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
#
# 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] ASTCC speaks and cut RTP channel
This is not much help but it is more of a suggestion than I've seen on this list on this issue. Would you be able to try an older release of asterisk say 1.00 or right around there. I really do believe it was working back then but I've not played with that exact issue for a long time. Darren wiebe [EMAIL PROTECTED] Ricardo Poppi wrote: Hi list. I have a fine running Ser+Asterisk environment and have just installed ASTCC. It´s working fine either, including its caller-id authentication feature (the one we pass the card-number as CALLERID variable and number-to-dial as EXTEN variable). The issue, a great one, is that when the credit is about one minute to end, the ASTCC prompt gets into the call, says that "you have one minute left..." and when it was suppose to leave and let the RTP traffic of the original call be "reestablished", it never happens. The RTP packets - I could see that at asterisk debug screen - stop running and the call is still signaled as active, but no media at all. This is a serious problem I´m having and, as I could see, I´m not the only one. Mr. Chilini reported that around jun 30th this year, as you can see bellow: (I just added a comment at this voip-info page to see if anyone could give some clues about that) http://www.voip-info.org/tiki-index.php?page=ASTCCGuide#comments Do anyone here in this list had any situation alike? Do you have any clues do help me? (and others because it will be documented, of course). Thanks in advance, Ricardo Poppi. ___ --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] Cisco Callmanager & Asterisk for Voicemail revisited
> Some of you may remember back in May the thread on using Asterisk as a
> voicemail server for a Cisco Callmanager system.
>
> My own Callmanager system is integrated into an Asterisk server for
> voicemail (and other things). Back in May I was using H323 for
> integration, but since I've upgraded to CCM 4.1 I have switched over
> to SIP.
>
> The integration with H323 required using Call forwarding to send the
> call to an extension on Asterisk. For example, extension 7443 would
> forward to 27443 on Asterisk which looked something like:
> exten => _27XXX,1,Voicemail(u${EXTEN:1})
>
> Obviously setting this for each and every phone on Callmanager was not
> an option for any wide deployments, and Paul Davidson investigated
> some of the other options. Paul discovered that it was possible to
> setup a voicemail pilot, tick the voicemail box, etc. but you would
> lose the ability to have the caller ID information added to the
> voicemail.
>
> This wasn't an option for us, as caller ID is quite important.
>
> Up until now, I have continued with the custom extension option,
> setting up the appropriate call forwarding when new phones were added
> to the system. The trunk between CCM and Asterisk changed to SIP after
> the CCM upgrade, but everything else stayed the same until I revisited
> this today.
>
> To summarise what I have accomplished:
>
> Full voicemail integration between CCM and Asterisk with the following
> features:
> - MWI
> - Voicemail on the CCM side is enabled by selecting Forward to
> 'Voicemail' rather than yucky custom extensions. Allows for wide
> deployment.
> - Messages are accessed by pressing the 'Messages' button on the CCM
> phones, or dialing the VM pilot number.
> - If a CCM user doesn't want to take a call, they can press the
> iDivert softkey to send to voicemail immediately.
> - CCM users can forward all calls to voicemail in the ccmuser pages,
> or by pressing CFwdAll and entering the pilot number or messages.
> - All the standard Asterisk voicemail features work just fine, eg: vm to
> email.
> - more
>
> Bugs with the setup:
> - If there's a SIP device registered with the Asterisk machine
> handling the voicemail, and the call path is something like: Sip
> Device -> Asterisk -> CCM. If the call subsequently reaches voicemail,
> Asterisk prompts for proxy authentication and CCM drops the call. This
> problem can be avoided by using usernames that don't match the caller
> id, eg: [sip7345], or having a machine dedicated to Voicemail.
> - That's all I've found
>
> These options have been tested with Asterisk 1.0.8 and CCM 4.1(2)sr1.
> If this is something that people would be interested (and you made it
> this far), I'd be quite happy to whip up some instructions and add it
> to the wiki.
I recall several posts asking for this type of config over the last
year or so. Placing it on the wiki would be excellent. Make sure you
note the ccm version requirements (and why), etc, as someone is bound
to say something like 'I have ccm v3.x and I can't make it work'.
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Re: [Asterisk-Users] limiting calls per day based on amount of time
Hello, I'm going to be starting implementing something like this in ASTPP. I don't know of any very easy way to do it at present though. We did implement a solution similar to this using ASTCC. If you're interested email me off list. Darren Wiebe [EMAIL PROTECTED] Joseph wrote: How to implement call limitation based on amount of time call per day. I've implemented the Dial "L" parameter but in addition I would like to limit an extension to certain amount of call time per day. ___ --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] E1 configuration problem
Hi, First of all make sure you have inserted zaptel and wxfx* modules and Then execute ztcfg -vvv It will show you the channels configured and see its green signal now. :) Regards Gurminder On 9/19/05, manish kumar <[EMAIL PROTECTED]> wrote: > I am trying to configure E1 card (Digium) but not able to do that. The > green light doesn't come up when it starts. > > What can be the problem. I have also changed the jumper settings of the > card from T1 to E1 but still no relief. > > Thanks in advance > > Manish > > ___ > --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
[Asterisk-Users] E1 configuration problem
I am trying to configure E1 card (Digium) but not able to do that. The green light doesn't come up when it starts. What can be the problem. I have also changed the jumper settings of the card from T1 to E1 but still no relief. Thanks in advance Manish ___ --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
[Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited
Some of you may remember back in May the thread on using Asterisk as a
voicemail server for a Cisco Callmanager system.
My own Callmanager system is integrated into an Asterisk server for
voicemail (and other things). Back in May I was using H323 for
integration, but since I've upgraded to CCM 4.1 I have switched over
to SIP.
The integration with H323 required using Call forwarding to send the
call to an extension on Asterisk. For example, extension 7443 would
forward to 27443 on Asterisk which looked something like:
exten => _27XXX,1,Voicemail(u${EXTEN:1})
Obviously setting this for each and every phone on Callmanager was not
an option for any wide deployments, and Paul Davidson investigated
some of the other options. Paul discovered that it was possible to
setup a voicemail pilot, tick the voicemail box, etc. but you would
lose the ability to have the caller ID information added to the
voicemail.
This wasn't an option for us, as caller ID is quite important.
Up until now, I have continued with the custom extension option,
setting up the appropriate call forwarding when new phones were added
to the system. The trunk between CCM and Asterisk changed to SIP after
the CCM upgrade, but everything else stayed the same until I revisited
this today.
To summarise what I have accomplished:
Full voicemail integration between CCM and Asterisk with the following features:
- MWI
- Voicemail on the CCM side is enabled by selecting Forward to
'Voicemail' rather than yucky custom extensions. Allows for wide
deployment.
- Messages are accessed by pressing the 'Messages' button on the CCM
phones, or dialing the VM pilot number.
- If a CCM user doesn't want to take a call, they can press the
iDivert softkey to send to voicemail immediately.
- CCM users can forward all calls to voicemail in the ccmuser pages,
or by pressing CFwdAll and entering the pilot number or messages.
- All the standard Asterisk voicemail features work just fine, eg: vm to email.
- more
Bugs with the setup:
- If there's a SIP device registered with the Asterisk machine
handling the voicemail, and the call path is something like: Sip
Device -> Asterisk -> CCM. If the call subsequently reaches voicemail,
Asterisk prompts for proxy authentication and CCM drops the call. This
problem can be avoided by using usernames that don't match the caller
id, eg: [sip7345], or having a machine dedicated to Voicemail.
- That's all I've found
These options have been tested with Asterisk 1.0.8 and CCM 4.1(2)sr1.
If this is something that people would be interested (and you made it
this far), I'd be quite happy to whip up some instructions and add it
to the wiki.
-Shaun
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[Asterisk-Users] HW Question (TDM400)
Hi, all Did anyone tried TDM400 card on old main board (Intel 440LX chipset -- PII)? The reason I am asking is because TDM400 needs PCI2.2 and main board is PCI2.1 I do not want to upgrade yet. Thanks, Rudolf ___ --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
[Asterisk-Users] Monitor and sox mix quality
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 ___ --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] Voicemail
Download cvs head and look at
/usr/src/asterisk/configs/voicemail.conf.sample
All of the variables for email, page, etc are listed in the sample
files, it is more comprehensive than many of the other samples.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt
> Sent: Sunday, September 18, 2005 6:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Voicemail
>
> While we are on the subject.. how do you modify the TXT message that
> gets send to the 'pager'... Is that hard coded.. or can that be
> changed?No variables I change in voicemail.conf seem to change the
> from address, etc.
>
> On 9/18/05, Damon Estep <[EMAIL PROTECTED]> wrote:
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > [EMAIL PROTECTED]
> > Sent: Sunday, September 18, 2005 8:32 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] Voicemail
> >
> > When I receive a voicemail notify via e-mail I would like receive
not
> > the sender phone-number, but the sender name. Where can I configure
> > this and how? Is it possible to have some example?
> > Thank
> > Luca
> >
> > This will do it (in voicemail.conf) but I think the default does as
> > well. Are you actually getting caller ID name delivered when a call
> > comes it?
> >
> > emailbody=${VM_NAME} <${VM_MAILBOX}>\n\nYou have a new voicemail
message
> > from ${VM_CALLERID}. The message is ${VM_DUR} long and was left on
> > ${VM_DATE}.
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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Re: [Asterisk-Users] AstriCon 2006 Location
I think the best place would be Louisville, Kentucky. Its drivable from Cincinnati, Indianapolis, Tennesee, and a short plane ride from Chicago, Atlanta, NYC. Best of all, its near me :) On 9/18/05, Rich Adamson <[EMAIL PROTECTED]> wrote: > > > The best place for Astri Con 2006 would definatly be > > Omaha, Nebraska! ;) very central > > > > ...ah one could hope. > > or Lincoln, better facilities ;) > > > ___ > --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 > -- Tad Heckaman ___ --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] AstriCon 2006 Location
Heh, Why don't we do a tele-conference :) -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Reid Forrest Sent: Sunday, September 18, 2005 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] AstriCon 2006 Location > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Brian Capouch > Sent: Sunday, September 18, 2005 12:32 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] AstriCon 2006 Location > > Senad J wrote: > >>If you are looking for the maximum number of cheap flights from around > >>the world, and plenty of convention and room space, the answer is Las > >>Vegas :-) > > > > > > I would definitively agree! > > > > Yes, but what would one do there? > > One who doesn't gamble, drink, or carouse, that is. > > I am making my first trip to LV later this Fall, and I dread it. I > can't imagine what I'll be able to find to do when I'm not at the > conference. > Have you been there before? I spent a week there a few years ago and found plenty to do outside the casinos. There are plenty of good (clean) shows, exhibits, and attractions. Even if you don't like to shop (I don't) the Forum Shops at Caesar's Palace are great. If you're a Star Trek fan, the Star Trek Experience is a must see. R ___ --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] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
No, no, it's more SIMPLE than that. Try this:
[incoming]
exten => s,1,setcallerid("NAME")
exten => s,2,dial()
That's all (after couple of hours of investigation).
If call origin from SIP, i see NUMBER on my phone, if call origin from
PSTN, i see NAME on my phone.
- Original Message -
From: "Shaun Ewing" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: pon 19. sep 2005 2:04
Subject: Re: [Asterisk-Users] sometimes CIDNUM shows, sometimes
CIDNAME??? Why,why, why, why?
On 9/19/05, Goran Dj. <[EMAIL PROTECTED]> wrote:
> Why Asterisk showing (on SCCP and H323 phones) different CID related
to
> type of Incoming channel:
> If incoming channel is SIP, on phone is displayed CALLERIDNUM
> If incoming channel is ZAP, on phone is displayes CALLERIDNAME
>
> It vas very frustrating! I lost couple hours of my time to find that
my
> dialplan is not faulty, but asterisk is!
Have you considered the possibility that your SIP provider may not be
sending you the caller id name?
CNAM looksup do cost money, and it's probably the exception rather
than the norm to find a VoIP provider that will deliver it.
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Re: [Asterisk-Users] Voicemail
> While we are on the subject.. how do you modify the TXT message that
> gets send to the 'pager'... Is that hard coded.. or can that be
> changed?No variables I change in voicemail.conf seem to change the
> from address, etc.
Not sure this answers your question, but in voicemail.conf, I have:
; Skip the "[PBX]:" string from the message title
pbxskip=yes
; Change the From: string
;fromstring=The Asterisk PBX
fromstring=ABC VM
pagerfromstring=VM
emailsubject=VM
emailbody=For ${VM_MAILBOX} from ${VM_CALLERID} dur: ${VM_DUR} \n
At least the above reduces the number of characters to something
reasonable for cell phone text msgs.
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Re: [Asterisk-Users] Voicemail
While we are on the subject.. how do you modify the TXT message that
gets send to the 'pager'... Is that hard coded.. or can that be
changed?No variables I change in voicemail.conf seem to change the
from address, etc.
On 9/18/05, Damon Estep <[EMAIL PROTECTED]> wrote:
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Sunday, September 18, 2005 8:32 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Voicemail
>
> When I receive a voicemail notify via e-mail I would like receive not
> the sender phone-number, but the sender name. Where can I configure
> this and how? Is it possible to have some example?
> Thank
> Luca
>
> This will do it (in voicemail.conf) but I think the default does as
> well. Are you actually getting caller ID name delivered when a call
> comes it?
>
> emailbody=${VM_NAME} <${VM_MAILBOX}>\n\nYou have a new voicemail message
> from ${VM_CALLERID}. The message is ${VM_DUR} long and was left on
> ${VM_DATE}.
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>
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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [Asterisk-Users] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
On 9/19/05, Goran Dj. <[EMAIL PROTECTED]> wrote: > Why Asterisk showing (on SCCP and H323 phones) different CID related to > type of Incoming channel: > If incoming channel is SIP, on phone is displayed CALLERIDNUM > If incoming channel is ZAP, on phone is displayes CALLERIDNAME > > It vas very frustrating! I lost couple hours of my time to find that my > dialplan is not faulty, but asterisk is! Have you considered the possibility that your SIP provider may not be sending you the caller id name? CNAM looksup do cost money, and it's probably the exception rather than the norm to find a VoIP provider that will deliver it. ___ --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
[Asterisk-Users] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is! ___ --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] New User -> couple of dumb questions
> Just installed * 1.0.9 on a FC4 (full install)
>
> I am using 2 X100P clones
>
>
>
> I do not remember what all steps I took to get everything installed.
>
> Every time I reboot, I have to modprobe zaptel & modprobe wcfxo
>
> before asterisk will work. Did I miss a step somewhere?
Yes, the step isn't well documented. Do a 'make config' from within
the /usr/src/zaptel directory and the startup scripts should get
installed. The startup scripts will handle the modprobes, etc.
> also,
>
> I have Xlite on another machine in my network, I have not been able to
>
> find (or dont know enough to recognize) Can I setup in such a way that
>
> I can call from my Xlite to an extension in my * box?
Yes, the key to understanding this stuff is in how 'context' is used
within the various config files. (See example below.)
> I setup the extensions.conf to enable outbound calls (noise/echo are
> atrocious)
This is likely an issue of what you have in the zapata.conf file. For
the x100p's, you probably want these statements:
; This is the incoming 464-0718 pstn line on Zap/3
context=inbound-home
signalling=fxs_ks
echotraining=800
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=3.0
txgain=0.0
callgroup=2
immediate=no
callprogress=no
musiconhold=default
channel => 2
In the above, the key statements that impact echo will be the
statements that have the keyword 'echo' in them. The rxgain and
txgain will need to be adjusted to different values for "your"
pstn line loss. Start with the above values; if echo is to bad,
reduce the gain values. If echo is okay but volume is low, increase
the gain (more positive). You'll need to stop asterisk and restart
it for gain value changes to take effect (a reload won't cut it).
Without some better description of 'noise', I can't comment on it
very much. But, if the above statements don't address your problems,
then do a 'cat /proc/interrupts'. If you see the x100p device driver
sharing an interrupt, then you will need to address that. Hundreds
of postings over the last couple of years on how to address
interrupt sharing issue.
> but cannot seem to get the entry right to call an internal extension.
>
>
>
> Im sure both of these are quite simple, I have probably missed
> some little thing in my frustrated state.
Let's see if this helps a little
In your sip.conf file you define each sip phone (eg, xlite) something
like this:
[3001]
type=friend ; This device takes and makes calls
username=3001 ; Username on device
secret=mypassword ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
dtmfmode=rfc2833
mailbox=3001 ; Activate the MWI. use [EMAIL PROTECTED] to use a
; context other than default (for Voicemail).
When xlite registers with asterisk, all calls generated by the phone
will be analyzed/executed in the "from-sip" context located in
extensions.conf. For a small installation, you probably want all
of your sip.conf phone definitions to include 'context=from-sip'.
In the extensions.conf file, you _must_ have a matching context for
'from-sip', and whatever is located in that section is the only
extensions that can be called. So, you'll want something like this:
[from-sip]
include => local-extns
include => local-calls
Each of the above refers to yet another context, like:
[local-extns]
exten => 0,1,Dial(SIP/3000,15)
exten => 3000,1,Dial(SIP/3000,15,r)
exten => 3000,2,Voicemail(u3000)
exten => 3000,102,Voicemail(b3000)
exten => 3000,103,Hangup
exten => 3001,1,Dial(SIP/3001,15,r)
exten => 3001,2,Voicemail(u3001)
exten => 3001,102,Voicemail(b3001)
exten => 3001,103,Hangup
; This section for local numbers Only.
[local-calls]
exten => _21X,1,Dial(Zap/1/${EXTEN})
exten => _30X,1,Dial(Zap/1/${EXTEN})
exten => _32X,1,Dial(Zap/1/${EXTEN})
exten => _41X,1,Dial(Zap/1/${EXTEN})
exten => _420,1,Dial(Zap/1/${EXTEN})
exten => _421,1,Dial(Zap/1/${EXTEN})
exten => _423,1,Dial(Zap/1/${EXTEN})
Or, for the local-calls context, you could simply do:
exten => _9XXX,1,Dial(Zap/1/${EXTEN:1})
which forces you to dial a 9 before all local calls. The EXTEN:1
says to drop the first digit from the number dialed
(eg, 94231234 will be sent out the zap/1 interface, dropping
the leading 9).
The above are examples only to provide at least a little hint
on how contexts function. You might want to read more in the
asterisk-docs, wiki, and other references on this. There are
also a couple of O'Reilly books on asterisk.
Lots of references listed on www.asterisk.org at the bottom of
the 'support' page.
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[Asterisk-Users] Julien COURTEMANCHE/TELINTRANS/FR est absent(e).
Je serai absent(e) du 16/09/2005 au 01/10/2005. Je suis actuellement absent mais vous pouvez adresser vos mails à [EMAIL PROTECTED] qui vous répondra.___ --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
[Asterisk-Users] SpanDSP
Hey Everybody, Well, I knew better. I Competely broke my faxing in the current CVS (09/18/2005) of Asterisk. Needed to do a clean install to see if that would fix my caller id problem and thought that I had backed up all directories (Missed the modules directory). Modified the make file, downloaded and installed the pre3 release of spandsp. There was no rxfax.c or txfax.c, or updated patch file for pre3, so I used the pre2 ones. Everything compiles fine, but Asterisk complains about an unresolved symbol. Has anybody been successful with the current CVS and rxfax/txfax? My Makefile below: APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\ app_system.so app_echo.so app_record.so app_image.so app_url.so app_disa.so \ app_adsiprog.so app_getcpeid.so app_milliwatt.so \ app_zapateller.so app_setcallerid.so app_festival.so \ app_queue.so app_senddtmf.so app_parkandannounce.so app_striplsd.so \ app_setcidname.so app_lookupcidname.so app_substring.so app_macro.so \ app_authenticate.so app_softhangup.so app_lookupblacklist.so \ app_waitforring.so app_privacy.so app_db.so app_chanisavail.so \ app_enumlookup.so app_transfer.so app_setcidnum.so app_cdr.so \ app_hasnewvoicemail.so app_sayunixtime.so app_cut.so app_read.so \ app_setcdruserfield.so app_random.so app_ices.so app_eval.so \ app_nbscat.so app_sendtext.so app_exec.so app_sms.so \ app_groupcount.so app_txtcidname.so app_controlplayback.so \ app_talkdetect.so app_alarmreceiver.so app_userevent.so app_verbose.so \ app_test.so app_forkcdr.so app_math.so app_realtime.so \ app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \ app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \ app_dictate.so app_externalivr.so app_directed_pickup.so app_rxfax.so app_txfax.so ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/spandsp.h),) APPS+=app_rxfax.so app_txfax.so endif ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp.h),) APPS+=app_rxfax.so app_txfax.so endif app_rxfax.so: app_rxfax.o $(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff app_txfax.so: app_txfax.o $(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff ___ --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
[Asterisk-Users] Differ between "private" and "out of area"?
Is there any method to make difference between Hidden ("Private") and
unknown ("Out of area") incoming calls on ZAP/x101p? I want to block any
hidden call, and to allow unknow calls, but ZAP channel (X101P) always
delivering empty CALLERID=""<> in both cases.
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[Asterisk-Users] ChanSpy not loading
Hi All, I have installed ChanSpy but its not compiling gives me the following errorif anyone can help that will be really appreciated... [EMAIL PROTECTED] asterisk-1.0.9]# contrib/scripts/astxs -install -autoload apps/app_chanspy.c make: *** No rule to make target `apps_env'. Stop. -c apps/app_chanspy.c -o apps/app_chanspy.o Thanks Umair. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] Integration between Asterisk andSiemensHiCom150e over ISDN
Hi Simon, Point-to-Point (P2P) ist set if you want to use DID (In Germany called "Anlagenanschluss") Point-to-Multipoit (ptmp) is if you want to use the asterisk with singelpoint-entry (in Germany called Mehrgeräteanschluss) (also possible to dial extensions after this, but look at the maximum length for international calls!). NT or TE is another question, one (NT) gives the Synchronisation, the other (TE) gets it. But you dont need an NTBA, cause Hicom and HFC-Chipset can synchronise depending on configuration. Crossover cable is needed anyway! Regards Marco - Original Message - From: Simon D To: asterisk-users@lists.digium.com Sent: Sunday, September 18, 2005 10:57 PM Subject: RE: [Asterisk-Users] Integration between Asterisk andSiemensHiCom150e over ISDN Hi, Thanks for the advice on this.The Hicom can be set in Point-to-Point or Point-to-Multipoint mode (amongst others), I assume one is NT and the other is TE mode? If not, I cannot find any specific option to set NT and TE. Anyway, I have a crossover cable, however, the article here: http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom suggests that I need an NT1 (NTBA) as well - is this correct? Do I also need termination? I have set asterisk as bri_net_ptmp. Many thanks for your help people, Simon From: "Sander" <[EMAIL PROTECTED]>Reply-To: Asterisk Users Mailing List - Non-Commercial DiscussionTo: "'Asterisk Users Mailing List - Non-Commercial Discussion'"Subject: RE: [Asterisk-Users] Integration between Asterisk and SiemensHiCom150e over ISDNDate: Tue, 13 Sep 2005 17:04:57 +0200MIME-Version: 1.0Received: from lists.digium.com ([69.16.138.164]) by MC8-F11.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Tue, 13 Sep 2005 08:12:27 -0700Received: from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 146717C381;Tue, 13 Sep 2005 10:05:06 -0500 (CDT)Received: from psmtp.com (exprod5mx131.postini.com [64.18.0.45])by lists.digium.com (Postfix) with SMTP id 7E76C7C368for ;Tue, 13 Sep 2005 10:05:01 -0500 (CDT)Received: from source ([217.19.16.130]) by exprod5mx131.postini.com([64.18.4.10]) with SMTP; Tue, 13 Sep 2005 11:05:02 EDTReceived: (qmail 22951 invoked from network); 13 Sep 2005 15:05:00 -Received: from 84-245-20-233.dsl.cambrium.nl (HELO pc1) (84.245.20.233)by gollum.cambrium.nl with SMTP; 13 Sep 2005 15:05:00 ->Just setup the stls4 card to work in NT mode and connect the siemens pbx to>the asterisk with a crossover cable. Then you will be able to make calls to>from the hicom to the asterisk machine you do not need to have an nt box to>make the connection. With the nt box you can power an ISDN phone if the>phone needs power.-Oorspronkelijk bericht->Van: [EMAIL PROTECTED]>[mailto:[EMAIL PROTECTED] Namens Simon D>Verzonden: dinsdag 13 september 2005 15:14>Aan: asterisk-users@lists.digium.com>Onderwerp: [Asterisk-Users] Integration between Asterisk and Siemens>HiCom150e over ISDN>>Hi,>>I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and>wondered if anyone is able to offer any advice.>>In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk>box. e.g: HiCom user dials access code and can call Asterisk extension or>establish SIP call over Internet. Likewise, I'd like Asterisk to be able to>present a call to the Hicom, either Asterisk extension calling HiCom>extension, or an incoming Sipgate call presented to the HiCom, for example.>>My hardware:>>- Asterisk:>>I have an Asterisk box configured with 1x Sitecom DC105 PCI ISDN Card (HFC>chipset, TE/NT capable). [and 2x X100P Analogue FXOs, but that's not>relevant here]>>My understanding is that I should configure the ISDN card in NT mode and>power the bus with an NT1, or will a crossover cable in to the HiCom>suffice? Reference to NT1's and line power here:>http://isdn.jolly.de/download/v3.0/PBX4Linux-3.0.pdf>>>- Siemens HiCom 150e:>>I currently have a HiCom 150e switch with digital and analogue stations and>an analogue trunk card (TLA4).>I also have an STLS4 card. Initially, I thought this would be the answer to>my prayers but now am not so sure...>According to http://www.webco.com/siemens/interfaces.html, my STLS4 is>defined as the following:>>"Connects up to 4 ISDN S0 terminals (8 channels) for data equipment and>video conferencing applications to the OfficePoint or OfficeCom. Connected>devices must provide their own power. Some special wiring for the ISDN S0>device connection is required. Also requires connection of ISDN BRI trunks>(TMQ4) for network access.">>Documentation on ISDN BRI:>http:
RE: [Asterisk-Users] Integration between Asterisk and SiemensHiCom150e over ISDN
Hi, Thanks for the advice on this.The Hicom can be set in Point-to-Point or Point-to-Multipoint mode (amongst others), I assume one is NT and the other is TE mode? If not, I cannot find any specific option to set NT and TE. Anyway, I have a crossover cable, however, the article here: http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom suggests that I need an NT1 (NTBA) as well - is this correct? Do I also need termination? I have set asterisk as bri_net_ptmp. Many thanks for your help people, Simon From: "Sander" <[EMAIL PROTECTED]>Reply-To: Asterisk Users Mailing List - Non-Commercial DiscussionTo: "'Asterisk Users Mailing List - Non-Commercial Discussion'"Subject: RE: [Asterisk-Users] Integration between Asterisk and SiemensHiCom150e over ISDNDate: Tue, 13 Sep 2005 17:04:57 +0200MIME-Version: 1.0Received: from lists.digium.com ([69.16.138.164]) by MC8-F11.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Tue, 13 Sep 2005 08:12:27 -0700Received: from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 146717C381;Tue, 13 Sep 2005 10:05:06 -0500 (CDT)Received: from psmtp.com (exprod5mx131.postini.com [64.18.0.45])by lists.digium.com (Postfix) with SMTP id 7E76C7C368for ;Tue, 13 Sep 2005 10:05:01 -0500 (CDT)Received: from source ([217.19.16.130]) by exprod5mx131.postini.com([64.18.4.10]) with SMTP; Tue, 13 Sep 2005 11:05:02 EDTReceived: (qmail 22951 invoked from network); 13 Sep 2005 15:05:00 -Received: from 84-245-20-233.dsl.cambrium.nl (HELO pc1) (84.245.20.233)by gollum.cambrium.nl with SMTP; 13 Sep 2005 15:05:00 ->Just setup the stls4 card to work in NT mode and connect the siemens pbx to>the asterisk with a crossover cable. Then you will be able to make calls to>from the hicom to the asterisk machine you do not need to have an nt box to>make the connection. With the nt box you can power an ISDN phone if the>phone needs power.-Oorspronkelijk bericht->Van: [EMAIL PROTECTED]>[mailto:[EMAIL PROTECTED] Namens Simon D>Verzonden: dinsdag 13 september 2005 15:14>Aan: asterisk-users@lists.digium.com>Onderwerp: [Asterisk-Users] Integration between Asterisk and Siemens>HiCom150e over ISDN>>Hi,>>I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and>wondered if anyone is able to offer any advice.>>In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk>box. e.g: HiCom user dials access code and can call Asterisk extension or>establish SIP call over Internet. Likewise, I'd like Asterisk to be able to>present a call to the Hicom, either Asterisk extension calling HiCom>extension, or an incoming Sipgate call presented to the HiCom, for example.>>My hardware:>>- Asterisk:>>I have an Asterisk box configured with 1x Sitecom DC105 PCI ISDN Card (HFC>chipset, TE/NT capable). [and 2x X100P Analogue FXOs, but that's not>relevant here]>>My understanding is that I should configure the ISDN card in NT mode and>power the bus with an NT1, or will a crossover cable in to the HiCom>suffice? Reference to NT1's and line power here:>http://isdn.jolly.de/download/v3.0/PBX4Linux-3.0.pdf>>>- Siemens HiCom 150e:>>I currently have a HiCom 150e switch with digital and analogue stations and>an analogue trunk card (TLA4).>I also have an STLS4 card. Initially, I thought this would be the answer to>my prayers but now am not so sure...>According to http://www.webco.com/siemens/interfaces.html, my STLS4 is>defined as the following:>>"Connects up to 4 ISDN S0 terminals (8 channels) for data equipment and>video conferencing applications to the OfficePoint or OfficeCom. Connected>devices must provide their own power. Some special wiring for the ISDN S0>device connection is required. Also requires connection of ISDN BRI trunks>(TMQ4) for network access.">>Documentation on ISDN BRI:>http://web2.tac.siemenscom.com/pub/150e/Config/Note020.pdf>Documentation on ISDN S0 Device Install:>http://web2.tac.siemenscom.com/pub/150e/Config/Note009.pdf>>I can't figure out if the SLTS4 is the correct card for my requirements, or>do I need a TMQ4 *as well* or a TMQ4 *instead*?..>>Trawling through previous posts, I've found reference to the STLS4 here:>http://lists.digium.com/pipermail/asterisk-users/2005-January/081449.htmlIs anyone able to help?Many thanks in advance,>>Simon>England, UK>>>___>--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] AstriCon 2006 Location
John Novack wrote: Paul wrote: Yes, but what would one do there? One who doesn't gamble, drink, or carouse, that is. I am making my first trip to LV later this Fall, and I dread it. I can't imagine what I'll be able to find to do when I'm not at the conference. Rent a car and drive out to Hoover Dam. It is a magnificent sight I found LV boring otherwise, and FULL of drunk drivers. Seems DUI is accepted in LV, judging by the number of crumpled fenders I saw. For one who no longer drinks, isn't impressed by the lights and hotels, is terrified by an airport surrounded on 3 sides by mountains, and is bored by Wayne, Celene et al, Lincoln NE has better attractions. At least Lincoln has a Telephone Museum! John Novack I agree. And go to Cirque du Soleil, it is pretty spectacular too. ___ --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
[Asterisk-Users] New User -> couple of dumb questions
Just installed * 1.0.9 on a FC4 (full install) I am using 2 X100P clones I do not remember what all steps I took to get everything installed. Every time I reboot, I have to modprobe zaptel & modprobe wcfxo before asterisk will work. Did I miss a step somewhere? also, I have Xlite on another machine in my network, I have not been able to find (or don’t know enough to recognize) Can I setup in such a way that I can call from my Xlite to an extension in my * box? I setup the extensions.conf to enable outbound calls (noise/echo are atrocious) but cannot seem to get the entry right to call an internal extension. I’m sure both of these are quite simple, I have probably missed some little thing in my frustrated state. Thanks for any help. Shawn ___ --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
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 108
Hello All any body installed functions as below ?. i have 5 agent, logged in system. phone -> telco -> asterisk server --> queue (5 agents). when all agents busy. i want caller hear message from IVR and when any agent is available. ringback to agent then caller may be connected with that agent without press any DTMF. Any advice? __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
[Asterisk-Users] limiting calls per day based on amount of time
How to implement call limitation based on amount of time call per day. I've implemented the Dial "L" parameter but in addition I would like to limit an extension to certain amount of call time per day. -- #Joseph ___ --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] Asterisk Won't Process Call
Hi I have what seems like a similar problem. In the last few days, I stopped receiving calls from my Broadvoice number into Asterisk. The account activity page at Broadvoice does not show the calls as either missed or incoming. Unfortunately, Broadvoice's line is "we don't support Asterisk". :-/ Can you see the calls you're attempting to place in your Broadvoice activity page? Klaus --- Michael Stearne <[EMAIL PROTECTED]> wrote: > We have a basic application that runs a SIP channel to pick up a call > and process it. We are using Broadvoice and it's been working great. > We recently rebooted the machine and now when a call comes in Asterisk > picks up the call but does not process it. Asterisk seems to send the > call back to Broadvoice. Nothing at all has been changed in the > configuration to warrant this. Below is the output of sip debug. Any > help would be a life saver! > > <-- SIP read from 147.135.20.128:5060: > INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 > Call-ID: [EMAIL PROTECTED] > CSeq: 1 INVITE > From: "Brooklyn NY";tag=ikmn > To: "Michael Stearne" > Via: SIP/2.0/UDP 147.135.20.128:5060 > Contact: > Supported: 100rel > RPID-Privacy: party=calling;id-type=subscriber;privacy=off > Remote-Party-ID: > ;screen=yes;party=calling;privacy=off > Content-Length: 273 > Content-Type: application/sdp > > v=0 > o=2475103479 10 10 IN IP4 147.135.20.247 > s=- > c=IN IP4 147.135.20.250 > t=0 0 > m=audio 10690 RTP/AVP 0 8 2 18 96 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:96 iLBC/8000 > a=rtpmap:101 telephone-event/8000 > > --- (12 headers 12 lines)--- > Using INVITE request as basis request - [EMAIL PROTECTED] > Sending to 147.135.20.128 : 5060 (non-NAT) > Found peer 'sip2.broadvoice.com' > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 2 > Found RTP audio format 18 > Found RTP audio format 96 > Found RTP audio format 101 > Peer audio RTP is at port 147.135.20.250:10690 > Found description format PCMU > Found description format PCMA > Found description format G726-32 > Found description format G729 > Found description format iLBC > Found description format telephone-event > Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51c > (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc > (ulaw|alaw)Non-codec capabilities: us - 0x1 (telephone-event), peer - > 0x1 (telephone-event), combined - 0x1 (telephone-event) > Looking for 6092991xxx in from-broadvoice > Reliably Transmitting (no NAT) to 147.135.20.128:5060: > SIP/2.0 404 Not Found > Via: SIP/2.0/UDP 147.135.20.128:5060 > From: "Brooklyn NY";tag=ikmn > To: "Michael Stearne";tag=as38d08027 > Call-ID: [EMAIL PROTECTED] > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: > Content-Length: 0 > > > --- > > <-- SIP read from 147.135.20.128:5060: > ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 > Call-ID: [EMAIL PROTECTED] > CSeq: 1 ACK > From: "Brooklyn NY";tag=ikmn > To: "Michael Stearne";tag=as38d08027 > Via: SIP/2.0/UDP 147.135.20.128:5060;received=209.3.28.xx > Content-Length:0 > > > --- (7 headers 0 lines)--- > Destroying call '[EMAIL PROTECTED]' > ___ > --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] AstriCon 2006 Location
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Brian Capouch > Sent: Sunday, September 18, 2005 12:32 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] AstriCon 2006 Location > > Senad J wrote: > >>If you are looking for the maximum number of cheap flights from around > >>the world, and plenty of convention and room space, the answer is Las > >>Vegas :-) > > > > > > I would definitively agree! > > > > Yes, but what would one do there? > > One who doesn't gamble, drink, or carouse, that is. > > I am making my first trip to LV later this Fall, and I dread it. I > can't imagine what I'll be able to find to do when I'm not at the > conference. > Have you been there before? I spent a week there a few years ago and found plenty to do outside the casinos. There are plenty of good (clean) shows, exhibits, and attractions. Even if you don't like to shop (I don't) the Forum Shops at Caesar's Palace are great. If you're a Star Trek fan, the Star Trek Experience is a must see. R ___ --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
[Asterisk-Users] PRI to PRI passthrough with DID intact
Thanks for the info. I now have the phone systems switched.
The DID is working for both phone systems and the CID also.
The only problem I have now is that CDR records from the panasonic to
outside have a destination of "h" instead of the dialed number.
I originally started with an [EMAIL PROTECTED] setup, then later disabled the
mySQL management because it was too limiting.
I believe that my problem is in the [macro-dialout-trunk] that comes with
the [EMAIL PROTECTED] install.
The log shows the dialing and dialed number, then when it bridges the call
to the other PRI, the dialed number is an "h".
refs:
extensions.conf:
[panasonic]
include => ext-local
include => outrt-002-fwd
exten => _.,1,Macro(dialout-trunk,1,${EXTEN},)
exten => _.,2,Macro(outisbusy) ; No available circuits
[macro-dialout-trunk]
exten => s,1,GotoIf($[foo${ARG3} = foo]?3:2)) ; arg3 is pattern password
exten => s,2,Authenticate(${ARG3})
exten => s,3,Macro(record-enable,${CALLERIDNUM},OUT)
exten => s,4,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?7) ;check for CID
override for exten
exten => s,5,SetCallerID(${ECID${CALLERIDNUM}})
exten => s,6,Goto(9)
exten => s,7,GotoIf($[foo${OUTCID_${ARG1}} = foo]?9) ;check for CID
override for trunk
exten => s,8,SetCallerID(${OUTCID_${ARG1}})
exten => s,9,SetGroup(OUT_${ARG1})
exten => s,10,CheckGroup(${OUTMAXCHANS_${ARG1}})
exten => s,11,SetVar(DIAL_NUMBER=${ARG2})
exten => s,12,SetVar(DIAL_TRUNK=${ARG1})
exten => s,13,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial
string for this trunk
exten => s,14,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ; OUTNUM is
the final dial number
exten => s,15,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are prefixed with
"AMP:"
exten => s,16,GotoIf($[${custom} = AMP]?19)
exten => s,17,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial
exten => s,18,Goto(s-${DIALSTATUS},1)
exten => s,19,Cut(pre_num=OUT_${ARG1},$,1)
exten => s,20,Cut(the_num=OUT_${ARG1},$,2) ; this is where we expect to
find string OUTNUM
exten => s,21,Cut(post_num=OUT_${ARG1},$,3)
exten => s,22,GotoIf($[${the_num} = OUTNUM]?23:24) ; if we didn't find
"OUTNUM", then skip to Dial
exten => s,23,SetVar(the_num=${OUTNUM}) ; replace "OUTNUM" with the actual
number to dial
exten => s,24,Dial(${pre_num:4}${the_num}${post_num})
exten => s,25,Goto(s-${DIALSTATUS},1)
exten => s,111,Noop(max channels used up)
exten => s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten => s-BUSY,2,Busy()
exten => s-BUSY,3,Wait(60)
exten => s-BUSY,4,NoOp()
exten => _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})
log pieces:
Sep 18 11:45:11 VERBOSE[2223]: -- Executing Macro("Zap/47-1",
"dialout-trunk|1|181034xXxXx|") in new stack
Sep 18 11:45:11 DEBUG[2223]: Expression is '1'
Sep 18 11:45:11 VERBOSE[2223]: -- Executing GotoIf("Zap/47-1", "1?3:2)") in
new stack
Sep 18 11:45:11 VERBOSE[2223]: -- Goto (macro-dialout-trunk,s,3)
Sep 18 11:45:11 VERBOSE[2223]: -- Executing Macro("Zap/47-1",
"record-enable|24883xXxXx|OUT") in new stack
Sep 18 11:45:11 VERBOSE[2223]: -- Executing GotoIf("Zap/47-1", "0 > 0?2:4")
in new stack
Sep 18 11:45:11 VERBOSE[2223]: -- Goto (macro-record-enable,s,4)
Sep 18 11:45:11 DEBUG[2223]: Expression is '1'
Sep 18 11:45:11 VERBOSE[2223]: -- Executing GotoIf("Zap/47-1", "1?5:8") in
new stack
Sep 18 11:45:11 VERBOSE[2223]: -- Accepting call from '24883xXxXx' to
'181034xXxXx' on channel 0/23, span 2
Sep 18 11:45:11 DEBUG[2223]: Enabled echo cancellation on channel 47
Sep 18 11:45:11 VERBOSE[2223]: -- Goto (macro-record-enable,s,5)
Sep 18 11:45:11 VERBOSE[2223]: -- Executing DBget("Zap/47-1",
"RecEnable=RECORD-OUT/24883xXxXx") in new stack
Sep 18 11:45:11 VERBOSE[2223]: -- DBget: varname=RecEnable,
family=RECORD-OUT, key=24883xXxXx
Sep 18 11:45:11 DEBUG[2223]: Unable to find key '24883xXxXx' in family
'RECORD-OUT'
Sep 18 11:45:11 VERBOSE[2223]: -- DBget: Value not found in database.
Sep 18 11:45:11 VERBOSE[2223]: -- Executing SetVar("Zap/47-1",
"CALLFILENAME=OUT24883xXxXx-20050918-114511-1127058311.261") in new stack
Sep 18 11:45:11 VERBOSE[2223]: -- Executing Goto("Zap/47-1", "s|14") in new
stack
Sep 18 11:45:11 VERBOSE[2223]: -- Goto (macro-record-enable,s,14)
Sep 18 11:45:11 DEBUG[2223]: Expression is '0'
Sep 18 11:45:11 VERBOSE[2223]: -- Executing GotoIf("Zap/47-1", "0?15:99") in
new stack
Sep 18 11:45:11 VERBOSE[2223]: -- Goto (macro-record-enable,s,99)
Sep 18 11:45:11 VERBOSE[2223]: -- Executing NoOp("Zap/47-1", "NO RECORDING
NEEDED") in new stack
Sep 18 11:45:11 DEBUG[2223]: Expression is '1'
Sep 18 11:45:11 VERBOSE[2223]: --
Re: [Asterisk-Users] Two POTS in, but only want one out?
> Hi! I've got two POTS lines coming in to my * box, but I only want the
> primary of the two lines available for outbound dialing. I can't quite
> figure out how to make that happen. Suggestions?
How about something like this?
exten => _9XXX,1,Dial(Zap/2/${EXTEN})
which dials out on the second zap channel/port.
___
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[Asterisk-Users] Two POTS in, but only want one out?
Hi! I've got two POTS lines coming in to my * box, but I only want the primary of the two lines available for outbound dialing. I can't quite figure out how to make that happen. Suggestions? Thanks, -Ken D'Ambrosio ___ --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] AstriCon 2006 Location
Senad J wrote: [EMAIL PROTECTED] wrote: Brian Capouch wrote: Senad J wrote: If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) I would definitively agree! Yes, but what would one do there? One who doesn't gamble, drink, or carouse, that is. I am making my first trip to LV later this Fall, and I dread it. I can't imagine what I'll be able to find to do when I'm not at the conference. Man... you are literally talking about entertainment capitol of the world. Look here to start with: http://www.vegas.com/attractions/?f=submore&t=mattrt PSS... Bring PLENTY of money though :) Precisely. B. ___ --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] AstriCon 2006 Location
Paul wrote: Yes, but what would one do there? One who doesn't gamble, drink, or carouse, that is. I am making my first trip to LV later this Fall, and I dread it. I can't imagine what I'll be able to find to do when I'm not at the conference. Rent a car and drive out to Hoover Dam. It is a magnificent sight I found LV boring otherwise, and FULL of drunk drivers. Seems DUI is accepted in LV, judging by the number of crumpled fenders I saw. For one who no longer drinks, isn't impressed by the lights and hotels, is terrified by an airport surrounded on 3 sides by mountains, and is bored by Wayne, Celene et al, Lincoln NE has better attractions. At least Lincoln has a Telephone Museum! John Novack ___ --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] AstriCon 2006 Location
[EMAIL PROTECTED] wrote: > Brian Capouch wrote: > >> Senad J wrote: >> If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) >>> >>> >>> >>> I would definitively agree! >>> >> >> Yes, but what would one do there? >> >> One who doesn't gamble, drink, or carouse, that is. >> >> I am making my first trip to LV later this Fall, and I dread it. I >> can't imagine what I'll be able to find to do when I'm not at the >> conference. Man... you are literally talking about entertainment capitol of the world. Look here to start with: http://www.vegas.com/attractions/?f=submore&t=mattrt PSS... Bring PLENTY of money though :) ___ --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] AstriCon 2006 Location
Brian Capouch wrote: Senad J wrote: If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) I would definitively agree! Yes, but what would one do there? One who doesn't gamble, drink, or carouse, that is. I am making my first trip to LV later this Fall, and I dread it. I can't imagine what I'll be able to find to do when I'm not at the conference. They have lots of wedding chapels. Get married a few times a day. ___ --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] AstriCon 2006 Location
Senad J wrote: If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) I would definitively agree! Yes, but what would one do there? One who doesn't gamble, drink, or carouse, that is. I am making my first trip to LV later this Fall, and I dread it. I can't imagine what I'll be able to find to do when I'm not at the conference. B. ___ --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] AstriCon 2006 Location
> The best place for Astri Con 2006 would definatly be > Omaha, Nebraska! ;) very central > > ...ah one could hope. or Lincoln, better facilities ;) ___ --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] Asterisk Won't Process Call
On Sun, 18 Sep 2005, Michael Stearne wrote: > Looking for 6092991xxx in from-broadvoice > Reliably Transmitting (no NAT) to 147.135.20.128:5060: > SIP/2.0 404 Not Found So - go to your Asterisk CLI> and type "show dialplan from-broadvoice". Examine the list of extensions shown to figure out why 6092991xxx doesn't match anything. The dialplan contents come from extensions.conf. Steve ___ --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] AstriCon 2006 Location
> > If you are looking for the maximum number of cheap flights from around > > the world, and plenty of convention and room space, the answer is Las > > Vegas :-) > > quoted for truth. > > $79 flights from Dallas. Having just returned from LV yesterday, the cost of flights is very inexpensive, but all other costs in LV have tripled in the last couple of years. VERY expensive now compared to two years ago, and for most items, more expensive then many other cities mentioned on this list. ___ --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
[Asterisk-Users] Asterisk Won't Process Call
We have a basic application that runs a SIP channel to pick up a call and process it. We are using Broadvoice and it's been working great. We recently rebooted the machine and now when a call comes in Asterisk picks up the call but does not process it. Asterisk seems to send the call back to Broadvoice. Nothing at all has been changed in the configuration to warrant this. Below is the output of sip debug. Any help would be a life saver! <-- SIP read from 147.135.20.128:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: "Brooklyn NY";tag=ikmn To: "Michael Stearne" Via: SIP/2.0/UDP 147.135.20.128:5060 Contact: Supported: 100rel RPID-Privacy: party=calling;id-type=subscriber;privacy=off Remote-Party-ID: ;screen=yes;party=calling;privacy=off Content-Length: 273 Content-Type: application/sdp v=0 o=2475103479 10 10 IN IP4 147.135.20.247 s=- c=IN IP4 147.135.20.250 t=0 0 m=audio 10690 RTP/AVP 0 8 2 18 96 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:101 telephone-event/8000 --- (12 headers 12 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.135.20.128 : 5060 (non-NAT) Found peer 'sip2.broadvoice.com' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 101 Peer audio RTP is at port 147.135.20.250:10690 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 6092991xxx in from-broadvoice Reliably Transmitting (no NAT) to 147.135.20.128:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 147.135.20.128:5060 From: "Brooklyn NY";tag=ikmn To: "Michael Stearne";tag=as38d08027 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- <-- SIP read from 147.135.20.128:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: "Brooklyn NY";tag=ikmn To: "Michael Stearne";tag=as38d08027 Via: SIP/2.0/UDP 147.135.20.128:5060;received=209.3.28.xx Content-Length:0 --- (7 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' ___ --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] Who is going to AstriCon (The AsteriskConference)?
canuck15 wrote: -Original Message- From: Steven Sokol [mailto:[EMAIL PROTECTED] Sent: Saturday, September 17, 2005 10:31 AM To: Kurth Bemis Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Who is going to AstriCon (The AsteriskConference)? We've talked about doing several smaller events, but doing more than one full-fledged AstriCon per year in the US is cost and time prohibitive. The conference will have something like 50 speakers, 25 - 30 exhibitors, 15 staff members. It takes about 6 months of planning to arrange something like this. 25-30 exhibitors? Do you have a list of who/what they are? I think the twice a year in various cities format is fine. How about something in Canada next year eh? Yukon needs the business, eh? ___ --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] AstriCon 2006 Location
Kevin P. Fleming wrote: Steven Sokol wrote: Actually, my wife's company holds several events each year in Orlando and the turn-out is always great -- people love getting a bit of vacation with their business travel. (An Universal's Islands of Adventure is just plain awesome!). If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) They treat me well there and I am no high roller. I never had a bad trip to Las Vegas. ___ --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] Who is going to AstriCon (The AsteriskConference)?
> -Original Message- > From: Steven Sokol [mailto:[EMAIL PROTECTED] > Sent: Saturday, September 17, 2005 10:31 AM > To: Kurth Bemis > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Who is going to AstriCon (The > AsteriskConference)? > > We've talked about doing several smaller events, but doing > more than one full-fledged AstriCon per year in the US is > cost and time prohibitive. The conference will have > something like 50 speakers, 25 > - 30 exhibitors, 15 staff members. It takes about 6 months > of planning to arrange something like this. > 25-30 exhibitors? Do you have a list of who/what they are? I think the twice a year in various cities format is fine. How about something in Canada next year eh? ___ --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] AstriCon 2006 Location
> > Atlanta is hub for Delta and Airtran > > > > Dallas is hub for American > > > > Chicago is hub for ATA > > > > All good central locations with cheap non stop flights. > > Atlanta is central for who? With all of the tornados, hurricanes, etc. > I would definately vote no for there. Dallas and Chicago are both good. As is Denver. If you really want to get down to earth, how about a resort in the mountains? (Denver airport to anywhere requires some thought on transportation; adding a few miles to a resort in the mountains doesn't amount to much. ;) ___ --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] AstriCon 2006 Location
4th'ed! No lack of things to do, cheap hotels (off the strip...or on depending on your definition of cheap), cheap flights from just about everywhere... On 9/18/05, asterisk <[EMAIL PROTECTED]> wrote: > Viva Las Vegas !!! > > Neil > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Senad J > Sent: 18 September 2005 14:45 > To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users > Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] AstriCon 2006 Location > > > If you are looking for the maximum number of cheap flights from around > > the world, and plenty of convention and room space, the answer is Las > > Vegas :-) > > I would definitively agree! > > > Senad ___ --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
[Asterisk-Users] voicemail context. macro, and directory
I use a macro to dial most of my extensions and in the macro the
variable ${macro_exten} is used to send calls to the right voicemail
box.
I also use realtime MySQL for SIP, voicemail, and extensions.
When I put a voicemail box in a voicemail context other than default the
method of getting the caller to voicemail fails as it does not include
the context.
Is there a built in channel variable that I could read to get the
'mailbox' field from realtime sip? If so I could then store the mailbox
value with the context attached, like [EMAIL PROTECTED]
Not sure how MySQL will handle the @, but I assume that since it is in a
text string it should be fine.
If there is no existing variable I guess I could read the entire row
from the sip table to the macro with realtime()
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Re: [Asterisk-Users] AstriCon 2006 Location
If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) quoted for truth. $79 flights from Dallas. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstriCon 2006 Location
Actually, we have thought about that too. The problem is scheduling so as not to conflict with any of the other shows that our major sponsors are doing (remember, they help keep the cost of the conference down). Within roughly week of AstriCon on one side or ther other we have the CompTel/ASCENT show, WISPCON, ISPCON, Telecom '05, Phreaknic, ITU Americas, and TMC's Internet Telephony. VON is only 3 weeks prior. In general the fall is chock-full-o telephony shows. I'm skipping Comptel this year to go to Astricon. I tried to get out to Orlando for just one day but the flights from LAX to MCO are ridiculously long time wise, and they all landed outside of exhibition hours. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, September 18, 2005 8:32 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemail
When I receive a voicemail notify via e-mail I would like receive not
the sender phone-number, but the sender name. Where can I configure
this and how? Is it possible to have some example?
Thank
Luca
This will do it (in voicemail.conf) but I think the default does as
well. Are you actually getting caller ID name delivered when a call
comes it?
emailbody=${VM_NAME} <${VM_MAILBOX}>\n\nYou have a new voicemail message
from ${VM_CALLERID}. The message is ${VM_DUR} long and was left on
${VM_DATE}.
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[Asterisk-Users] Voicemail
When I receive a voicemail notify via e-mail I would like receive not the sender phone-number, but the sender name. Where can I configure this and how? Is it possible to have some example? Thank Luca ___ --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 : RE : [Asterisk-Users] Asterisk realtime beta
Ok I use Libiobc instead of unixodbc. But Now asterisk doesn't find the database... Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Olivier Taylor Envoyé : dimanche 18 septembre 2005 12:58 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE : [Asterisk-Users] Asterisk realtime beta Well, in fact I have compilation error, Unixodbc is installed. But I get : gcc -shared -Xlinker -x -o res_odbc.so res_odbc.o -lodbc /usr/bin/ld: cannot find -lodbc gmake[1]: *** [res_odbc.so] Error 1 gmake[1]: Leaving directory `/usr/local/src/asterisk-1.2.0-beta1/res' gmake: *** [subdirs] Error 1 Any help is welcome Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Urban Envoyé : samedi 17 septembre 2005 19:28 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Asterisk realtime beta Olivier Taylor wrote: >Does anybody has intalled it on freebsd with unixodbc or libiodbc and >have it working? > >Regards, > >Olivier > >___ >--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 > > I have installed unixodbc for realtime on freebsd 5.4 and it works fine. I currently not use it due to some limitations in * realtime . cheers urban ___ --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 ___ --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
[Asterisk-Users] (no subject)
When I receive voicemail notify via e-mail I would like receive not the phone-number, but the sender name. Where can I configure this and how? Is it possible to have some example? Thank Luca ___ --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] TFTP and DHCP...
On Sunday 18 Sep 2005 15:15, Francois Meehan wrote: > Hi all, > > I have bought an Aastra 480i phone. > > In order to configure the phone for using a TFTP server, I had to enter > the TFTP ip address directly in the phone, and then reboot the phone > again. > > Is it possible to configure a DHCP server so it sends a TFTP server > coordinate for the phone to use? Yes, man dhcp-options, but will the phone use it? ___ --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
[Asterisk-Users] TFTP and DHCP...
Hi all, I have bought an Aastra 480i phone. In order to configure the phone for using a TFTP server, I had to enter the TFTP ip address directly in the phone, and then reboot the phone again. Is it possible to configure a DHCP server so it sends a TFTP server coordinate for the phone to use? Best regards, Francois Random Thought: --- A bottle of wine begs to be shared; I have never met a miserly wine lover. - Clifton Fadiman, 1904 - 1999 Random Thought: --- A bottle of wine begs to be shared; I have never met a miserly wine lover. - Clifton Fadiman, 1904 - 1999 ___ --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] AstriCon 2006 Location
Viva Las Vegas !!! Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad J Sent: 18 September 2005 14:45 To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] AstriCon 2006 Location > If you are looking for the maximum number of cheap flights from around > the world, and plenty of convention and room space, the answer is Las > Vegas :-) I would definitively agree! Senad ___ --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] AstriCon 2006 Location
> If you are looking for the maximum number of cheap flights from around > the world, and plenty of convention and room space, the answer is Las > Vegas :-) I would definitively agree! Senad ___ --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] AstriCon 2006 Location
Steven Sokol wrote: Actually, my wife's company holds several events each year in Orlando and the turn-out is always great -- people love getting a bit of vacation with their business travel. (An Universal's Islands of Adventure is just plain awesome!). If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) ___ --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] moh - turn off randomization?
Damon Estep wrote: That is what I thought I read somewhere, but it is not so. I will check, but I THINK that * reads the file names left to right top to bottom and my FC4 box lists them with an ls top to bottom left to right! Oh you're right... they are not sorted. My fault, I forgot about the way that readdir() works. The other poster is right, they are returned in the order they are stored in the directory. I suppose an enhancement to sort them at load time would be useful. ___ --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] g729 passthrough?
Better is really not to use G723... ;) Can't you use others alternatives? Regards, -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Asterisk guy Enviada: segunda-feira, 25 de Abril de 2005 3:17 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] g729 passthrough? i am trying to get G723 passthrough get the same error. how to configure passthrough for g723/g729 ? On 4/24/05, Brian Capouch <[EMAIL PROTECTED]> wrote: > jltaylor wrote: > > > ;;; > > > > Brian, > > > > Add to the [general] section in sip.conf the following: > > > > disallow=all > > allow=g729 > > allow=ulaw > > allow=alaw > > > > > > For some reason Asterisk will not pass audio through itself without trying > > to transcode unless you have this in your config. > > Don't ask me why it will not work with allow=g729 under the individual peer. > > This has to go in the [general] section. > > > > Still no joy. Added the allow=g729 to general, too, and I still get the > same errors. > > Thanks anyways. > > B. > ___ > 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 ___ --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] moh - turn off randomization?
On Saturday 17 Sep 2005 23:20, Kevin P. Fleming wrote: > Damon Estep wrote: > > I do not have the "r" option in the MOH class, but the files are played > > in an order I can figure out, they do not appear to be random either, > > same pattern repeats. > > Oh come on, its obvious :-) > > Have you figured it out yet? > > Yet? > > Now? > > OK... I'll tell you. > > See the order you get when you do 'ls' in that directory? It's that > order :-) If you want to force the files to play in a particular order, > add a numeric prefix to each one. I think they are in what ever order they are stored in the directory listing, which can be found with "ls -f" or "ls -fl". I cannot see anywhere in * of any sorting of filenames. B ___ --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] DID from an analog phone
> how can i dial DID to asterisk from analog telephone? Zapata does not report a DID for the incoming calls - as I mentioned in a recent post about seperating incoming calls on a TDM02B (see the archive). Effectively you will need to point each port to the appropriate dialplan context in your zapata.conf. Hatton ___ --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] AstriCon 2006 Location
Orlando would be great in November, and you can get cheap and frequent flights there from just about anywhere in the country. But it's about a far from a "central" location for people in the US as you can get. Although South American attendees can get direct flights to Orlando so it is easier for them. MATT---On 9/17/05, Steven Sokol <[EMAIL PROTECTED]> wrote: On 9/17/05, Reid Forrest <[EMAIL PROTECTED]> wrote:>> >> > Matthew Simpson wrote:> > > Atlanta is hub for Delta and Airtran > > >> > > Dallas is hub for American> > >> > > Chicago is hub for ATA> > >> > > All good central locations with cheap non stop flights.> > > > Atlanta is central for who? With all of the tornados, hurricanes, etc.> > I would definately vote no for there. Dallas and Chicago are both good.> >>> How about Orlando? Plenty to do here, and no lack of flights! My only hesitation would be the chance of hurricanes this time ofyear. What with the events of last year and now Katrina... Perhapsif we held it in November when the chance of storms drops off.Actually, my wife's company holds several events each year in Orlando and the turn-out is always great -- people love getting a bit ofvacation with their business travel. (An Universal's Islands ofAdventure is just plain awesome!).-S--Steven SokolCEO/Manager Sokol & Associates, LLCAsk Me About AstriCon 2005!http://www.astricon.net/___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
[Asterisk-Users] DID from an analog phone
hi all, I wanted to call my asterisk on the Zapchannel with mainnumber+DID number: it's ok for calls from handy and from sip i get the right extension(DID) on phone, but when i call from an analog telephone the DID number is not mentioned by asterisk. in my zapata.conf: overlapdial = yes immediate = no That should take care of the waiting for DID numbers, which it doesn't. Asterisk how can i dial DID to asterisk from analog telephone? crs ___ --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] Asterisk realtime beta
Well, in fact I have compilation error, Unixodbc is installed. But I get : gcc -shared -Xlinker -x -o res_odbc.so res_odbc.o -lodbc /usr/bin/ld: cannot find -lodbc gmake[1]: *** [res_odbc.so] Error 1 gmake[1]: Leaving directory `/usr/local/src/asterisk-1.2.0-beta1/res' gmake: *** [subdirs] Error 1 Any help is welcome Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Urban Envoyé : samedi 17 septembre 2005 19:28 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Asterisk realtime beta Olivier Taylor wrote: >Does anybody has intalled it on freebsd with unixodbc or libiodbc and >have it working? > >Regards, > >Olivier > >___ >--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 > > I have installed unixodbc for realtime on freebsd 5.4 and it works fine. I currently not use it due to some limitations in * realtime . cheers urban ___ --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
