Re: [asterisk-users] Asterisk routine maintenance activities
On Wed, Apr 22, 2009 at 11:05:38AM +0530, Kurian Thayil wrote: > On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote: > > Daily Asterisk restart > > Do you think its mandatory in production env? No. > > > > > Daily log rotation A simple logrotate file takes care of that. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue() Ignore Hangup Request
I saw a few posts of this problem before I could not figure out the
reason I am getting it.
I am running RHEL 5, Asterisk 1.4.21.2, zaptel 1.4.11 and libpri 1.4.4
Basically, if I dial into a queue and hang up the phone, Asterisk did
not detect the hangup request and Asterisk will only hang up when the
timer expires.
There is no such problem if I do not use Queue().
Any thoughts?
Here is my zaptel.conf
loadzone=au
defaultzone=au
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-21
unused=22-31
dchan=16
span=2,0,0,esf,b8zs
fxols=32-55
Here is the log:
-- Accepting call from '28835666' to '98857843' on channel 0/2, span
1
-- Executing [98857...@incoming:1] Answer("Zap/2-1", "") in new
stack
-- Executing [98857...@incoming:2] Goto("Zap/2-1",
"ael-queue-office-incoming|s|1") in new stack
-- Goto (ael-queue-office-incoming,s,1)
-- Executing [...@ael-queue-office-incoming:1] Answer("Zap/2-1", "")
in new stack
-- Executing [...@ael-queue-office-incoming:2] Set("Zap/2-1",
"quv_que_nam=office") in new stack
-- Executing [...@ael-queue-office-incoming:3] Wait("Zap/2-1", "2") in
new stack
-- Executing [...@ael-queue-office-incoming:4] Set("Zap/2-1",
"cdv_sts_dbd=2") in new stack
-- Executing [...@ael-queue-office-incoming:5] Set("Zap/2-1",
"~~EXTEN~~=s") in new stack
-- Executing [...@ael-queue-office-incoming:6] Goto("Zap/2-1",
"sw-104-2|10") in new stack
-- Goto (ael-queue-office-incoming,sw-104-2,10)
-- Executing [sw-10...@ael-queue-office-incoming:10] Set("Zap/2-1",
"nsv_sts_dbd=2") in new stack
-- Executing [sw-10...@ael-queue-office-incoming:11] Set("Zap/2-1",
"nsv_div_exs=0") in new stack
-- Executing [sw-10...@ael-queue-office-incoming:12] Set("Zap/2-1",
"~~EXTEN~~=sw-104-2") in new stack
-- Executing [sw-10...@ael-queue-office-incoming:13] Goto("Zap/2-1",
"sw-106-2|10") in new stack
-- Goto (ael-queue-office-incoming,sw-106-2,10)
-- Executing [sw-10...@ael-queue-office-incoming:10] Goto("Zap/2-1",
"ael-queue-office-au|s|1") in new stack
-- Goto (ael-queue-office-au,s,1)
-- Executing [...@ael-queue-office-au:1] SetMusicOnHold("Zap/2-1",
"cpwr") in new stack
-- Executing [...@ael-queue-office-au:2] GotoIf("Zap/2-1", "1?3:5") in
new stack
-- Goto (ael-queue-office-au,s,3)
-- Executing [...@ael-queue-office-au:3] Queue("Zap/2-1", "office|r")
in new stack
-- SIP/343-098f5268 is ringing
-- Channel 0/2, span 1 got hangup, cause 102
== Spawn extension (ael-queue-office-au, s, 3) exited non-zero on
'Zap/2-1'
-- Hungup 'Zap/2-1'
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Re: [asterisk-users] run dialplan when open line
2009/4/22 michel freiha > Hi all, > Does asterisk support the following scenario? I need when a customer who > own an endpoint registered on asterisk open the line, the asterisk will run > a specific AGI script inside the endpoint context? > You mean when they pick up the phone it'll automatically run the AGI? If so, easy to do with dahdi channels using immediate=yes, the s extension in that channels context would be executed on the phone going offhook, with other channel types it depends on whether the endport device supports some way of really automatically opening a channel... Most SIP devices will generate their own dial tone and at a certain point, based on their local dialplan, send the dialled string to asterisk to open a channel, if nothing is dialled, they wouldn't contact the server... That said, their may be some devices that would allow configuration such that they will automatically dial a number when going offhook... d ___ -- 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] Asterisk routine maintenance activities
> > Daily Asterisk restart > > Do you think its mandatory in production env? > It could be a pre-1.6 advice but I still stick to it. I did it to all my production Asterisk servers. ___ -- 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] Faxing and TIFF files
2009/4/22 Michael > I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing. > > Does anyone know of a way, either while producing the file, or after, to > tell > how many pages have been produced? (without manually viewing the file) > tiffinfo? then count the number of data blocks... d ___ -- 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] Asterisk routine maintenance activities
On Wed, 22 Apr 2009, James Mutuku wrote: > I know this might be test book question or one best suited for google > but I will take the risk of asking. Here I go. What common routine > maintenance tasks do you run on your asterisk box? None. I configure Asterisk to log everything to syslog on a central loghost. All of the AGIs connect to a separate (MySQL) database server to read call setup information and keep call state. All CDRs are written to the database. When we need to take a host out of production, we update OpenSER's dispatcher to stop sending calls to the host. When there are no calls left, we update the OS and Asterisk. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk routine maintenance activities
On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote: > Daily Asterisk restart Do you think its mandatory in production env? > > Daily log rotation > > Voicemail clean up for people leaving an organization. > > > > > __ > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James > Mutuku > Sent: Wednesday, 22 April 2009 3:15 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Asterisk routine maintenance activities > > > > > Hello(s), > > > > > I know this might be test book question or one best suited for google > but I will take the risk of asking. Here I go. What common > routine maintenance tasks do you run on your asterisk box? > > > > > > Thanks > > > James. > > > > > > ___ > -- 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 -- Kurian Mathew Thayil. (GPG KeyID: E232394F) signature.asc Description: This is a digitally signed message part ___ -- 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] Asterisk routine maintenance activities
Daily Asterisk restart Daily log rotation Voicemail clean up for people leaving an organization. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Mutuku Sent: Wednesday, 22 April 2009 3:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk routine maintenance activities Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk routine maintenance activities
Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxing and TIFF files
I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing. Does anyone know of a way, either while producing the file, or after, to tell how many pages have been produced? (without manually viewing the file) Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] notifyringing=no does not work
"
Hello,
If anybody has any idea's to where I should start looking to fix the below
subscription problem. If there is another mailing list I should post this to
please let me know.
Thank you,
Brad Finberg
- Original Message -
From: Brad Finberg
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Date: Thursday, April 9 2009 9:42 AM
Subject: notifyringing=no does not work
Hello,
I have been trying to get my Grandstream GXP2000 phones to stop showing ringing
state on monitored extensions. But no matter where I put notifyringing=no
asterisk still sends the ringing state to the phones. Is this a bug I should
report or is there another way around it.
Here is how i have my subscriptions setup:
extensions.conf
[demo]
exten => 6100,hint,SIP/100
exten => 6101,hint,SIP/101
exten => 6102,hint,SIP/102
exten => 6103,hint,SIP/103
exten => 6104,hint,SIP/104
exten => 6105,hint,SIP/105
exten => _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3)
exten => _1XX,2,Dial(SIP/${EXTEN},20,Tt)
exten => _1XX,3,VoiceMail(${ext...@default,u)
exten => _1XX,104,VoiceMail(${ext...@default,b)
sip.conf
[general]
allowsubscribe=yes
;subscribecontext = default
notifyringing=no
notifyhold=yes
;limitonpeers=yes
[100]
type=peer
context=demo
callerid=Back Office <100>
username=100
secret=(Private)
host=dynamic
nat=no
qualify=yes
canreinvite=no
dtmfmode=rfc2833
call-limit=5
mailbox=...@default
disallow=all
allow=ulaw
allow=alaw
;allow=g723.1
allow=g729
;callingpres=allowed_passed_screen
notifyringing=no
callgroup=1
pickupgroup=1
Asterisk CLI:
Extension Changed 6100[demo] new state Ringing for Notify User 105
Extension Changed 6100[demo] new state Ringing for Notify User 104
Extension Changed 6100[demo] new state Ringing for Notify User 102
Extension Changed 6100[demo] new state Ringing for Notify User 101
Extension Changed 6100[demo] new state Ringing for Notify User 103
Thank you,
Brad Finberg___
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[asterisk-users] Asterisk 1.6.1.0-rc5 Now Available
The Asterisk Development Team is pleased to announce the fifth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc5 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release fixes a couple of issues with realtime music on hold that could cause Asterisk to crash, and an issue that caused hungup channels to stay up, leading to 100% CPU usage. Additionally, several minor issues and edge case scenarios have been resolved. For a full list of changes in this release candidate, please see the ChangeLog: http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc5/ChangeLog Issues found in this release candidate can be reported at http://bugs.digium.com Thank you for your continued support of Asterisk! ___ -- 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] Polycom wideband codecs?
In this case I had, in my hurry this morning, simply confused G.722.1C and G.722.2. These are both low bitrate wide bandwidth codecs. They are also known by the Polycom marketechure nomenclature of Siren7 and Siren14. G.722.1 supporting 7 KHz passband, while G.722.1C support 14 KHz passband. Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves FWD 54245 > Original Message > Subject: Re: [asterisk-users] Polycom wideband codecs? > From: randulo > Date: Tue, April 21, 2009 1:40 pm > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > On Tue, Apr 21, 2009 at 4:40 PM, Steve Underwood wrote: > > Which Polycom supports G.722.2? I think they are only supporting G.722, > > G.722.1 and G.722.1C right now. > > Could someone enlighten me, what is the difference (the result part > that matters, not the spec)? > > r > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom wideband codecs?
On Tue, Apr 21, 2009 at 4:40 PM, Steve Underwood wrote: > Which Polycom supports G.722.2? I think they are only supporting G.722, > G.722.1 and G.722.1C right now. Could someone enlighten me, what is the difference (the result part that matters, not the spec)? r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel to Dahdi
Even if Zaptel is compiled, you can also compile Dahdi because Asterisk will choose the DAHDI-module... it seems. So I left Zaptel... and compiled Dahdi (everything went well, I followed the steps) en then Asterisk again (with dahdi support!!). Yet another episode in this nightmare : [r...@asterisk jonas]# /sbin/service dahdi reload Rerunning dahdi_cfg: Notice: Configuration file is /etc/dahdi/system.conf line 0: Unable to open master device '/dev/dahdi/ctl' 1 error(s) detected [FAILED] [r...@asterisk jonas]# /sbin/service dahdi restart Unloading DAHDI hardware modules: doneLoading DAHDI hardware modules: FATAL: Error inserting dahdi (/lib/modules/2.6.18-128.1.6.el5/dahdi/dahdi.ko): Device or resource busy wctdm24xxp: FATAL: Error inserting wctdm24xxp (/lib/modules/2.6.18-128.1.6.el5/dahdi/wctdm24xxp/wctdm24xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg) [FAILED] wcfxo: FATAL: Error inserting wcfxo (/lib/modules/2.6.18-128.1.6.el5/dahdi/wcfxo.ko): Unknown symbol in module, or unknown parameter (see dmesg) [FAILED] wctdm: FATAL: Error inserting wctdm (/lib/modules/2.6.18-128.1.6.el5/dahdi/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) [FAILED] wcb4xxp: FATAL: Error inserting wcb4xxp (/lib/modules/2.6.18-128.1.6.el5/dahdi/wcb4xxp/wcb4xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg) [FAILED] wctc4xxp: FATAL: Error inserting wctc4xxp (/lib/modules/2.6.18-128.1.6.el5/dahdi/wctc4xxp/wctc4xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg) [FAILED] Error: missing /dev/dahdi! I think the reason there is no /dev/dahdi is because I did not yet edited /etc/dahdi/system.conf /etc/dahdi/system.conf states : "Autogenerated by /usr/sbin/dahdi_genconf on Tue Apr 21 20:23:17 2009 -- do not hand edit" I would put here the following : fxsks=1-3 loadzone=be defaultzone=be But like it says : no manual editing. So I want to run /usr/sbin/dahdi_genconf. But when I do that... nothing happens... Voipinfo.org states : /etc/dahdi/genconf_parameters Fine--tuning parameters for dahdi_genconf (replaces zapconf and also deprecates genzaptelconf). But check this : [r...@asterisk dahdi]# ls /etc/dahdi/ init.conf modules system.conf system.conf.bak NO such file ! With Zaptel, I had a working kernel-module that recognized my hardware... but no zapata.conf. Now that I've compiled DAHDI, I'm again missing some files. Can someone please give some advice. Thanks in advance... Greetingz, Jonas. On Mon, 2009-04-20 at 14:13 -0400, Joshua Kinard wrote: > Converting is actually pretty straightforward: > > Bare minimum: > /etc/zaptel.conf --> /etc/dahdi/system.conf > /etc/asterisk/zapata.conf --> /etc/asterisk/chan_dahdi.conf > > Any reference to ZAP/* becomes DAHDI/* in your asterisk conf files > (i.e., extensions.conf). > > Granted, all I use Asterisk for is a fax-to-email mechanism in > conjunction with my ~18yr-old Rolm system, but I imagine more complex > setups are probably not too hard to replace. Most of it was just > search & replace in the extensions.conf file. You can also leave the > older zapata.conf intact. I believe newer Asterisk version will > ignore the file's existance. Ditto on the older zaptel.conf, since > the dadhi code doesn't even reference it I believe. > > The only thing I miss, is I thought Zaptel was a pretty cool name. > Like, lasers shooting everytime a call comes in or something. "Dahdi" > makes me think of angels singing everytime a call comes in now. > > HTH, > > --J > > ___ -- 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] Asterisk Database
Benny Amorsen writes: > Asterisk DB is either an SQLite database or a Berkeley database, I > forget which (did it change?). Either way, 20,000 should be a problem > for the underlying database. Should NOT be a problem for the underlying database. Sorry! /Benny ___ -- 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] Voice mail does not contain a time?
On Tue, 21 Apr 2009, Tzafrir Cohen wrote:
> WAV is a pretty simple container format. The length is written in a very
> expected place in the header:
>
> http://en.wikipedia.org/wiki/.wav
> http://ccrma.stanford.edu/courses/422/projects/WaveFormat/
>
> E.g. the following:
>
> wav_size() {
> LANG=C cut -b 41-44 "$1" | head -n 1 | hexdump -e "\"$1:"'$1: %u\n"' |
> head -n 1
> }
Or:
wav_size()
{
dd bs=1 count=4 if="$1" skip=40 2>/dev/null\
| hexdump -e "\"$1:"'$1: %u\n"'
}
Or, if hexdump's "-s" did what the man page says it did:
wav_size()
{
dd count=1 if="$1" 2>/dev/null\
| hexdump -s 40 -n 4 -e "\"$1:"'$1: %u\n"'
}
but "-s" does a "seek" not a "skip" as documented thus seeking on stdin
fails.
I always learn something new from Tzafrir's postings :)
I still haven't figured out the format string...
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] MOH always plays from the start
On Fri, Apr 17, 2009 at 11:39 AM, Mike wrote: > True, my mistake: I upgraded to 1.4.24.1, and the MoH file still always > start from the beginning. I believe I'm experiencing the same thing with my music on hold. I also would prefer a continuous play in the background, and I'm using asterisk-1.6.0.6. In my usage these are each separate channels, and I'd prefer that when these people encounter hold music they are all getting the same message at the same time. I think I'm just misunderstanding the settings for music on hold classes, and that if I set a class to play continuously I can have any app that uses hold music be playing the same music in sync with any other channels using moh at that moment. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] run dialplan when open line
Hi all, Does asterisk support the following scenario? I need when a customer who own an endpoint registered on asterisk open the line, the asterisk will run a specific AGI script inside the endpoint context? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
On 21-Apr-09, at 1:06 PM, Anthony Francis wrote: >> You are correct, not seeing that means that the signaling was >> either in >> the audio stream (which doesn't survive compression) or it was sent >> in >> the sip signaling. However one must also note that your ITSP's >> gateway >> may have been having problems with their DTMF detection on their >> PRI's. >> >> Anthony >> > > Also, to determine if you are sending DTMF out of band (as part of IAX > signalling) do iax2 debug peer > in the CLI. > You will see when it creates DTMF events. Not being much of a DTMF guru, but trying to follow along the best I can in hopes of troubleshooting my own DTMF issues, I seem to be getting hung up having DTMF pass from a SIP channel to an IAX channel: 7970 -- SIP/ulaw -- ast -- IAX/ulaw -- ITSP With: core set debug channel IAX2/x.x.x.x-13779 and... core set debug channel SIP/phone-cisco-1-089a55d0 ... set, I generate DTMF via the ITSP and see: << [ TYPE: DTMF Begin (12) SUBCLASS: 2 (50) ] [IAX2/x.x.x.x-13779] >> [ TYPE: DTMF Begin (12) SUBCLASS: 2 (50) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: DTMF End (1) SUBCLASS: 2 (50) ] [IAX2/x.x.x.x-13779] >> [ TYPE: DTMF End (1) SUBCLASS: 2 (50) ] [SIP/phone-cisco-1-089a55d0] So that tells me that ast is receiving DTMF from ITSP and sending it to my phone. If I then try to generate DTMF via the phone towards the ITSP on the same call, I see: << [ TYPE: DTMF Begin (12) SUBCLASS: 2 (50) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: DTMF End (1) SUBCLASS: 2 (50) ] [SIP/phone-cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/phone- cisco-1-089a55d0] No activity on the IAX side. I've tried dtmfmode=rfc2833 in sip.conf to try to make sure the SIP side matches IAX's out of band, but no dice... This has been happening since my change to 1.4.23 from 1.2. so the upgrade is certainly the catalyst, but I can't figure out what has changed since 1.2 that is relevant to DTMF being received by a SIP channel not being transmitted out to an IAX channel. Any pointers would be appreciated. ___ -- 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] Asterisk Database
On Tue, 21 Apr 2009, Doug Lytle wrote: > Benny Amorsen wrote: >> Asterisk DB is either an SQLite database or a Berkeley database, I >> > > The last I knew, it was BerkeleyDB. > > Doug > > Just to add a few cents, if the object is to store and retrieve a single value with a single key, Berkeley DB is perfectly suited to the task. It shouldn't matter the number of rows, and is far less overhead than a giant SQL engine. I don't actually recall the original question, but it sounded at the time that he just wanted to store a single value against a single key, so this may be the most efficient method of going about it, and is certainly the least complex... j ___ -- 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] Asterisk Database
On Tue, 21 Apr 2009, Benny Amorsen wrote: > "Sriram" writes: > >> 1. I need to store the CallerId of the PSTN caller with his language >> preference so that next time he is played the prompt in his language that >> he chose the first time.What would be better - storing his number in the >> Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the >> dial plan and quering it everytime to see the callers record ? how many >> records can AstDB handle safely ? In my case the total records wont exceed >> 20,000 since there are many repeat callers ? > > Asterisk DB is either an SQLite database or a Berkeley database, I > forget which (did it change?). Either way, 20,000 should be a problem > for the underlying database. 1.2 uses Berkeley: file /var/lib/asterisk/astdb /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3, native byte-order) > I'd still go for the "real" database (using Postgres, but I guess you > can use MySQL if you feel like it), probably using func_ODBC. With > Asterisk DB you have to go through Asterisk to view or change contents > of the database; a real database makes management easier. +1 for MySQL (or whatever "real" DB you know). Bet on your boss asking questions like: "Can I have a web page with a pretty pie graph showing the breakdown of who joined with one of those calendar thingies so I can choose a date range?" Using the "-r -x foo" command line interface or talking to Asterisk's database "behind its back" both sound like bad ideas to me. Imagine if you muck something up and it corrupts the database and you can't restart Asterisk. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
Anthony Francis wrote: > Jeff LaCoursiere wrote: > >> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: >> >> >> >>> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: >>> >>> >>> > I went ahead and switched to SIP just for grins, and made sure > "dtmfmode=rfc2833" is in the peer config on both sides and in the entry > for the phone. So now it is: > > polycom501---SIP/ulaw--->ast1---SIP/g729--->ast2---IAX/ulaw--->ITSP > > >>> A bit more information. ast1 is running 1.4.23.1 and I noticed a debug >>> line >>> in rtp.c: >>> >>>if (rtpdebug || option_debug > 2) >>>ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", >>> event, len); >>> >>> So I set debug to 10 and caught this line: >>> >>> [Apr 17 17:28:02] DEBUG[27264] rtp.c: - RTP 2833 Event: 0002 (len = 4) >>> >>> So I guess that proves that from the phone to ast1 RFC2833 is in effect (I >>> did actually press the digit '2', which I assume is the event code above?). >>> >>> I tried to do the same on ast2, which is running 1.4.22.1, and with debug >>> set >>> to 10 I did *not* get this message, which makes me think that RCF2833 is >>> NOT >>> in effect for the trunk between ast1 and ast2. Is that reasonable? >>> >>> >>> >> The main problem turned out to be at my ITSP, and is now resolved. The >> question remains for me, though, how to interpret the debug lines I was >> able to catch (or not) above. >> >> How do you really know if RFC2833 signalling is being received? I caught >> the debug message on ast1 but not on ast2. I am using ulaw between ast2 >> and the ITSP, and I am now wondering if the DTMF is being sent inband on >> that last leg since I could not catch the debug messages on ast2. Perhaps >> what they did to fix on their end is simply remove compression between >> themselves and the PSTN. >> >> I would really like a concrete method of verifying that DTMF signalling is >> being sent out of band on my outbound IAX link. Any ideas? >> >> Thanks, >> >> j >> >> >> > You are correct, not seeing that means that the signaling was either in > the audio stream (which doesn't survive compression) or it was sent in > the sip signaling. However one must also note that your ITSP's gateway > may have been having problems with their DTMF detection on their PRI's. > > Anthony > Also, to determine if you are sending DTMF out of band (as part of IAX signalling) do iax2 debug peer in the CLI. You will see when it creates DTMF events. ___ -- 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] Asterisk Database
Benny Amorsen wrote: > Asterisk DB is either an SQLite database or a Berkeley database, I > The last I knew, it was BerkeleyDB. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
Jeff LaCoursiere wrote: > On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: > > >> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: >> >> I went ahead and switched to SIP just for grins, and made sure "dtmfmode=rfc2833" is in the peer config on both sides and in the entry for the phone. So now it is: polycom501---SIP/ulaw--->ast1---SIP/g729--->ast2---IAX/ulaw--->ITSP >> A bit more information. ast1 is running 1.4.23.1 and I noticed a debug line >> in rtp.c: >> >>if (rtpdebug || option_debug > 2) >>ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", >> event, len); >> >> So I set debug to 10 and caught this line: >> >> [Apr 17 17:28:02] DEBUG[27264] rtp.c: - RTP 2833 Event: 0002 (len = 4) >> >> So I guess that proves that from the phone to ast1 RFC2833 is in effect (I >> did actually press the digit '2', which I assume is the event code above?). >> >> I tried to do the same on ast2, which is running 1.4.22.1, and with debug >> set >> to 10 I did *not* get this message, which makes me think that RCF2833 is NOT >> in effect for the trunk between ast1 and ast2. Is that reasonable? >> >> > > The main problem turned out to be at my ITSP, and is now resolved. The > question remains for me, though, how to interpret the debug lines I was > able to catch (or not) above. > > How do you really know if RFC2833 signalling is being received? I caught > the debug message on ast1 but not on ast2. I am using ulaw between ast2 > and the ITSP, and I am now wondering if the DTMF is being sent inband on > that last leg since I could not catch the debug messages on ast2. Perhaps > what they did to fix on their end is simply remove compression between > themselves and the PSTN. > > I would really like a concrete method of verifying that DTMF signalling is > being sent out of band on my outbound IAX link. Any ideas? > > Thanks, > > j > > You are correct, not seeing that means that the signaling was either in the audio stream (which doesn't survive compression) or it was sent in the sip signaling. However one must also note that your ITSP's gateway may have been having problems with their DTMF detection on their PRI's. Anthony ___ -- 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] Asterisk Database
I second the "Real" database idea. AFAIK, the Asterisk database is still a Berkley DB. I'm accessing Postgres using an AGI and returning dialplan variables with what I want to process. The Asterisk database is best for small, non-critical information, though there are good procedures documented for backup and reload of it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen Sent: Tuesday, April 21, 2009 11:39 AM To: Sriram Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Database "Sriram" writes: > 1. I need to store the CallerId of the PSTN caller with his language > preference so that next time he is played the prompt in his language that > he chose the first time.What would be better - storing his number in the > Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the > dial plan and quering it everytime to see the callers record ? how many > records can AstDB handle safely ? In my case the total records wont exceed > 20,000 since there are many repeat callers ? Asterisk DB is either an SQLite database or a Berkeley database, I forget which (did it change?). Either way, 20,000 should be a problem for the underlying database. I'd still go for the "real" database (using Postgres, but I guess you can use MySQL if you feel like it), probably using func_ODBC. With Asterisk DB you have to go through Asterisk to view or change contents of the database; a real database makes management easier. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Database
"Sriram" writes: > 1. I need to store the CallerId of the PSTN caller with his language > preference so that next time he is played the prompt in his language that > he chose the first time.What would be better - storing his number in the > Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the > dial plan and quering it everytime to see the callers record ? how many > records can AstDB handle safely ? In my case the total records wont exceed > 20,000 since there are many repeat callers ? Asterisk DB is either an SQLite database or a Berkeley database, I forget which (did it change?). Either way, 20,000 should be a problem for the underlying database. I'd still go for the "real" database (using Postgres, but I guess you can use MySQL if you feel like it), probably using func_ODBC. With Asterisk DB you have to go through Asterisk to view or change contents of the database; a real database makes management easier. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk process ended
Hi All, Thanks for your answers. Asterisk (v1.2.7.1) runs on RedHat AS 4 without -g option. It's really a crash, the process not running at all according to "ps aux". Regards, A.L ___ -- 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] DTMF
On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: > > On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: > >>> I went ahead and switched to SIP just for grins, and made sure >>> "dtmfmode=rfc2833" is in the peer config on both sides and in the entry >>> for the phone. So now it is: >>> >>> polycom501---SIP/ulaw--->ast1---SIP/g729--->ast2---IAX/ulaw--->ITSP > > A bit more information. ast1 is running 1.4.23.1 and I noticed a debug line > in rtp.c: > >if (rtpdebug || option_debug > 2) >ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", > event, len); > > So I set debug to 10 and caught this line: > > [Apr 17 17:28:02] DEBUG[27264] rtp.c: - RTP 2833 Event: 0002 (len = 4) > > So I guess that proves that from the phone to ast1 RFC2833 is in effect (I > did actually press the digit '2', which I assume is the event code above?). > > I tried to do the same on ast2, which is running 1.4.22.1, and with debug set > to 10 I did *not* get this message, which makes me think that RCF2833 is NOT > in effect for the trunk between ast1 and ast2. Is that reasonable? > The main problem turned out to be at my ITSP, and is now resolved. The question remains for me, though, how to interpret the debug lines I was able to catch (or not) above. How do you really know if RFC2833 signalling is being received? I caught the debug message on ast1 but not on ast2. I am using ulaw between ast2 and the ITSP, and I am now wondering if the DTMF is being sent inband on that last leg since I could not catch the debug messages on ast2. Perhaps what they did to fix on their end is simply remove compression between themselves and the PSTN. I would really like a concrete method of verifying that DTMF signalling is being sent out of band on my outbound IAX link. Any ideas? Thanks, j ___ -- 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] Asterisk process ended
Marco Sambo escribió: Hi, I have the same problem: sometimes my Asterisk box crash (or similar) and in asterisk log doesn't appear nothing. Also into syslog. I don't understand what is it Marco Well, when asterisk dies "without leaving trace", it's generally a core dump. That means at asterisk instantly died for some internal or app malfunction. If you work with asterisk as a service (on RedHat/CentOS), or with a startup script, the asterisk process is normally started with the -g option that leaves a convenient core dump file in /tmp that let you see what caused the crash. This info is usually complicated to understand if you are not a programmer. Take a look here: http://www.voip-info.org/wiki/view/Asterisk+debugging On section "Backtracing a core dump file in /tmp". Regards, 2009/4/21 Adrien Lemoine mailto:alemo...@legos.fr>> Hi all, I experienced for a second time the crash of asterisk process during the night. Nothing in Asterisk messages logs, nothing in /var/log/messages can explain that. Maybe someone experienced something similar and can drive me in the resolution ? Regards, A.L ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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] Polycom wideband codecs?
mgra...@mstvp.com wrote: > Doing a little research before Friday's Voip Users Conference call with > Dan Behringer. > > Are any of the newer Polycom wideband codecs implemented in v1.6? > Specifically, G.722.1 or G.722.2? > Which Polycom supports G.722.2? I think they are only supporting G.722, G.722.1 and G.722.1C right now. Steve ___ -- 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] Asterisk process ended
Is it a crash or you find that the phones are not registered ? Do you loose internet connection during the night ? Do you have SIP trunks ? Is it vanilla asterisk or a specific distro ? G. On Tue, Apr 21, 2009 at 6:25 AM, Barry L. Kline wrote: > Adrien Lemoine wrote: > > > Maybe someone experienced something similar and can drive me in the > > resolution ? > > You have given no information about your hardware, OS, Asterisk version > or what you need to do to recover the system (e.g. reboot, just restart > Asterisk, etc) so no one is going to be able to do much to offer help. > > Barry > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom wideband codecs?
mgra...@mstvp.com wrote: > Doing a little research before Friday's Voip Users Conference call with > Dan Behringer. > > Are any of the newer Polycom wideband codecs implemented in v1.6? > Specifically, G.722.1 or G.722.2? Asterisk 1.6 has passthrough/record/playback support for G.722.1 (Siren7) and G.722.1 Annex C (Siren14). There is no support for G.722.2, I'm not sure that is even a Polycom codec. We are working on producing transcoder (codec) modules for these codecs as well, so hopefully in the near future we'll be able to transcode between these codecs and the others already supported. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Database
i'd use mysql... and i do use mysql for this... 2009/4/21 Sriram > My setup : Trixbox 2.6.1 & TE410P running well .: > > 1. I need to store the CallerId of the PSTN caller with his language > preference so that next time he is played the prompt in his language that he > chose the first time.What would be better - storing his number in the > Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial > plan and quering it everytime to see the callers record ? how many records > can AstDB handle safely ? In my case the total records wont exceed 20,000 > since there are many repeat callers ? > > rgds > Sriram > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom wideband codecs?
Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves ___ -- 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] Voice mail does not contain a time?
On Mon, Apr 20, 2009 at 10:45:00AM -0400, Justin Piszcz wrote:
> Hello,
>
> When a voice message is saved and e-mailed as a wav, the total time of the
> voice mail does not show up in, e.g., windows media player, why is this?
>
> I have only used wav49/wav:
>
> ; Use wav49 format for all voicemail messages
> format=wav49|gsm|wav
WAV is a pretty simple container format. The length is written in a very
expected place in the header:
http://en.wikipedia.org/wiki/.wav
http://ccrma.stanford.edu/courses/422/projects/WaveFormat/
E.g. the following:
wav_size() {
LANG=C cut -b 41-44 "$1" | head -n 1 | hexdump -e "\"$1:"'$1: %u\n"' |
head -n 1
}
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
___
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Re: [asterisk-users] Asterisk process ended
Adrien Lemoine wrote: > Maybe someone experienced something similar and can drive me in the > resolution ? You have given no information about your hardware, OS, Asterisk version or what you need to do to recover the system (e.g. reboot, just restart Asterisk, etc) so no one is going to be able to do much to offer help. Barry ___ -- 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] Voice mail does not contain a time?
On Tuesday 21 April 2009 08:07:17 Justin Piszcz wrote: > On Mon, 20 Apr 2009, Justin Piszcz wrote: > > When a voice message is saved and e-mailed as a wav, the total time of > > the voice mail does not show up in, e.g., windows media player, why is > > this? > > > > I have only used wav49/wav: > > > > ; Use wav49 format for all voicemail messages > > format=wav49|gsm|wav > > Any clues? Seems like you should be asking the makers of Windows Media Player. That's not us. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk process ended
Hi, I have the same problem: sometimes my Asterisk box crash (or similar) and in asterisk log doesn't appear nothing. Also into syslog. I don't understand what is it Marco 2009/4/21 Adrien Lemoine > Hi all, > > > > I experienced for a second time the crash of asterisk process during the > night. > > > > Nothing in Asterisk messages logs, nothing in /var/log/messages can explain > that. > > > > Maybe someone experienced something similar and can drive me in the > resolution ? > > > > Regards, > > > > A.L > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Should I go for Asterisk 1.6 ?
--[ UxBoD ]-- wrote: > I am going to be building a new home Asterisk server this weekend > (Dual core Intel Atom & 2GB RAM) and would like to ask whether it > would be worth starting fresh with a 1.6 install instead of the 1.4 > one I have at the moment ? I do not have a complicated dialplan as it > only serves a couple of number and three extensions. For inbound and > outbound I am using the IAX2 protocol instead of SIP. > > Any thoughts or help would be most gratefully accepted. I have been using 1.6.0.x now for a while with minor (non-critical) issues. If you already have a working Asterisk system, so you don't need to replace things right NOW, why not load up 1.6? That way you can learn about the changes between 1.4 and 1.6 at your leisure. For example, if you're still using Zaptel then you will need to learn the minimal changes required to go to DAHDI. Once you are done testing your server will be at the current level and can drop in as a replacement. Barry ___ -- 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] Voice mail does not contain a time?
On Mon, 20 Apr 2009, Justin Piszcz wrote: > Hello, > > When a voice message is saved and e-mailed as a wav, the total time of the > voice mail does not show up in, e.g., windows media player, why is this? > > I have only used wav49/wav: > > ; Use wav49 format for all voicemail messages > format=wav49|gsm|wav > > Justin. > Any clues? Justin. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Should I go for Asterisk 1.6 ?
Hi, I am going to be building a new home Asterisk server this weekend (Dual core Intel Atom & 2GB RAM) and would like to ask whether it would be worth starting fresh with a 1.6 install instead of the 1.4 one I have at the moment ? I do not have a complicated dialplan as it only serves a couple of number and three extensions. For inbound and outbound I am using the IAX2 protocol instead of SIP. Any thoughts or help would be most gratefully accepted. Best Regards, UxBoD -- SplatNIX IT Services :: Innovation through collaboration ___ -- 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] Asterisk Database
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Sriram wrote: > > 1. I need to store the CallerId of the PSTN caller with his language > preference so that next time he is played the prompt in his language > that he chose the first time.What would be better - storing his number > in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL > from the dial plan and quering it everytime to see the callers record ? > how many records can AstDB handle safely ? In my case the total records > wont exceed 20,000 since there are many repeat callers ? 20K records? While I'm not sure exactly how many records AstDB could handle it would seem to me that 20K would be a high number. My inclination would be to use a full database... perhaps you'd like to store more about that callerID than just the caller's preferred language. Using a real DB would certainly make that easier. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ7bRRCFu3bIiwtTARAqTvAJ4jS0/kZeHo33+w9gjZ88dYB3SeDACgg2+t LhVIBsPzxyQ/g542/NjMo8U= =d+JZ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Database
My setup : Trixbox 2.6.1 & TE410P running well .: 1. I need to store the CallerId of the PSTN caller with his language preference so that next time he is played the prompt in his language that he chose the first time.What would be better - storing his number in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial plan and quering it everytime to see the callers record ? how many records can AstDB handle safely ? In my case the total records wont exceed 20,000 since there are many repeat callers ? rgds Sriram___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cleared Event Log
I am using IBM Server I cleared the event log from BIOS and asterisk couldn't run which file i have to create ? and what is its permission? thanks a lot _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/products/events.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk process ended
Hi all, I experienced for a second time the crash of asterisk process during the night. Nothing in Asterisk messages logs, nothing in /var/log/messages can explain that. Maybe someone experienced something similar and can drive me in the resolution ? Regards, A.L ___ -- 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] asterisk 420 Bad Response
On 21 Apr 2009, at 10:46, Khaled W. Chehab wrote: > Dears, Hi.. > When my GW send a call to asterisk v 1.4.24 , What is your GW. Hardware, software etc etc > Asterisk send Status: 420 bad extension (unsupported) Ok. SIP trace available? > Why? Show us the logs/sip trace. > Any modifications should be done one sip.conf Why? > regards Ditto. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 420 Bad Response
Dears, When my GW send a call to asterisk v 1.4.24 , Asterisk send Status: 420 bad extension (unsupported) Why? Any modifications should be done one sip.conf regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
