Asterisk can play other formats (for example some wav format): search on wiki 3) for sound conversion see wiki: http://www.voip-info.org/wiki-Asterisk%20sound%20files (try googling around on wiki) Giorgio Victor Moreno wrote: Hi, I'm newby here, reading the handbook and starting playing with *. What are the audio .gsm files in /var/lib/asterisk/sounds ? Playback command can only play .gsm ? how do i convert from .wav to .gsm ? Thanks a lot Victor ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gsm file
Victor Moreno wrote: Hi, I'm newby here, reading the handbook and starting playing with *. What are the audio .gsm files in /var/lib/asterisk/sounds ? Playback command can only play .gsm ? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playback how do i convert from .wav to .gsm ? http://www.voip-info.org/wiki/view/Asterisk+sound+files Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI on the PA168V in IAX mode?
Oh well. It would have been a nice feature, but with Asterisk's voicemail-to-email it's not really a necessity. Thanks for the information! On 6/7/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: http://www.aredfox.com/eqa.htm#line_10 Check this Dan On 08/06/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Thomas Kenyon wrote: Lachek Butalek wrote: I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Afaik, none of the aredfox ATA firmware images support MWI, one reason I've never bought one. Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) Uses a PA168S. by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. I think just stating mailbox=number will work too. 1. There is a PA1688 mailing list on Yahoo: http://groups.yahoo.com/group/pa1688/ 2. What firmware version do you have? Latest is 1.51 3. The PA1688 chip is being replaced by the AR1688, so new products will use those instead. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: asterisk and nortel meredian option 11c
Muhammad, I have been struggling with M1 and * over an E1 for a while myself, but know it's running fine. Here's my d-channel config: ADAN DCH 18 CTYP MSDL CARD 08 PORT 1 DES Asterisk1 USR PRI DCHL 8 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC EURO CNTY BEL PINX_CUST 0 ISDN_MCNT 300 CLID OPT1 PROG NCHG CO_TYPE STD SIDE USR CNEG 1 RLS ID ** RCAP COLP MBGA NO OVLR NO OVLS NO T310 120 INC_T306 0 OUT_T306 0 T200 3 T203 10 N200 3 N201 260 K 7 It's a config where Asterisk is master and Meridian is slave in euroisdn. The zapata.conf that goes with that: #---[trunkgroups][channels]context=incoming-prabusydetect=nousecallerid=yescidsignalling=v23usecallingpres=yescallerid=asreceivedswitchtype=euroisdn signalling=pri_net group=1channel=1-15,17-31 #--- Make sure your [trunkgroups] section is empty! I lost a lot of time on that one myself! Zaptel.conf: #--- span=1,0,0,ccs,hdb3bchan=1-15,17-31dchan=16loadzone=bedefaultzone=be#--- We don't use crc4 here, but you can add it if you wish. Good luck! K On 6/8/06, Muhammad Zeeshan Latif [EMAIL PROTECTED] wrote: Hi I want to connect asterisk 1.0.9 ( kernel 2.6.8-2 debian )with TE110P and Nortel meridian option 11c release 25.40 with NTBK50AA card which is An E1 card. But the main problem is the first stage that no sync occurs the * card never syncs with meridian card I am using euroisdn, ccs , hdb3, crc4 , pri_net on asterisk And I am assuming that meridian is using same as it is connected to Nortel passport mvpe card which is an e1 isdn card and using the same config as astresk but the card never see each other. On the contrary when I connect the asterisk with the Nortel passport mvpe card it does detect the mvpe card but the d chan flaps btwn up and down and the hell of HDLC BAD FCS messages appears on the cli of * . I have also tried yellow alarm on the span but not of any help . Can any one tell me the config of meridian option 11c and asterisk and what I am doing wrong. thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN master clock issue ?
I 'll make some tests with a TE210P and see what happens, I'll post as soon as I have results... Asterisk is planned to be at the end of every PRI connection, providing voip to the PBX and IVR to the customers calling on the 2 E1 lines connected to Asterisk ... The MATRA PBX is connected to its own E1 line and Asterisk is directly connected to a secondary S2 bus. Setting this S2 bus to be a slave cannot be done easily as we have to manipulate tables inside the Matra PBX... I don't know how to do that... Anyway I'll test it and see what happens, but if I understand well to be totally sure there'll be no troubles I should buy a Sangoma card no ? Rich Adamson a écrit : Tristan wrote: The fact is that I have 2 different E1 (euroisdn) providers and an E1 (euroisdn) connection to a Matra PBX... The PBX needs to be master and as far as I know the PSTN providers needs it too... So I want to be sure that the quad E1 card I'll buy will work without troubles in this kind of setup... Can I only put the synchro to the PBX and forget about the PRI providers ? No. It is highly unlikely the PBX truly needs to be master. If that were an actual requirement, the vendor would never be able to sell their PBX into any environment where they connected to a E1 provider. The telco providers never slave their equipment from a customer-owned PBX. You will need to find the option in the Matra PBX to define it as syncing from the E1. (In fact, I'd bet a small amount of money the default implementation in the PBX is to sync from any attached E1.) There is a 99.99% probability the two E1 providers obtain their clock sync from a higher level (hierarchical source), and are already in sync with each other. If you use a digium card, you select one of the providers as your first choice sync source, and the second provider as your second choice sync source when the first choice provider's E1 is down. You definitely want your digium/sangoma card to support the hierarchical design of the digital network, and that well known design requires you to sync from your upstream provider, and pass that sync along to your downstream PBX. If you don't do that, calls originating from the PBX and passing through the digium/sangoma card to the PRI network will incur clock slips (out of sync). If the clock slips are too great, you will experience clicks, etc, during a call. Also, if all the components are not in sync, any use of modems (eg, faxes or pc modems) will be significantly degraded if not impossible to use. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
Use format_mp3 from asterisk-addons. It will enable your * to play mp3 without the use of an external process... (if I got it right) On 6/8/06, Richard Reina [EMAIL PROTECTED] wrote: Turby,Thanks for your replay, but does this mean that * can't play mp3s? I was hoping not to have convert the MOH mp3s that came with * and are supposed to be the native default that * is supposed to be able to play -- not to mention that I don't have sox installed with whick to convert them. Does this mean that 1.2.7.1 has a bug? If so can someone tell me if I should, and how I would go about reporting it.Thanks again for the reply. turby [EMAIL PROTECTED] wrote: convert the moh sounfile to pcm or sln save the file to /var/lib/asterisk/moh/default set the musiconhold.conf [default]mode=filesdirectory=/var/lib/asterisk/moh/default turby@ www.canistec.com From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Richard ReinaSent: Wednesday, June 07, 2006 5:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install Thank you very much for your relply. No I did not install mpg123 as the instructions at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conffor version 1.2 say the mpg123 is no longer needed.| Rurouni Alucard | [EMAIL PROTECTED] wrote: Did you check your mpg123 version ?, asterisk needs a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincluding installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.However, no sound is heard and I get this message from the CLI when accessing MOH: -- Started music on hold, class 'default', on channel 'Zap/19-1'-- Stoped music on hold on Zap/19-1This happens whether it's a parked call or whether I access MOH directly via:exten = 800,1,Answer exten = 800,2,MusicOnHold()Any help would be greatly appreciated.Thank you very much.Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan-capi and dtmf
Hi List, I'm having a problem with detecting incoming dtmf tones with chan_capi, using an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0, expecting that the capi module will detect the tones, but it did not. I also set both to 1, expecting that the asterisk dsp functions will detect them but it did not either. Can anyone provide any ideas how to overcome this issue? Esteban _ realestate.com.au: the biggest address in property http://ninemsn.realestate.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config Revision Control
On 06/03/06 22:10 Kevin P. Fleming said the following: - Michiel van Baak [EMAIL PROTECTED] wrote: Then the svn automerge thingie Kevin wrote for the asterisk svn tree is automerging changes to the 'common' tree to all the server trees. unrelated to asterisk obviously, but is there somewhere i can download the svn automerge patch of kevin's ? i'd love to have automerge running on our internal svn servers here. :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unicall local_unblocking_expired error
Not sure, may be somebody else can confirm what im going to tell. From reading the code, it seems the expired timer means the other end have not recognized the Idle status of your local end (your box). When you start Asterisk, chan_unicall set the ABCD bits to the unblocked status and start the timer with BLOCKING_RELEASE_TIME seconds as grace time for the other end to detect your new unblocked status. It seems to me that is not a problem. In fact chan_unicall use WARNINGS for every debug message, so I think is not even a true warning. What real problem do you have? cannot place calls? On 6/7/06, Frederic Jean [EMAIL PROTECTED] wrote: Hello all, and especially Steve, It seems my libunicall installation is having a little problem when initializing. Should I play with these ? #define DEFAULT_BLOCKING_RELEASE_TIME 450 #define DEFAULT_ANSWER_GUARD_TIME 100 #define DEFAULT_RELEASE_GUARD_TIME 20 #define DEFAULT_T1 15000 #define DEFAULT_T1A 150 #define DEFAULT_T1B 6 #define DEFAULT_T2 24000 #define DEFAULT_T3 15000 #define DEFAULT_MAX_SEIZE_ACK_WAIT 2000 #define DEFAULT_MAX_WAIT_FOR_GROUP_B_SIGNAL 15000 #define DEFAULT_MAX_AWAIT_ANSWER6 Thanks for any inputs, Fred --- Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/3 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/2 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/5 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/4 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/8 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/7 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/6 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/10 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/9 local_unblocking_expired Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/1 event Local end unblocked -- Unicall/1 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/3 event Local end unblocked -- Unicall/3 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/2 event Local end unblocked -- Unicall/2 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/5 event Local end unblocked -- Unicall/5 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/4 event Local end unblocked -- Unicall/4 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/8 event Local end unblocked -- Unicall/8 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/7 event Local end unblocked -- Unicall/7 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/6 event Local end unblocked -- Unicall/6 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/10 event Local end unblocked -- Unicall/10 local unblocked Jun 7 15:43:38 WARNING[3445]: chan_unicall.c:2694 handle_uc_event: Unicall/9 event Local end unblocked -- Unicall/9 local unblocked -- I have 10 channels configured and I get this output: SNET-PBX*CLI UC show channels Channel Extension Context Status Language MusicOnHold 21externalIdle br default 20externalIdle br default 19externalIdle br default 18externalIdle br default 17externalIdle br default 15externalIdle br default 14externalIdle br default 13externalIdle br default 12externalIdle br default 11externalIdle br default 10externalIdle br default 9externalIdle br default 8externalIdle br default 7externalIdle br default 6externalIdle br default 5externalIdle br
[Asterisk-Users] Anyone with GSM488 experience?
I need another fxo line. Has anyone had any experience with connecting the gsm488 into asterisk? Thanks, Jim. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip.c on debian testing - weird
Greetings all: In sip.conf, I have configured an entry for Australian VoIP provider Engin. Sometimes, however, the following error turns up constantly WARNING: chan_sip.c: Don't know how to indicate condition 9 ERROR: . channel.c: Unable to handle indication 9 for 'SIP/engin-5a0a' This is followed by unreliable connectivity. Any ideas on what I might do to fix it? Cheers, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.7.1 bad file descriptor
Hi all,
could someone tell me what this does mean bad file descriptor when
trying to start asterisk. It goes till the CLI command and then die with
this message. Below an strace output from asterisk -vc
It's on debian Sarge kernel 2.6.7 with packages from debian VoIP team.
The server was running fine till now with this version. Filesystem is
ok, checked with fsck (ext3).
Thanks
stat64(/etc/asterisk/enum.conf, {st_mode=S_IFREG|0660, st_size=586,
...}) = 0
== Parsing '/etc/asterisk/enum.conf': ) = 61n..., 61
open(/etc/asterisk/enum.conf, O_RDONLY) = 9
write(1, Found\n, 6Found
) = 6
fstat64(9, {st_mode=S_IFREG|0660, st_size=586, ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1,
0) = 0x4103f000
read(9, ;\n; ENUM Configuration for resol..., 4096) = 586
read(9, , 4096) = 0
close(9)= 0
munmap(0x4103f000, 4096)= 0
Asterisk Ready.[1;37;40mAsterisk Ready.\n, 27
) = 27
write(1, \33[0;37;40m, 10)= 10
rt_sigprocmask(SIG_UNBLOCK, [HUP INT PIPE TERM WINCH], NULL, 8) = 0
time([1149625396]) = 1149625396
rt_sigprocmask(SIG_BLOCK, [INT], [], 8) = 0
ioctl(0, TIOCGWINSZ, {ws_row=37, ws_col=111, ws_xpixel=0, ws_ypixel=0}) = 0
ioctl(0, TIOCSWINSZ, {ws_row=37, ws_col=111, ws_xpixel=0, ws_ypixel=0}) = 0
ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, {B38400 opost isig icanon echo
...}) = 0
ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, {B38400 opost isig icanon echo
...}) = 0
ioctl(0, SNDCTL_TMR_STOP or TCSETSW, {B38400 opost isig -icanon -echo
...}) = 0
ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, {B38400 opost isig -icanon -echo
...}) = 0
rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0
rt_sigaction(SIGINT, {0x4003e1a0, [], 0}, {0x80a6420, [INT],
SA_RESTART}, 8) = 0
rt_sigaction(SIGTERM, {0x4003e1a0, [], 0}, {0x80a6420, [TERM],
SA_RESTART}, 8) = 0
rt_sigaction(SIGQUIT, {0x4003e1a0, [], 0}, {SIG_DFL}, 8) = 0
rt_sigaction(SIGALRM, {0x4003e1a0, [], 0}, {SIG_DFL}, 8) = 0
rt_sigaction(SIGTSTP, {0x4003e1a0, [], 0}, {SIG_DFL}, 8) = 0
rt_sigaction(SIGTTOU, {0x4003e1a0, [], 0}, {SIG_DFL}, 8) = 0
rt_sigaction(SIGTTIN, {0x4003e1a0, [], 0}, {SIG_DFL}, 8) = 0
rt_sigaction(SIGWINCH, {0x4003e270, [], SA_RESTART}, {SIG_DFL}, 8) = 0
write(1, *CLI , 6*CLI ) = 6
rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0
read(-1, 0xb830, 511) = -1 EBADF (Bad file descriptor)
write(2, \nDisconnected from Asterisk serv..., 57
) = 57
open(/home/dh/.asterisk_history, O_WRONLY|O_CREAT|O_TRUNC, 0600) = 9
write(9, , 0) = 0
close(9)= 0
write(1, Executing last minute cleanups\n, 31Executing last minute
cleanups
) = 31
tgkill(2974, 2974, SIGURG) = 0
--- SIGURG (Urgent I/O condition) @ 0 (0) ---
rt_sigaction(SIGURG, {0x80a5cc0, [URG], SA_RESTART}, {0x80a5cc0, [URG],
SA_RESTART}, 8) = 0
sigreturn() = ? (mask now [])
== Destroying musiconhold processes;40mDestro..., 59
) = 59
tgkill(2974, 2974, SIGURG) = 0
--- SIGURG (Urgent I/O condition) @ 0 (0) ---
rt_sigaction(SIGURG, {0x80a5cc0, [URG], SA_RESTART}, {0x80a5cc0, [URG],
SA_RESTART}, 8) = 0
sigreturn() = ? (mask now [])
time(NULL) = 1149625396
kill(2980, SIGKILL) = 0
poll([{fd=8, events=POLLIN|POLLPRI, revents=POLLIN}], 1, 100) = 1
read(8, \1\0\377\377\1\0\377\377\0\0\0\0\0\0\1\0\377\377\0\0\0...,
8192) = 8192
time(NULL) = 1149625396
poll([{fd=8, events=POLLIN|POLLPRI, revents=POLLHUP}], 1, 100) = 1
read(8, , 8192) = 0
close(8)= 0
Asterisk cleanly ending (0). ending (0).\n, 30
) = 30
tgkill(2974, 2974, SIGURG) = 0
--- SIGURG (Urgent I/O condition) @ 0 (0) ---
rt_sigaction(SIGURG, {0x80a5cc0, [URG], SA_RESTART}, {0x80a5cc0, [URG],
SA_RESTART}, 8) = 0
sigreturn() = ? (mask now [])
close(3)= 0
unlink(/var/run/asterisk/asterisk.pid) = 0
write(1, \33[0m, 4) = 4
munmap(0x40407000, 4096)= 0
munmap(0x40405000, 4096)= 0
munmap(0x40018000, 4096)= 0
exit_group(0) = ?
--
Daniel
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
- trixter aka Bret McDanel [EMAIL PROTECTED] wrote: but they do in 2004 mark said it was one of their biggest revenue streams. Or do you mean that they dont make any money selling asterisk Please post a link (or something) to this quote; selling G.729 licenses has never been a significant revenue stream for Digium, and certainly not the 'biggest'. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP to SIP connection problem
Martin Joseph wrote: On Jun 7, 2006, at 6:55 PM, M.Hockings wrote: I have a small asterisk setup here with one POTS line, one VOIP SIP connection an FXS connection to the house phones and a bunch of softphones. Local calls are routed out through the POTS line and long distance through the VOIP line. This works great for the old house phones but the softphones on the computers can only make local calls. That is any attempt to connect through the VOIP line end in silence as soon as the called party picks up and asterisk attempts to connect the VOIP SIP connection and the softphone SIP connection. This is using xTen softphones on Linux and Windows. I was thinking that it might have to do with mismatched codecs or some such? In the [general] section of the sip.conf I see that freePBX has put disallow=all allow=ulaw allow=alaw and none of the softphone definitions set any different requirements. If I connect a softphone directly to the VOIP provider it appears to use the g711u codec. This is all using asterisk 1.2.9.1 and freepbx 2.1.1 running on CentOS 4.3. Thanks for any suggestions. Sounds more like a port issue to me. Looking in the asterisk Console and setting verbosity up when attempting these calls might give you more info. Also, you might try using an IAX softphone instead, as these are much less of a hassle in my opinion. There are several available. Marty Hi Marty, can you expand on the port issue a bit. I will admit that my understanding of sip connection handling is still a bit weak yet. I can say that the VOIP provider is on the far side of a firewall from the asterisk box and seems to work OK when talking to an old phone on a Digium connection. Also the softphones are on the same side of the firewall as the asterisk box. Is this a case that asterisk is trying to directly connect the VOIP sip connection and the softphone sip connection to each other or do both connect to asterisk and it manages the data flow between the two? So, I'm not sure how an IAX softphone would help other than forcing asterisk to be between the voip sip and the softphone iax connection? Again, thanks for any thoughts or suggestions as to how to get this to work right. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plainvoip problem.
Do you have the g729 codec? On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote: When calling through Plainvoip from my Asterisk at Home box I get the following log entries. Jun 8 01:27:25 VERBOSE[5550] logger.c: -- Called plainvoip/1insert # Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Call accepted by 66.199.240.2 (format g729) Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729 Jun 8 01:27:26 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 is ringing Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 stopped sounds Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 answered SIP/503-6d4c Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256 What I hear on the phone is one ring and then nothing. This has only been in the past few days. Has anybody else had a problem like this? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
- Matt Florell [EMAIL PROTECTED] wrote: fixed within a couple weeks and the Digium side being fixed by having to manually disable the hardware DTMF detection in the wct4xxp.c driver code every time I upgrade zaptel. This is no longer needed (editing the source); there is a module parameter that can be used to control this functionality as well, so you can place it into /etc/modprobe.d/matts-power-rules and it will take effect on each module load :-) Is has a configurable tail length and is capable of dynamically being turned on and off as needed by it's firmware. The Digium card uses an Oki chipset that has a smaller echo tail length and is hard-coded into the firmware so you cannot change it. Small correction: the first generation VPM chips were manufactured by Oki, the current ones are manufactured at another facility... but neither of those companies designed them. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
- Steve Underwood [EMAIL PROTECTED] wrote: other DSP functions for telecoms. What makes you think these are foundry chips? They are (were). They are now being manufactured at a different facility. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
- Matt Riddell (IT) [EMAIL PROTECTED] wrote: What does the onboard DSP do when used with Asterisk? Did Digium or someone put code inside Asterisk to hand off the processing/transcoding to a Sangoma card? According the Sangoma data sheet, the Octasic part _is_ the DSP (which it is, in a logical sense). The board does not relieve Asterisk/Zaptel of any additional burden beyond echo cancellation and tone detection at this time; Asterisk/Zaptel don't know how to take advantage of any of the more advanced Octasic features yet. And yes, when Digium's Octasic-based module starts shipping (currently in beta testing), it will offer the identical functionality, so I guess we can say our boards have 'DSP processing' too :-) -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RSA Signature (key ***) failed
In a dual server configuration one of the two servers fail with:
WARNING[3705]: res_crypto.c:335 __ast_sign_bin: RSA Signature (key
sintel-voip) failed
NOTICE[3705]: chan_iax2.c:5187 authenticate: Unable to sign challenge
withy key
WARNING[3705]: chan_iax2.c:7191 socket_read: I don't know how to
authenticate sintel-user to 10.27.33.1
We use the rsa authenticatio: in the server A (mickymouse) we have
generated a two pair of key sintel-voip, while in the server B we have
generated a pair of keys mickymouse-voip.
What is strange is that the two configurations are exactly simmetric,
but only one of the two server fail to call the internal phone of the
simmetric asterisk server. What is more ever strange is that if I
replace the authenticate mode with md5 in [sintel-user] and
[mickymouse-peer] sections below, (and the correponding dial command),
all works fine!
Here is an extract from the server A (mickymouse)
iax.conf:
...
[sintel-peer]
type=peer
host=192.168.100.1
auth=rsa
qualify=yes
[sintel-user]
type=user
auth=rsa
inkeys=sintel-voip
context=sintel-int
qualify=yes
...
extensions.conf:
[voip_outcoming]
exten =
_2.,1,Macro(callout,IAX2/mickymouse-user:[EMAIL PROTECTED]:4569/[EMAIL PROTECTED])
[macro-callout]
exten = s,1,Dial(${ARG1},60,jtTwW)
exten = s,2,Hangup
exten = s,102,Answer
exten = s,103,Playtones(busy)
exten = s,202,Answer
exten = s,203,Wait(1)
exten = s,204,Playback(privacy-incorrect)
exten = s,205,Wait(10)
exten = s,206,Hangup
[sintel-int]
exten = _4.,1,Macro(callout,SIP/${EXTEN})
Here is the extract from the server B (sintel):
iax.conf:
[mickymouse-peer]
type=peer
host=10.27.33.1
auth=rsa
qualify=yes
[mickymouse-user]
type=user
auth=rsa
inkeys=mickymouse-voip
context=mickymouse-int
qualify=yes
extension.conf:
[general]
static=yes
writeprotect=no
[mickymouse-int]
exten = _2.,1,Macro(chiamain,${EXTEN})
exten = _2.,2,Hangup
[local]
exten =
_24.,1,Macro(chiamaout,IAX2/sintel-user:[EMAIL PROTECTED]:4569/${EXTEN:[EMAIL PROTECTED])
[chiamaout]
[macro-chiamaout]
exten = s,1,Dial(${ARG1},60,jtTwW)
exten = s,2,Hangup
exten = s,102,Answer
exten = s,103,Playtones(busy)
exten = s,202,Answer
exten = s,203,Wait(1)
exten = s,204,Playback(privacy-incorrect)
exten = s,205,Wait(10)
exten = s,206,Hangup
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Latest SVN with downloaded sounds. Update
- Dave Cotton [EMAIL PROTECTED] wrote: Update after another look it isn't, there is only a gsm version. That is correct; the Spanish sounds and the non-GSM sounds will not be available until Asterisk 1.4 is released. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to identify agi crash cause
Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the SAY NUMBER and GET DATA agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and debugged but could not reproduce the problem. I also tried enabling core file generation by specifying the following command in /etc/profile ulimit -c unlimited /dev/null 21 but to no avail, I did not get any core file in /tmp or other locations. Can any one suggest a way to get a core dump of crashing agi's or some other way I can isolate the problem. Any help will be appreciated. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
- Matt [EMAIL PROTECTED] wrote: I'm not using mpg123... I'm using NATIVE MOH! No, the native file playback method does not offer any means to manipulate the volume of the sound being played. If you need to, you can edit the MOH files themselves using your tool of choice (sox, Audacity, etc.) to set the desired volume level. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dial pattern
Hello, I have to dial prefix 9 for non local numbers however when i missed calls i Can't redial this number because of 9 is not append . I use polycom phones . What Can i do ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
Jason Lixfeld wrote: If you have to use it, make sure you only use the mpg123 bundled with the asterisk distribution. mpg123 from any other source (yes, evem the developer's website) will yield major issues. mpg123 is NOT bundled with Asteirsk. make mpg123 will DOWNLOAD the mpg123 source and compile it. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan-capi and dtmf
On Thu, 8 Jun 2006, Esteban Guana-Jarrin wrote: Hi List, I'm having a problem with detecting incoming dtmf tones with chan_capi, using an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0, expecting that the capi module will detect the tones, but it did not. I also set both to 1, expecting that the asterisk dsp functions will detect them but it did not either. Can anyone provide any ideas how to overcome this issue? CAPI itself cannot detect DTMF. If the CAPI card driver reports that it can detect DTMF, then chan-capi will activate that function automatically. You can verify that when calling tool capiinfo on the shell, but I don't know if AVM Fritz can do that. When you set softdtmf/relaxdtmf, then the voice stream is just sent to Asterisk for dsp processing. If this does not work, I don't have any idea. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: wctdm.c RING_DEBOUNCE
Hi All, I have setup [EMAIL PROTECTED] 2.8 and using Digium TDM400P cards Whenever I dial out and finish the conversation and put the SIP Snom320 phone down, it rings back twice!!! If you pick up the phone there is no answer.although you think it's a genuine call!! If I change the RING_DEBOUNCE value in wctdm.c from 64 to 128 and then recompiling zaptel would it resolve problem?? I have also attached the logs capture after a call has been made; please can anyone help on how to stop this. Regards Ash __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/200-c98a' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/200-c98a' -- Executing Macro(SIP/200-d6c5, dialout-trunk|1|90775x||) in new stack -- Executing GotoIf(SIP/200-d6c5, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/200-d6c5, user-callerid) in new stack -- Executing DBget(SIP/200-d6c5, AMPUSER=DEVICE/200/user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=200/user -- DBget: set variable AMPUSER to 200 -- Executing DBget(SIP/200-d6c5, AMPUSERCIDNAME=AMPUSER/200/cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=200/cidname -- DBget: set variable AMPUSERCIDNAME to Reception -- Executing GotoIf(SIP/200-d6c5, 0?5) in new stack -- Executing SetCallerID(SIP/200-d6c5, Reception 200) in new stack -- Executing NoOp(SIP/200-d6c5, Using CallerID Reception 200) in new stack -- Executing Macro(SIP/200-d6c5, record-enable|200|OUT) in new stack -- Executing GotoIf(SIP/200-d6c5, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/200-d6c5, recordingcheck|20060606-110927|1149588567.614) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060606-110927|1149588567.614: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/200-d6c5, No recording needed) in new stack -- Executing Macro(SIP/200-d6c5, outbound-callerid|1) in new stack -- Executing DBget(SIP/200-d6c5, USEROUTCID=AMPUSER/200/outboundcid) in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=200/outboundcid -- DBget: set variable USEROUTCID to -- Executing GotoIf(SIP/200-d6c5, 0?4) in new stack -- Executing SetCallerID(SIP/200-d6c5, 02077292040) in new stack -- Executing GotoIf(SIP/200-d6c5, 1?6) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing NoOp(SIP/200-d6c5, CallerID set to 02077292040) in new stack -- Executing SetGroup(SIP/200-d6c5, OUT_1) in new stack -- Executing CheckGroup(SIP/200-d6c5, ) in new stack -- Executing SetVar(SIP/200-d6c5, DIAL_NUMBER=90775x) in new stack -- Executing SetVar(SIP/200-d6c5, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/200-d6c5, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Removed prefix. New number: 0775xx -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(SIP/200-d6c5, OUTNUM=0775xxx) in new stack -- Executing Cut(SIP/200-d6c5, custom=OUT_1|:|1) in new stack -- Executing GotoIf(SIP/200-d6c5, 0?16) in new stack -- Executing Dial(SIP/200-d6c5, ZAP/g0/0775xxx) in new stack -- Called g0/0775 -- Zap/1-1 answered SIP/200-d6c5 -- Hungup 'Zap/1-1' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/200-d6c5' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/200-d6c5' == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'Zap/1-1' -- Executing Set(Zap/1-1, FROM_DID=s) in new stack -- Executing Goto(Zap/1-1, ext-group|1|1) in new stack -- Goto (ext-group,1,1) -- Executing Macro(Zap/1-1, user-callerid|) in new stack -- Executing DBget(Zap/1-1, AMPUSER=DEVICE//user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=/user -- DBget: Value not found in database. -- Executing DBget(Zap/1-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname -- DBget: Value not found in database. -- Executing GotoIf(Zap/1-1, 1?5) in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing GotoIf(Zap/1-1, 0?NEWPREFIX) in new stack -- Executing Set(Zap/1-1, CALLERID(name)=) in new stack -- Executing Set(Zap/1-1, RGPREFIX=) in new stack -- Executing
Re: [Asterisk-Users] ISDN master clock issue ?
Tristan wrote: I 'll make some tests with a TE210P and see what happens, I'll post as soon as I have results... Asterisk is planned to be at the end of every PRI connection, providing voip to the PBX and IVR to the customers calling on the 2 E1 lines connected to Asterisk ... The MATRA PBX is connected to its own E1 line and Asterisk is directly connected to a secondary S2 bus. Setting this S2 bus to be a slave cannot be done easily as we have to manipulate tables inside the Matra PBX... I don't know how to do that... Then hire someone that does. Anyway I'll test it and see what happens, but if I understand well to be totally sure there'll be no troubles I should buy a Sangoma card no ? If you design this correctly, either card will work. If you want to do a complete (and most accurate) test, then simply try to send a fax through whatever design you want. If it passes, voice will not be a problem. Rich Adamson a écrit : Tristan wrote: The fact is that I have 2 different E1 (euroisdn) providers and an E1 (euroisdn) connection to a Matra PBX... The PBX needs to be master and as far as I know the PSTN providers needs it too... So I want to be sure that the quad E1 card I'll buy will work without troubles in this kind of setup... Can I only put the synchro to the PBX and forget about the PRI providers ? No. It is highly unlikely the PBX truly needs to be master. If that were an actual requirement, the vendor would never be able to sell their PBX into any environment where they connected to a E1 provider. The telco providers never slave their equipment from a customer-owned PBX. You will need to find the option in the Matra PBX to define it as syncing from the E1. (In fact, I'd bet a small amount of money the default implementation in the PBX is to sync from any attached E1.) There is a 99.99% probability the two E1 providers obtain their clock sync from a higher level (hierarchical source), and are already in sync with each other. If you use a digium card, you select one of the providers as your first choice sync source, and the second provider as your second choice sync source when the first choice provider's E1 is down. You definitely want your digium/sangoma card to support the hierarchical design of the digital network, and that well known design requires you to sync from your upstream provider, and pass that sync along to your downstream PBX. If you don't do that, calls originating from the PBX and passing through the digium/sangoma card to the PRI network will incur clock slips (out of sync). If the clock slips are too great, you will experience clicks, etc, during a call. Also, if all the components are not in sync, any use of modems (eg, faxes or pc modems) will be significantly degraded if not impossible to use. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where has the outbound call directory gone
I have installed 1.2.9.1 and it has no /var/spool/asterisk/outgoing directory. I must have missed some change in this addition when upgrading. Does anyone know where the automatic outgoing call directory has gone? Jordan Novak Communications Technician ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RSA Signature (key ***) failed
In a dual server configuration one of the two servers fail with:
WARNING[3705]: res_crypto.c:335 __ast_sign_bin: RSA Signature (key
sintel-voip) failed
NOTICE[3705]: chan_iax2.c:5187 authenticate: Unable to sign challenge
withy key
WARNING[3705]: chan_iax2.c:7191 socket_read: I don't know how to
authenticate sintel-user to 10.27.33.1
We use the rsa authenticatio: in the server A (mickymouse) we have
generated a two pair of key sintel-voip, while in the server B we have
generated a pair of keys mickymouse-voip.
What is strange is that the two configurations are exactly simmetric,
but only one of the two server fail to call the internal phone of the
simmetric asterisk server. What is more ever strange is that if I
replace the authenticate mode with md5 in [sintel-user] and
[mickymouse-peer] sections below, (and the correponding dial command),
all works fine!
Here is an extract from the server A (mickymouse)
iax.conf:
...
[sintel-peer]
type=peer
host=192.168.100.1
auth=rsa
qualify=yes
[sintel-user]
type=user
auth=rsa
inkeys=sintel-voip
context=sintel-int
qualify=yes
...
extensions.conf:
[voip_outcoming]
exten =
_2.,1,Macro(callout,IAX2/mickymouse-user:[EMAIL PROTECTED]:4569/[EMAIL
PROTECTED])
[macro-callout]
exten = s,1,Dial(${ARG1},60,jtTwW)
exten = s,2,Hangup
exten = s,102,Answer
exten = s,103,Playtones(busy)
exten = s,202,Answer
exten = s,203,Wait(1)
exten = s,204,Playback(privacy-incorrect)
exten = s,205,Wait(10)
exten = s,206,Hangup
[sintel-int]
exten = _4.,1,Macro(callout,SIP/${EXTEN})
Here is the extract from the server B (sintel):
iax.conf:
[mickymouse-peer]
type=peer
host=10.27.33.1
auth=rsa
qualify=yes
[mickymouse-user]
type=user
auth=rsa
inkeys=mickymouse-voip
context=mickymouse-int
qualify=yes
extension.conf:
[general]
static=yes
writeprotect=no
[mickymouse-int]
exten = _2.,1,Macro(chiamain,${EXTEN})
exten = _2.,2,Hangup
[local]
exten =
_24.,1,Macro(chiamaout,IAX2/sintel-user:[EMAIL PROTECTED]:4569/${EXTEN:[EMAIL
PROTECTED])
[chiamaout]
[macro-chiamaout]
exten = s,1,Dial(${ARG1},60,jtTwW)
exten = s,2,Hangup
exten = s,102,Answer
exten = s,103,Playtones(busy)
exten = s,202,Answer
exten = s,203,Wait(1)
exten = s,204,Playback(privacy-incorrect)
exten = s,205,Wait(10)
exten = s,206,Hangup
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does RELAXDTMF do?
Peter, Perhaps you have not followed the thread over the last few days about DTMF feedthru??? Here is what I sent out to another list kind of summing it up Regarding DTMF pass thru problems when using the SPA-3000 and *. The problem manifests itself as the inability to pass DTMF over the FXO to a PSTN call once the call is established. This would be used to call a bank, external voicemail or other service and use DTMF signaling to their service. To make a long story short (you can go thru the * mailist archives) this is an * problem in RFC-8233. It has been known for awhile and is being worked on in the form of a total RFC-8233 rewrite coming in 1.4 * hopefully this summer. Until then here is the fix I came up with. The FXO port Sipura setup (PSTN) should be set to INBAND for dtmf and the codec limited to g711u (or a), on the * side in sip.config FXO context set dtmf=inband and limit the codec to only g711u (or a) When you call yourself (say using your cell) and listen on the opposing phone hitting a key one listening on the other you should hear at least a half second or so of audible tone. Check this before and after changing these settings. Using RFC-8233 all I heard was a click and little or no audible tone. One other thing is that you CANNOT use features via tones over the FXO (TtWw,etc flags in dial). This is another broken issue in *. When you listen over the phone and hit a lead in character, defined in features.config, * mutes that character and it never gets sent. The correect action should be that it should mute it and wait until the second character. If the second character is not sent in a defined time then send the first character. This is not happenning. This might be an INBAND issue though and once RFC-8233 is fixed and can be used it might then work. If you have no need to send DTMF on a connected call via FXO then this change is not needed and you can use the current RFC-8233 as well as features. Just remember when you try to send DTMF over FXO port to PSTN that you know why it does not work!! This problem was/has been blamed on Sipura but is really an admitted * problem. It exists with other (but certainly not all) fxo devices also. As I said the best way to troubleshoot this is to actually call yourself and listen. Otherwise you are shooting in the dark and guessing. Doug On Thu, 8 Jun 2006, Peter J Dean wrote: I have an issue with DTMF. DTMF is being partly recognised by some external IVR systems (banks, billing, etc), other IVR systems have intermittent issues. Call our VSP directly and using their IVR system without issue, and our internal IVR works just fine. Currently i have all voip devices using RFC2833, which is what is recommended, and thus the sip.conf file has dtmfmode=rfc2833 and relaxdtmf=yes. I have not seen any information that clearly defines the purpose of the relaxdtmf parameter in the sip.conf file, and wondering of flicking it from yes to no will have an impact, and if so what sort of impact will it have? Redhat FC4 + updates Asterisk v1.2.9.1 SNOM v6.0.3 beta SPA3000 v3.1.10d ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plainvoip problem.
Do you have the g729 codec? On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote: Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729 ... Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256 Yes, and that works fine when talking with the phone itself, as you see the connection to the phone is g729. Then it changes from g729 to g729? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
2006/6/8, Kevin P. Fleming [EMAIL PROTECTED]: And yes, when Digium's Octasic-based module starts shipping (currently in beta testing),Could you elaborate ?Any schedule ?Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial pattern
Harry,You can use the prefix in your dial string instead of actually dialing it. Dial(Zap/g0/9${EXTEN})AlexOn 6/8/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,I have to dial prefix 9 for non local numbers howeverwhen i missed callsi Can't redial this numberbecause of 9 is not append .I use polycom phones .What Can i do ?Harry
__Do You Yahoo!?En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités
http://mail.yahoo.fr Yahoo! Mail___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial pattern
On 6/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have to dial prefix 9 for non local numbers however when i missed calls i Can't redial this number because of 9 is not append . I use polycom phones . What Can i do ? My preferred answer to this question is to not use a '9' prefix. This is a throwback to old analogue systems which needed the user to identify that a call was external because internal and external numbers could overlap to some degree. With a well designed modern digital system, this is usually not the case. Another method is to prefix any non-local callerID numbers with a '9' before Dial()ing the user, so that they are presented with the extended version of the number. The VoIP wiki has details of setting Caller ID in different versions of Asterisk. Hope that helps, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
Kevin P. Fleming wrote: - Matt Riddell (IT) [EMAIL PROTECTED] wrote: What does the onboard DSP do when used with Asterisk? Did Digium or someone put code inside Asterisk to hand off the processing/transcoding to a Sangoma card? According the Sangoma data sheet, the Octasic part _is_ the DSP (which it is, in a logical sense). The board does not relieve Asterisk/Zaptel of any additional burden beyond echo cancellation and tone detection at this time; Asterisk/Zaptel don't know how to take advantage of any of the more advanced Octasic features yet. And yes, when Digium's Octasic-based module starts shipping (currently in beta testing), it will offer the identical functionality, so I guess we can say our boards have 'DSP processing' too :-) Will it have a 1024 tap echo can on all 96 channels? What about 8 T1 support like sangoma? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP-2000
Is the 94x any better? seems without backlighting, any are next to useless. The SPA-9x2 have backlit displays. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial pattern
On Thu, 2006-06-08 at 16:28 +0200, [EMAIL PROTECTED] wrote: Hello, I have to dial prefix 9 for non local numbers however when i missed calls i Can't redial this number because of 9 is not append . I use polycom phones . What Can i do ? RTFM? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [HELP] - Sound cutting and dropping out - 1.2.7.1/Sangoma/PowerEdge 2850
Hi all,We are running Asterisk 1.2.7.1 on our Dell Poweredge 2850 and are having massive sound quality issues.We are experiencing call quality issues for our remote location, namely calls cutting out and breaking up for our agents. The two main issues seem to be 'popping' and 'dropping' - popping would be pops and crackles on the line, where dropping would be complete audio dropouts. Most of the time, these issues are occuring on ONE end of the audio stream. The building houses about 60 users, 30 or so of which are on calls at any one time. The location is connected to our main office via a 10Mbit low-latency fiber trunk, and gigabit switches on either side of the fiber endpoints. The floor at the remote location is all 100Mbit. Each user is running a Dell Optiplex 170L, 2.8GHz or greater, XP SP2, 256MB RAM, and Eyebeam 1.10n for their softphone, with ulaw as the codec. We have several managers that use Polycom IP501's, and they also are experiencing the issues. Server is a Dell Poweredge 2850, 2 x 2.8GHz Zeons, 4GB RAM, 73GB x 2 u320 hard drives in RAID-1, with a hotspare. Running a stock Fedora Core 4 install, with only mysql and apache running. Disabled ACPI and framebuffer, and have the Sangoma card interrrupting on CPU0 only, all other devices interrupting on CPU1. Using onboard gig-E NIC with current drivers. We are connected to the PSTN through a Sangoma A104D (current firmware), using EM Wink signalling. Sangoma drivers are the current 2.3.4-beta drivers recommended by Sangoma. I have spent the past three days working on this issue, and have opened the issue with Sangoma and Counterpath - neither has been very helpful. I have been monitoring our bandwidth closely - we're averaging around 3.5Mbit, so we should have plenty available. Sangoma statistics aren't showing anything out of the ordinary - the system is performing as it should. We have two other servers that are identical in configuration that serve the main office, and they have no sound quality issues whatsover - the only difference between the server having the issues and the ones that aren't is the connection to the users - one is a local LAN connection, the other is the WAN. I have set up an extension that calls the milliwatt app, and records the call to a file. I can call in from either a Zap or SIP channel and have sound quality issues, so the network is probably not causing the issue - a purely Zap channel still experiences pops and drops. Same with a purely SIP channel. The recorded call doesn't seem to reflect the audio issues - in other words, pops on the phone are not necessarily recorded into the file. We also have every call being recorded via the Monitor app - that was disabled early this morning to see if some of the issues with Monitor were causing problems - that made no difference either.We just upgraded from 1.2.4 to 1.2.7.1 about a half hour ago - so far, no difference.If I cannot come up with something by close of business today, I will completely rebuild the server from scratch, something I am not excited about doing. Otherwise, if anyone has any suggestions, questions, comments, or encouragement, I am in dire need of any/all.weswhaut at fc500 dot com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native Music On Hold Volume LOUD! How to adjust?
Kevin P. Fleming wrote: - Matt [EMAIL PROTECTED] wrote: I'm not using mpg123... I'm using NATIVE MOH! No, the native file playback method does not offer any means to manipulate the volume of the sound being played. If you need to, you can edit the MOH files themselves using your tool of choice (sox, Audacity, etc.) to set the desired volume level. With sox try -V 0.25 (or -v 0.25). I can't remember if it is an uppercase V or not. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail to Email on Blackberry
Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] early session audio on zap channel
Sorry about stupid question but I would liek to get
help about Zap channel.
We would like to get early media on session in
progress from zap channel.
But using the standard exten =
_X.,1,Dial(Zap/g1/${EXTEN}|60|o) I don't hear any audio until someone pickup the
phone.
Now I can't now if there is a message from a mobile
phone comany on session in progress.
please help.
regards
Rosario Pingaro
D. Lgs 196/2003Il presente messaggio contiene
informazioni confidenziali, indirizzate esclusivamente alle persone sopra
indicate. Se il ricevente non è tra dette persone, non dovrà intraprendere
alcuna azione, tipo copia, stampa o trasmettere il suo contenuto a terzi ed i
relativi allegati, ma solo informare il mittente dell'errore e cancellare il
messaggio. Il mittente dovrà, altresì, accertarsi che gli allegati non
contengano virus prima di aprirli.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
Mike Fedyk wrote: I have heard good things about the D-Link DES-1226G switch ($150 at newegg). If you can run a separate cable to the computer and phone. If you can't run the extra cables, then configure your phone to tag itself as part of the voip vlan and let the switch tag everything else as the computer vlan. I happen to have asterisk running as a router, so I use it doing QoS with tc (traffic control) and wondershaper set to prioritize based on port ranges. I sent a patch to the debian bug tracking system a while back with a few improvements -- I should check on that. It basically prioritizes smaller packets before larger packets with ~8 levels of priority and groups of sizes for the packets. Just doing that automatically handles 80% of the need for prioritization without specifying port ranges for the sip rtp packets. Mike Mike, Have you tried AstShape? Shapping based on port ranges is totally hit or miss. TOS is the way to go: http://www.krisk.org/files/astlinux-i586/usr/sbin/astshape Comment out the . /etc/rc.conf and you should be okay! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf
Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the file, changing its ownership, changing its permissions, restarting the portal, all without any success. Web resources on this issue claim the opposite problem - that custom changes to extensions_additional.conf will be automatically rewritten every time FreePBX/AMP is updated. If that was true, I'd be done - unfortunately, it seems this is not the case. I really don't want to reinstall FreePBX and redo my entire configuration again... :( Any assistance would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [HELP] - Sound cutting and dropping out - 1.2.7.1/Sangoma/PowerEdge 2850
On 6/8/06, whois wes [EMAIL PROTECTED] wrote: I can call in from either a Zap or SIP channel and have sound quality issues, so the network is probably not causing the issue - a purely Zap channel still experiences pops and drops. Same with a purely SIP channel. The recorded call doesn't seem to reflect the audio issues - in other words, pops on the phone are not necessarily recorded into the file. This is a very strange problem thank you for the very specific description you gave, looks like it has to be that server hardware. Have you tried possibly downgrading to Asterisk 1.2.6? not that I think it would be the cause, but recompiling Asterisk is a heck of a lot easier than rebuilding a server. I would also try swapping out the power supply. MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] early session audio on zap channel
Try
the 'g' option in your dial statement:
exten =
_X.,1,Dial(Zap/g1/${EXTEN}|60|og)
hth
-Original Message-From: Rosario Pingaro
[mailto:[EMAIL PROTECTED]Sent: Thursday, June 08, 2006 10:00
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] early session audio on zap channel
Sorry about stupid question but I would liek to
get help about Zap channel.
We would like to get early media on session in
progress from zap channel.
But using the standard exten =
_X.,1,Dial(Zap/g1/${EXTEN}|60|o) I don't hear any audio until someone pickup
the phone.
Now I can't now if there is a message from a
mobile phone comany on session in progress.
please help.
regards
Rosario Pingaro
D. Lgs 196/2003Il presente messaggio contiene
informazioni confidenziali, indirizzate esclusivamente alle persone sopra
indicate. Se il ricevente non è tra dette persone, non dovrà intraprendere
alcuna azione, tipo copia, stampa o trasmettere il suo contenuto a terzi ed i
relativi allegati, ma solo informare il mittente dell'errore e cancellare il
messaggio. Il mittente dovrà, altresì, accertarsi che gli allegati non
contengano virus prima di aprirli.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip
hello how can i configure asterisk to use soft sip phone and when asterisk is running how can I know he work correctly thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
- Olivier Krief [EMAIL PROTECTED] wrote: Could you elaborate ? Any schedule ? No, there is nothing really to elaborate... and this is not a commercial mailing list, so I'm not comfortable talking about it more here anyway :-) If you need more details, contact our sales department. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
- Mike Fedyk [EMAIL PROTECTED] wrote: Will it have a 1024 tap echo can on all 96 channels? What about 8 T1 support like sangoma? Those are completely unrelated questions; there is no need for an 8-span echo can module when there is no 8-span T1 card :-) It uses the identical Octasic part as the Sangoma board does, so the capabilities will be the same in that regard. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to identify agi crash cause
STDERR from your agi will be shown on asterisks tty. If youre using safe-asterisk to start, I believe this is redirected to tty9 Or, if you can afford to take asterisk down momentarily, you could just start asterisk without backgrounding it and youll see what your script has to say there. Josh McAllister From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danish Samad Sent: Thursday, June 08, 2006 8:25 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to identify agi crash cause Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the SAY NUMBER and GET DATA agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and debugged but could not reproduce the problem. I also tried enabling core file generation by specifying the following command in /etc/profile ulimit -c unlimited /dev/null 21 but to no avail, I did not get any core file in /tmp or other locations. Can any one suggest a way to get a core dump of crashing agi's or some other way I can isolate the problem. Any help will be appreciated. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail to Email on Blackberry
I couldn't find one but I didn't look too hard. To be honest, the Blackberry is so easy to use with one hand I dropped the issue. We actually switched to Windows Mobile devices which suck compared to the Blackberry for email/ease of use but I can now one click listen to my voicemail without dialing in (and using the horrible on screen only keypad of my new phone...which is the only reason I listen to the attachment via a phone now. Bill From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Thu 6/8/2006 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Voicemail to Email on Blackberry Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.techdatapros.com http://www.techdatapros.com/ winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
