Re: [asterisk-users] Asterisk-Addons 1.6.0.6 and 1.6.1.4 Now Available
-Original Message- From: Asterisk Development Team asteriskt...@digium.com Sent: Tuesday, June 08, 2010 11:20 AM To: asteriskt...@digium.com Subject: [asterisk-users] Asterisk-Addons 1.6.0.6 and 1.6.1.4 Now Available The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only. The releases of Asterisk-Addons 1.6.0.6 and 1.6.1.4 resolves issues reported by the community, and would have not been possible without your participation. Thank you! * chan_ooh323.c: Don't read rtp data from channel without private structure. (Closes issue #17227. Reported, tested by jin. Patched by may213) * chan_ooh323.c: Don't pass zero length callerid to ooh323 stack. (Closes issue #17186. Reported vmikhelson. Patched by may213) More information about the changes to maintenance support can be found at: http://www.asterisk.org/node/49924 Information about the Asterisk maintenance schedule is available at: http://www.asterisk.org/asterisk-versions For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-addons-1.6.06 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-addons-1.6.14 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content by MailScanner -- Scanned for viruses and dangerous content by MailScanner -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + TC400B - Clock Trouble
I'm not sure if the kernel timing HZ has anything to still do with things anymore. You might need to recompile your kernel with HZ=1000 -Jon lf...@leurent.eu wrote: Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call, it takes 25 seconds for the voice to go to the other end... I have Asterisk 1.4.25.1, Dahdi 2.2.0-rc5 on a CentOS-5.3 (x86_64) server with a 2.6.18-128.1.10.el5 linux kernel _Ast CLI when calling with g729_ ast-01*CLI transcoder show 1/1 encoders/decoders of 92 channels are in use. _Dahdi start returns:_ (SCREEN):r...@ast-01:[~]# /etc/init.d/dahdi start Loading DAHDI hardware modules: wctc4xxp:[ OK ] No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: [ OK ] _DMESG returns:_ dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0-rc5 dahdi_transcode: Loaded. wctc4xxp: tc400b0: Attached to device at :0f:03.0. wctc4xxp: tc400b0: (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) wctc4xxp: tc400b0: Installed a Wildcard TC: Wildcard TC400P+TC400M dahdi_transcode: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) dahdi_transcode: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) dahdi: Registered tone zone 30 (Switzerland) -- -- Marc LEURENT -- Scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content by MailScanner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Slackware 10.2
I believe I had to do the udev permissions file and also cause udevd to launch at bootup before modprobe'ing zaptel stuff. Check to make sure that udevd is launching automatically on bootup and that the udev rules and permissions are in place. -Jon T.S wrote: Yes I use Slackware 10.2, but I'm running kernel 2.4.31 Terrelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fernando Lujan Sent: Wednesday, May 17, 2006 3:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Slackware 10.2 Hi guys, I'm trying to use asterisk with my slackware 10.2 box. Kernel 2.6.13 from the testing... The udevd are not creating the /dev/zap devices. Someone already have success installing asterisk over slackware? Thanks in advance. Fernando Lujan ___ --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] Double Ring - TelIAX/Cisco 79[46]0
Try adding the following to sip.conf -- [general] progressinband=no -Jon Brent Torrenga wrote: Anyone experience the double ringing when calling out over TelIAX? I am using a Cisco 79[46]0, and do not use the r option in the Dial() command. I always thought that the r is what causes double ring, and is never really needed except to cause problems... Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.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 ___ --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] Transferring a call with IAX
I must be missing something here. Have you tried option "g" on your
dial command to the acd server? If option g is not specified, then dial
will hangup the call when exiting regaurdless of what the other iax box
did.
-Jon
Douglas Garstang wrote:
I just changed the macro to:
exten = s,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],25,wW)
exten = s,2,NoOp(${DIALSTATUS})
and the NoOp doesn't get executed. Bloody hell!
Console has:
-- Hungup 'IAX2/acdserver1-3'
== Spawn extension (macro-DialIAX, s, 1) exited non-zero on 'SIP/2944093-9ef2' in macro 'DialIAX'
== Spawn extension (macro-DialIAX, s, 1) exited non-zero on 'SIP/2944093-9ef2'
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Re: [Asterisk-Users] TDM2400P FXS Only vs. T1/E1 to FXS Channel Banks
Appreciate the thought on the handsets - but these lines will be going into an apartment complex - that is why faxing must work on any line and it must be analog. The astribank will not be a valid solution with the number of them I would require. -Jon Hans Witvliet wrote: On Thu, 2006-02-09 at 14:09 -0800, Jonathan Feally wrote: Hello All, I'm looking to get some feedback on which solution of providing FXS is going to have the best results with faxing. I'm only looking to see what method is going to provide the best digitization into Asterisk, not for transmission from Asterisk to else where. Any recommendations of specific channel banks are welcome. I will need to provide approximatly 216 FXS Ports and need to make sure my conversion from FXS to digital is the best I can get. Thanks in advance! -Jon Do you need 216 fax-lines From brief scan on the net: One TDM2400 with six FXS modules costs about 1700 euro's. You need nine (9*24=216) For hosting the TDM's, you'll probably need 5 machines, costing One Rhino channelbank with 24 lines cost 2700 euro's. You need nine (9*24=216) To interface to the rhino's you'll need 9 * T1 lines. TE411 are about 1900 Euro's With channelbanks, you might be spending a little bit more money, but you'll probably only need one ot two machines, instead of of pile. But why do you really need 216 POTS-lines? With channelbanks and T1 lines, you'll be spending about 130 euro's per line. You can have nice desktops phones for less. Why not one or two channelbanks and 200 new iax-phones? My 0,02 euro's ___ --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] No Voice when canreinvite=no
Upgrade to 1.2.4 - bug in 1.2.2 - see www.asterisk.org front page. -Jon Kamran Ahmad wrote: Hi all I am using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten = 500,1,Playback(demo-abouttotry) this is working). here is sip.conf //sip.conf// [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allow=all nat=no [6000] type=peer host=dynamic context=default canreinvite=yes allow=all [1000] type=peer host=dynamic secret=1000 canreinvite=yes allow=all __ 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 ___ --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] TDM2400P FXS Only vs. T1/E1 to FXS Channel Banks
Hello All, I'm looking to get some feedback on which solution of providing FXS is going to have the best results with faxing. I'm only looking to see what method is going to provide the best digitization into Asterisk, not for transmission from Asterisk to else where. Any recommendations of specific channel banks are welcome. I will need to provide approximatly 216 FXS Ports and need to make sure my conversion from FXS to digital is the best I can get. Thanks in advance! -Jon ___ --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] 911 and ISDN PRI
From me looking at it - it looks like the Telco is not accepting a
3 digit number. Have you tried 411 on the PRI to see if you are getting
the same error?
My 2 Cents
-Jon
Michael Collins wrote:
Joe,
It is
entirely possible, even probable,
that you spoke with someone who doesnt know the difference between PRI
and good ol fashion T1 trunks. If he insists that the channel
never comes up then he is definitely looking in the wrong place.
Assuming
hes talking about the B channel, obviously its not coming up
because thats what youre troubleshooting. If hes
insisting that the D channel isnt coming up then obviously none of
your
calls would be working, DID or otherwise.
Sounds like
youve got a case of vendor
wheel-of-blame going on. Please contact me off list and Ill be
happy to help you out. I used to be a vendor so I know the routine.
Ive
got a dozen T1s, half of which are PRIs, from 3 different telcos
so Im used to this kind of stuff.
-MC
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Joe Pukepail
Sent: Wednesday,
February 08, 2006
3:31 PM
To: Asterisk Users
Mailing List -
Non-Commercial Discussion
Subject: Re:
[Asterisk-Users] 911
and ISDN PRI
I've talked to the carrier (verizon), what
they said is that the call
is not leavingmy phone equipment. I tried to tell him that I'm getting
an
error back from his system, but he insists that the channel never comes
up. Their answers was "talk to your telco vendor, its on their
end". So I guess I'm pretty much SOL when it comes to using 911 with
the PRI.
Below is the debug, they wanted me to try all
the DID numbers to see if
it worked on any of them (40 numbers) and the billing number, wouldn't
work
with any of them.
-- Executing SetCallerID("IAX2/sycam-16384",
"8157548823") in new stack
-- Executing Dial("IAX2/sycam-16384",
"Zap/g2/911") in new stack
-- Making new call for cr 33385
-- Requested transfer capability: 0x00 - SPEECH
Protocol Discriminator: Q.931 (8) len=39
Call Ref: len= 2 (reference 617/0x269) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: u-Law (34)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 1 ]
[1e 02 80 83]I
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: User (0)
Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ]
[6c 0c 41 80 38 31 35 37 35 34 38 38 32 33]
Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4)
NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number not screened (0)
'8157548823'
]
[70 04 a1 39 31 31]
Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911' ]
-- Called g2/911
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 617/0x269) (Terminator)
Message type: RELEASE COMPLETE (90)
[08 02 82 9c]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Public network serving the local user (2)
Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ]
-- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 2 got hangup
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/25-1'
On
2/8/06, Watkins,
Bradley [EMAIL PROTECTED]
wrote:
It looks
like the outbound caller ID is
not being set properly. Most of the carriers that I've dealt with will
act exactly as you said if you do not set it to what is expected at the
911
center.
In
particular:
Calling Number (len= 8) [ Ext: 0 TON:
Subscriber Number
(4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number not screened (0)
'3251' ]
Your user
number being sent is just the
caller ID of the SIP channel.
Regards,
- Brad
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
] On Behalf Of Joe
Pukepail
Sent: Tuesday, February 07,
2006 3:26 PM
To:
Asterisk Users
Mailing List - Non-Commercial Discussion
Subject:
Re:
[Asterisk-Users] 911 and ISDN PRI
I have a call in with the carrier, below is
the PRI debug, looks like
it is getting hungup because of "Invalid Number format", I did try to
use Setcallerid to change the callerID to a DID number in a previous
attempt,
but it still didn't go through. Not sure if that "invalid
number format" is the calling number or the number I'm calling. I'll
let the
Re: [Asterisk-Users] ztdummy on opteron
This basicaly means you need to recompile the kernel with HZ=1000. On a 2.6.x kernel in make menuconfig you can find this under Processor type and features --- Timer frequency (1000 HZ) --- 100,250,1000 -Jon Sean Cook wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Been running asterisk in test for a while on ix86 and wanted to ramp up the system a bit. So we got a dual opteron server in for the production install and in my preliminary configuration i noticed that the insmod ztdummy produces the following: ztdummy: This module requires the kernel HZ setting to be 1000 ticks per second has anyone run across this? Is this just a bonehead mistake on my part? Sean -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFD0UT/y9wPyZpnL2URAhHhAKCHD98/YpHhvYEjmfjf7RyzclnoswCghijk x+VVfV2elg0cSsxbvSWgaBU= =mL9+ -END PGP SIGNATURE- ___ --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] Mediatrix windows-based setup?
You need the unit manager software that should have come with your box. Your box most likely only speaks SNMP, so this is the only tool I know that has the MIB's and setup to know how to set the MIB values. However there are many more tweaks in manually tuning some of the MIB's through the unit manager that can fine tune settings, such as setting a dialmap that will disable call-waiting, etc. If you don't have the original disk, contact me off-list so I can send you a zip of the CD. -Jon I am in no way a mediatrix expert! I have only played with 1124's. Kerry Garrison wrote: Can anyone recommend a tool that can be used on Windows XP to configure the Mediatrix 1204? -Kerry ___ --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] No native bridge on outbound SIP channels
I'm guessing that you have a similar entry in your sip.conf for the 7960?? The 7960 has a setting for preferred codec. It defaults to g711 U-Law. You might try changing this setting also as the 7960 doesn't know that you only want to speak A-Law. You will also want to make sure that the nat settings are disabled on both devices as they are on the same network. nat=never is a better choice than nat=no. You might also check your extensions.conf to verify that the calling from 1760 to 7960 is the same as from 7960 to 1760. You could also try moving both devices to using U-Law instead. -Jon Eric Bishop wrote: Hi all, I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via Asterisk. Both are running g711A codecs and SIP. On inbound calls I get a native bridge, however on outbound calls I never get a native bridge. With other SIP gateways I do get a native bridge on the outbound call. My sip.conf is as follows: [cisco1760] type=friend context=incoming host=192.168.0.55 insecure=yes nat=no canreinvite=no dtmfmode=rfc2833 disallow=all allow=alaw I have also confirmed while on an outbound calls that both are using the exact same codecs. sip show channels shows pbx*CLI sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.0.55 123456789 4ea2e1314cd 00102/0 alaw No Tx: ACK 192.168.0.58 200 0013c427-f4 00101/00102 alaw No Rx: ACK 2 active SIP channels Anyone have an idea what's going on? ___ --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] No native bridge on outbound SIP channels
You will probably want canreinvite=yes on your sip entries unless you are going to be using monitoring or some other feature in which asterisk needs to hear the conversation. Also, Is asterisk answering the call from the 7960 or is the 1760 doing it through the dial cmd? If asterisk answers the call, then this could be part of the problem. Can you send an output of the console for a call from 1760 - 7960 with a show channel for each SIP device, and then the same thing for 7960-1760. -Jon Eric Bishop wrote: Yes the 7960 is also set only to use alaw. I was under the impression though that nat=yes did not effect this. And if it does why does it native bridge ok on inbound calls with the same nat=yes On 1/15/06, Jonathan Feally [EMAIL PROTECTED] wrote: I'm guessing that you have a similar entry in your sip.conf for the 7960?? The 7960 has a setting for preferred codec. It defaults to g711 U-Law. You might try changing this setting also as the 7960 doesn't know that you only want to speak A-Law. You will also want to make sure that the nat settings are disabled on both devices as they are on the same network. nat=never is a better choice than nat=no. You might also check your extensions.conf to verify that the calling from 1760 to 7960 is the same as from 7960 to 1760. You could also try moving both devices to using U-Law instead. -Jon Eric Bishop wrote: Hi all, I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via Asterisk. Both are running g711A codecs and SIP. On inbound calls I get a native bridge, however on outbound calls I never get a native bridge. With other SIP gateways I do get a native bridge on the outbound call. My sip.conf is as follows: [cisco1760] type=friend context=incoming host=192.168.0.55 insecure=yes nat=no canreinvite=no dtmfmode=rfc2833 disallow=all allow=alaw I have also confirmed while on an outbound calls that both are using the exact same codecs. sip show channels shows pbx*CLI sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.0.55 123456789 4ea2e1314cd 00102/0 alaw No Tx: ACK 192.168.0.58 200 0013c427-f4 00101/00102 alaw No Rx: ACK 2 active SIP channels Anyone have an idea what's going on? ___ --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 ___ --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] Integrating with Toshiba Strata DK40i KSU
Philip, Your problem with dialing an extension on the Toshiba and only getting a second of music on hold has to deal with the fact that you are using an analog trunk. Asterisk will always say that the analog channel has answered as soon as it is done sending dtmf on the line. You could help hide this problem by adding a couple of w's to your dial string, but you could run into issues where the called extension answers and says hello, but is cut off. I'd say 3 w's would be your max, giving 1.5 seconds more of music on hold.. Ex: Dial(Zap/7-1, Zap/6/351www|5|m) As far as dialing multiple extensions, you need to setup a hunt group on your toshiba, then dial the hunt groups number instead of the individual extensions. The toshiba will then connect asterisk to the first extension that answers. You will most likely want to put busydetect=yes in your zapata.conf to help with busy signals and it may help with your phantom calls on hangups. Good Luck, -Jon Philip Edelbrock wrote: We've done a direct swap of an old Amanda voicemail system with a shiney new Asterisk system (Asterisk 1.0.9). The system consists of 4 FXO ports on the * box (TDM400P), and three old Wildcards we aren't using (too buggy we found). CO lines- Toshiba - FXO ports on * We want to branch out a little more and use it as an auto-attendant. The first problem seems to be an asterisk problem. When ringing extensions, it thinks the first ringback is an answer: == CDR updated on Zap/7-1 -- Executing Macro(Zap/7-1, dialexten|35) in new stack -- Executing Dial(Zap/7-1, Zap/6/351|5|m) in new stack -- Called 6/351 -- Started music on hold, class 'default', on Zap/7-1 -- Zap/6-1 answered Zap/7-1 -- Stopped music on hold on Zap/7-1 -- Attempting native bridge of Zap/7-1 and Zap/6-1 To the caller, they hear on-hold music for just a brief second, and then ringing. When they hang up, the lines remained bridged and the extension continues to ring until I log in and do some 'soft hangup' commands. The second problem is more of a Toshiba problem (or rather my lack of knowledge of). I hope that perhaps somebody might be able to help me? I want to have a way to ring multiple extensions if sombody, say, hits zero. The Toshiba can ring mutliple extensions for fresh new incoming calls, but once answered I can't seem to 'unanswer' the call to get it ringing at multiple stations (we have no designated reception phone that is staffed 100% of the time). Thanks! Phil ___ --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] Exit Voicemail
I am having the same issue. There was a patch put in that is supposed to rewite a blank context to default, but it looks like in the process this patch has killed the realtime variable passed to the query. -Jon C F wrote: Voicemail in itself does not hangup, * will bring you back to the DP (to exten a). So if a user exits VM (I think they can exit by pressing # after recording) then you can drop them in a context that does what you want, you can do the same at exten a. On 12/8/05, Joe Pukepail [EMAIL PROTECTED] wrote: Is there a way to have control go back to the dialplan after a call gets to voicemail? I'm looking to implement findme and campon, but I want the options to be "hidden", so if someone calling got a voicemail they could key in "*1" (or whatever) and it would go back to the dialplan so I can implement fineme in the dial plan. The same with campon, if you got a busy voicemail you could key in "*2" (or whatever) and it would take them to the piece of the dialplan where it would wait for person to get off the phone. I realize I could do this by having the user key in another option (Hit 1 to leave a voicemail, hit 2 to findme) but would prefer not to, users could record this as part of their voicemail message if they want the public to know about the findme and camping on a busy extension. ___ --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] Trying to genereate dial tone, but stop after first digit dialed.
I seem to be missing something here. Basically I'm trying to do what a full CO would do in terms of *70 to disable call waiting. I have a *70 exten setup, it does the work to set the extension to not take in a second call, then does a playtones(dialrecall). This works except that all digits dialed after the *70 have the tone still playing until the dialplan kicks back in for the new exten dialed. Does somebody have a work around for this? I'd prefer to not use Background. Thanks, -Jon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor and sox mix quality
I believe it comes with sox. Both my sox and normalize are in /usr/bin. Elmar Haneke wrote: NORMALIZE=nice -n 20 /usr/bin/normalize --no-progress -a 1.0 --peak Which package comes this normalize from? Elmar ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor and sox mix quality
I have not noticed any issues with quality, just with caller volumes
being way different when mixing 2 channel types (ZAP and SIP
specifically). Here's my custom script for processing the recording
files. Make sure you use option m on your monitor command so that the
custom script will run. My script makes a stereo mp3 with the 2 people
split to left/right and makes the 2 sides have an equal max volume. Hope
this helps. You can always modify the script to adjust how the mp3 is
encoded.
extensions.conf:
[globals]
MONITOR_EXEC=/usr/local/bin/2wav2mp3
[macro-callext]
s,1,monitor(wav|${ARG1}_${TIMESTAMP}|m)
s,2,dial(SIP/${ARG1})
[EMAIL PROTECTED]:/usr/local/bin# cat /usr/local/bin/2wav2mp3
#!/bin/sh
# 2wav2mp3 - create stereo mp3 out of two mono wav-files
# source files will be deleted
#
# usage: 2wav2mp3 wave1 wave2 mp3
#
# extensions.conf
# use option m on monitor command
# add this variable to [globals]
# MONITOR_EXEC=/usr/local/bin/2wav2mp3
# location of SOX and SOXMIX
# (set according to your system settings, eg. /usr/bin)
SOX=nice -n 20 /usr/bin/sox
SOXMIX=nice -n 20 /usr/bin/soxmix
LAME=nice -n 20 /usr/local/bin/lame -S --cbr -b32 -m s
NORMALIZE=nice -n 20 /usr/bin/normalize --no-progress -a 1.0 --peak
# command line variables
LEFT=`echo $1 | awk -F.wav '{print $1}'`
RIGHT=`echo $2 | awk -F.wav '{print $1}'`
OUT=`echo $3 | awk -F.wav '{print $1}'`
#test if input files exist
test ! -r $LEFT.wav exit
test ! -r $RIGHT.wav exit
# convert mono to stereo, adjust balance to -1/1
$NORMALIZE $LEFT.wav
$NORMALIZE $RIGHT.wav
# left channel
$SOX $LEFT.wav -c 2 $LEFT-tmp.wav pan -1
# right channel
$SOX $RIGHT.wav -c 2 $RIGHT-tmp.wav pan 1
# in case an old version of sox is used, encoding
# can be done afterwards
$SOXMIX -v 1 $LEFT-tmp.wav -v 1 $RIGHT-tmp.wav -v 1 $OUT.wav
$LAME $OUT.wav $OUT.mp3
#remove temporary files
test -w $LEFT-tmp.wav rm $LEFT-tmp.wav
test -w $RIGHT-tmp.wav rm $RIGHT-tmp.wav
test -w $OUT.wav rm $OUT.wav
#remove input files if successfull
test -r $OUT.mp3 rm $LEFT.wav $RIGHT.wav
# eof
Good Luck!
-Jon
[EMAIL PROTECTED] wrote:
Hello All,
I am using monitor with soxmix, however the quality seems somewhat low
after sox converts to mp3.
Does anyone know a way to get a higher quality file? Some of my lines
are coming in on isdn.
Regards,
Greg
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Re: [Asterisk-Users] PABX and Asterisk Dial Plan
You will want to use the D(digitstopluseindtmf) option on your dial cmd. That is a capital D for the option! ex. Dial(SIP/2100,D(1000)) -Jon Stephen wrote: Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone - Asterisk -- ATA (FXS) -- (CO side) PABX - Extension (eg. 1000) (2100 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by dialing 2100 from sip phone after PABX answer my call. But that's too troublesome, Can I just dial 21001000 instead ? which mean the first 4 numbers are for pabx and the next 4 numbers are for extension? Thanks, Stephen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatically setting mailbox on registration of SIP device by SIP device's line #
Hello, Is there any way to have a SIP device get it's mailbox automatically set upon registration when it does not have an entry in sip.conf? I would prefer to not define all my 200+ devices in my sip.conf's. All my CID is set via an AGI from database or ldap. All I need it to do is set mailbox=sip_line_#. If there is a patch in progress for this, I'd love to test it out. Thanks in advance. -Jonathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Poweredge 1850 and Zaptel
Adam Robins wrote: If anyone out there is running Asterisk with Zaptel and a TDM400P card on a Dell Poweredge 1850 server, please let me know what OS and kernel version you are running. I keep getting errors when modprobing zaptel and am running out of possibilities, other than motherboard incompatibility. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have just build an 1850 with a TDM410P on Slackware 10.1 - Works great - I did however upgrade my kernel to 2.4.30 from 2.4.29 - you will need to use scsi2.s to get it installed - then just make a new kernel with e1000 and megaraid2 drivers and any other options you like. So far it appears to be stable. -Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DS3 PCI in asterisk
Kevin P. Fleming wrote: Race Vanderdecken wrote: Why can't I convert the DS3 input to SIP Output, no transcoding, straight G.711, all in one box? Yes, that is what you would want to do. Probably even better would be DS-3 to IAX, and try to get trunking support for G.711 working to keep down the IP overhead as well. Yes I know the bandwidth from a DS3 to SIP calls means I need 2 DS3's worth of space back out the door. Not a big deal; use a server with dual PCI-X busses, put the DS-3 card on one and a GigE card on the other. You're still only pushing 120-150Mbps over the bus combined, which is not a problem at all. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I was just about to say follow the IAXy model (Just a wee bit more channels!) and make the box speak IAX instead of SIP/H.323. I also suggest setting up this standalone DS3-IAX box to have each channel assigned to a specifiy asterisk box. While an asterisk box may be able to run 600+ channels at a time, other limitations come into play, such as hard drive access for VM and other applications that run on the machine either seperate from asterisk or as a module/AGI of asterisk in which CPU and other resources can cause lag times, reducing the number of channels that can be processed jitter free. My 2 Cents -Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automated calling/Bridging and takedown in Asterisk?
Jack, I'd suggest using the .call files to initiate your call to phone a - the call script will automatically bridge the call to a destination station. From that you could simply create a goto loop in extenstions.conf that calls the same agi script over and over, allowing the agi to actually place the call to the party already on the phone, once the party called from the agi hangs up the process will repeat until phone a hangs up! I have writen some stuff in php that generates .call files and it so far seems to be solid. -Jonathan Jack Turer wrote: I am working on a web phone interface to give normal phonesets more 'virtual buttons'..etc, like the expensive executive phones via control via the web. This lead me to the following issue: I am wondering if it is possible (it doesn't seem to as far as I can tell) to make a script (AGI or otherwise) that will have asterisk automatically do the following without the user needing to originate any calls on their telephone: -Call an extension, hold on to the call (call A) -Call another extension, hold on to the call (call B) -Bridge the two calls (A and B) together (so the two extensions can talk to each other) -Later the script drops call B, but keeps call A up -Then asterisk calls another extension (call C) -Then asterisk bridges A with C so then they can talk to each other ..then later the same thing again (call D, then bridge with A)..etc...etc, Allthis would be AGI or script driven without any user having to press anything on his phone. Is this possible (the main issue I see in asterisk is that I cannot find a command in the asterisk API to bridge/unbridge calls like this without something being originated by a call into asterisk from a user). I looked at meetme, but it doesn't seem appropriate for what I want to do above. Any ideas? Thank you! Jack ___ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
