Re: [asterisk-users] Busy on SIP
Hi Ira,
for Aastra phones I have done this application to resolve busy/xfer
transfer:
extensions.conf
===
exten = _1X,1,GotoIf($[${SIPPEER(${EXTEN}|curcalls)}1]?free:busy)
exten = _1X,n(free),Dial(SIP/${EXTEN},,tTr)
exten = _1X,n,Hangup()
exten = _1X,n(busy),Busy()
exten = _1X,n,Hangup()
sip.conf
===
[intphones](!)
type=friend
qualify=yes
host=dynamic
callgroup=1
pickupgroup=1
subscribecontext=BLF_group
dtmfmode=info
[10](intphones)
context=IntPhones
username=10
secret=1234
amaflags=documentation
accountcode=sip10
callerid=sip10 10
call-limit=2
dial=SIP/10
canreinvite=no
And this resolve for me problems for busy and for xfer Aastra button.
Marco
2009/3/17 Ira i...@extrasensory.com
At 01:29 AM 3/17/2009, you wrote:
But there is another little problem. On Aastra phone (on other
phones I don't try yet), the xfer button doesn't work, until I set
call-limit=2, but making this I find the phone not busy.
As far as I can tell on my Aastra phones it takes 2 links to complete
a transfer. Pressing transfer puts the first call on hold and allows
you to make a second call. Pressing transfer a second time then
connects those to calls together and removes you from the call. If
you only have 1 call allowed you'll need to implement that using
features.conf or implement the busy stuff in the dial plan.
Ira
___
-- 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 is not designed for University with large user base?
I am working in a university also , and nowadays, we are aking some tests to start using asterisk in some areas of our campus. Because it costs a lot more cheper than extending our PBX system. It seems ok for us to make a hybrid system in the campus area which should be about 1000 clients for the begining. Maybe in the future we can dump our old ericsson and swtich to asterisk completely..who knows :) On Mon, Mar 16, 2009 at 5:34 PM, Vincent Li vincent.mc...@gmail.com wrote: Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. http://www.networkworld.com/news/2007/071707-open-source-voip.html http://www.digium.com/en/company/casestudies/viewcasestudies/University-of-Pennsylvania Those links gets passed around every time this topic comes up. I don't know what metrics led to the conclusion of the project manager, nor the way things were configured in your particular pilot. Asterisk-1.6 has dramatically enhanced SIP handling compared to 1.4. It also has dramatically faster large-dialplan handling. You can read all about it in the files that come packaged with 1.6. It's possible (I would dare say likely) that the project manager is looking at old data, or that the pilot was done with old versions 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 ___ -- 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] Manager API Originate CDR Problem, all is NO ANSWER
hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten = s,1,Answer() exten = s,n,Wait(10) exten = s,n,Hangup() when the phone 888 pick up , it will come to callout context, after hangup, one cdr generate, but the cdr disposition is still NO ANSWER ... and the billsec is 0, Although i can use duration instead, is this a bug ? or how to set it right?? thanks a lot.___ -- 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] PBX to gate interface
There are various ways of doing this. You could use an analogue port/ATA and connect any good old fashioned intercom to it (Pantel are a good make). You can now get SIP intercom systems as well. I haven't tried on of these - but they look good (and can contain a camera as well if needed). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: 17 March 2009 13:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PBX to gate interface Has anyone found a good wayt o do a gate intercom using Asterisk? I am looking at a Xorcom PBX with programmable contact, so I have no issue with opening the gate, but the interface at the gate is a bit tricky. I thought about a weather proof housing containing a phone but it seems a bit tacky. I also looked at a handsfree erather proof phone, but at $600 it is a bit steep. Any solutions that have been implemented successfully? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI or Zaptel doesn't compile against 1.4.24
John Knight a écrit : make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. specifically Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing Are you using the stock Debian kernel? If so, do you have the linux kernel source and kernel headers source package installed? If so, make sure the source packages installed are the same version number of the current running kernel. No stock kernel, build with make-kpkg from linux-source 2.6.18 d...@keewi:/usr/src/dahdi-linux-2.1.0.4$ uname -a Linux keewi 2.6.18-custom.2 #1 Sat Nov 29 22:07:56 CET 2008 i686 GNU/Linux -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI or Zaptel doesn't compile against 1.4.24
Tzafrir Cohen a écrit : On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote: Hi, We installed the latest 1.4.24 on a test machine and can't get zaptel nor dahdi compile. It's a Linux Debian Etch. Errors we have: keewi:/usr/src/dahdi-linux-2.1.0.4# make make -C /lib/modules/2.6.18-custom.2/build ARCH=i386 SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.1.0.4/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.o In file included from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:38: /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:23:27: error: linux/version.h: Aucun fichier ou répertoire de ce type This is plain wrong. Your source tree is bad. What kernel version do you want to build dahdi against? What kernel version do you use? uname -a d...@keewi:/usr/src/dahdi-linux-2.1.0.4$ uname -a Linux keewi 2.6.18-custom.2 #1 Sat Nov 29 22:07:56 CET 2008 i686 GNU/Linux Well, it seems that something is wrong with my kernel tree. I will build a new kernel from sources. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Noisy Ring Back Tone with TE205P card
Hi,
If anyone has the same problem, I solved it doing:
genzaptelconf -sdv
It might have been a problema with the card or the module.
Regards
Imanol Pardavila escribió:
Hi,
I stilll continue with the problem but I have noticed something new
that maybe a clue. The noise during the call progress is made by the
appearance of the different lines in the asterisk CLI, I mean, each
line is posted in the CLI generates a noise in the call's signallling
tone. For example, if I try doing a call during a resetinterval
option, which reset all free channels, each line posted in the CLI
generates a burst noise.
Any ideas?
Thanks
Regards
Imanol Pardavila escribió:
Hi,
I am having problems with an Asterisk with a Digium TE205P card. The
issue is that the Ring Back Tone is noisy. I am making modem's calls
and this noise influences on the initial negotiation protocol, so
modems have to recall.
My configuration is:
Asterisk version: Asterisk 1.4.21.2
Linux version: CentOS release 5.2 (Final)
Card: Digium TE205P
##zapata.conf#
; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended
; to be #include-d by /etc/zapata.conf that will include the global
settings
;
[channels]
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1
language=es
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialpla=national
signalling=pri_cpe
resetinterval=never
group=1
channel = 1-15,17-31
; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
group=2
channel = 32-46,48-62
##zaptel.conf#
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
hardhdlc=16
# Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
hardhdlc=47
# Global data
loadzone= es
defaultzone = es
##extensions.conf###
exten =999888777,1,Goto(JUMP,s,1)
[JUMP]
exten = s,1,Dial(Zap/R2/6,15,r)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Goto(HANGUP,s,1)
exten = s-NOANSWER,1,Goto(HANGUP,s,1)
exten = s-CHANUNAVAIL,1,Goto(HANGUP,s,1)
exten = s-CONGESTION,1,Goto(HANGUP,s,1)
[HANGUP]
exten = s,1,Hangup
[DID_span_1]
include = default
[DID_span_2]
include = default
I have no idea about where could be the problem. I can't see anything
rare in the logs
Any ideas?
Thanks
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Performance of realtime for millions of SIP user
Hi, Would you please let me know the performance of asterisk realtime in case I will have millions of SIP users? Thanks, Krunal Patel ___ -- 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 and G.726 Codec
Le'an Liu wrote: My questions: 1. G.726 16/24/32/40 supported in asterisk-1.6.0.5? No. Only G726-32 is supported in all Asterisk versions. -- 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] SPA3102 - How to save config in a file
Stefan Schmidt wrote: hello, you could retrieve the config from you SPA with the following url: http://ipofyourphone/admin/spacfg.xml . That works well with the Linksys phones, but not with the SPA-3102 which isn't really a phone, but an ATA. My 3102 has software version 5.1.6. /Per Jessen, Zürich ___ -- 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] Callerid charset problems
Hi, I'm having problems when the callerid of a user defined in the sip.conf contains special characters such as: ñ, á, é, í, ó , etc. The strange thing is that these characters are displayed correctly in the dialplan by using the sip show peer command, but if this user makes a call, these characters are not displayed correcly in the SIP message. Any ideas of what might be happening? Thank you in advance. Regards, Santi ___ -- 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] Performance of realtime for millions of SIP user
Krunal Patel wrote: Hi, Would you please let me know the performance of asterisk realtime in case I will have millions of SIP users? I don't think it will work on a single server, with or without realtime. If only a very small amount of them would be online at any moment, maybe it will work (although i doubt it), it's easy to test by filling your database with a couple of million of bogus records. Zoa Thanks, Krunal Patel ___ -- 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] PBX to gate interface
How does a Push-to-talk intercom interface with Asterisk? Andrew Thomas wrote: There are various ways of doing this. You could use an analogue port/ATA and connect any good old fashioned intercom to it (Pantel are a good make). You can now get SIP intercom systems as well. I haven't tried on of these - but they look good (and can contain a camera as well if needed). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: 17 March 2009 13:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PBX to gate interface Has anyone found a good wayt o do a gate intercom using Asterisk? I am looking at a Xorcom PBX with programmable contact, so I have no issue with opening the gate, but the interface at the gate is a bit tricky. I thought about a weather proof housing containing a phone but it seems a bit tacky. I also looked at a handsfree erather proof phone, but at $600 it is a bit steep. Any solutions that have been implemented successfully? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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 is not designed for University large scale
IMHO when users scale up to such levels, Asterisk falls short, I made a c ouple large implementations and the best approach is using OpenSer as SIP engine (along with his own media proxy if required by your network schema) and use Asterisk as Vertical Services Provider, such as email, IVR, in general, expliding the benefits Asterisk overachieve, including TDM interconnection as well. ___ -- 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] PBX to gate interface
Have a look at http://www.northsupply.co.uk/ (under Door Access Systems). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: 18 March 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PBX to gate interface How does a Push-to-talk intercom interface with Asterisk? Andrew Thomas wrote: There are various ways of doing this. You could use an analogue port/ATA and connect any good old fashioned intercom to it (Pantel are a good make). You can now get SIP intercom systems as well. I haven't tried on of these - but they look good (and can contain a camera as well if needed). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: 17 March 2009 13:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PBX to gate interface Has anyone found a good wayt o do a gate intercom using Asterisk? I am looking at a Xorcom PBX with programmable contact, so I have no issue with opening the gate, but the interface at the gate is a bit tricky. I thought about a weather proof housing containing a phone but it seems a bit tacky. I also looked at a handsfree erather proof phone, but at $600 it is a bit steep. Any solutions that have been implemented successfully? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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 and G.726 Codec
2009/3/18 Kevin P. Fleming kpflem...@digium.com Le'an Liu wrote: My questions: 1. G.726 16/24/32/40 supported in asterisk-1.6.0.5? No. Only G726-32 is supported in all Asterisk versions. Perhaps the confusion in the voip-info page mentioned is due to the other G726 rates being supported in files with format_g726? ___ -- 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] PBX to gate interface
On Wed, 18 Mar 2009, Chris Mason (Lists) wrote: How does a Push-to-talk intercom interface with Asterisk? I think the generic answer is expensively. If Xorcom made just the IO part of their channel banks then it might be cheaper, however ... What I've seen so-far is an intelligent box with what's effectively an analogue phone interface that you connect to an FXS port. The intercom is basically a hardened phone - you push the button, the phone comes off-hook and dials a pre-set number. (or just goes off-hook and expects bat phone mode). The other end can send DTMF codes down to activate a relay which can then power a door opener - or not. Some boxes have a keypad, allowing a local code to be dialled which activates the relay directly, and some can dial internal extensions, letting the person who answers key in the unlock code. Some can even answer a call and listen for tones, so anyone could call the door-phone and unlock the door (eg. from your mobile phone while standing outside) But sometimes you just want a relay connected to the asterisk box because you have everything else - I eventually found a German company doing USB mains power cyclers which you could control from a Linux command-line program, so no need for a xorcom box after all. The options are endless as are the prices you can expect to pay. It all depends on the intelligence you need in the door. Ones I'm looking at right now: http://www.voipon.co.uk/2n-analogue-helios-door-entry-system-c-22_276_279.html Thats sort of the entry level. At least they have prices on their website unlike the recently posted North supply Link! (Why do sites do that?) Gordon ___ -- 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] T1 problem (call using a .call file)
On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno tipas...@gmail.com wrote: I have a weird problem with call using my T1 card. I can make calls fine using my analog and IP phones, but when I try to initiate a call using a .call file, I get the following error -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 127 received it happens on certain numbers I dial, but if I dial that same number with an ip or analog phone that use the T1 channel, the call is going through normally. Anybody knows why? Are you doing anything silly with prefixing or short-circuit dialing? in other words.. You dial 8 for an outside line, then 1+10 digits and you're forgetting to do that for some numbers? ___ -- 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 talking to mysql database through odbc
Hi, I have 2 instances of asterisk running and taking to common mysql database via ODBC connection. I am facing some issue while running bulk call (around 100 calls at a time),like few call gets error out of 100, i am suspecting that SQL query is failing and the error i get in one the call failure is .. [*Mar 12 03:29:55] WARNING[27752] chan_sip.c: sip_xmit of 0xb76a27fc (len 446) to 114.105.116.121:26991 returned -1: Address family not supported by protocol* *[Mar 12 03:29:56] WARNING[27752] chan_sip.c: sip_xmit of 0xb76a27fc (len 446) to 114.105.116.121:26991 returned -1: Address family not supported by protocol* *[Mar 12 03:29:58] WARNING[27752] chan_sip.c: sip_xmit of 0xb76a27fc (len 446) to 114.105.116.121:26991 returned -1: Address family not supported by protocol* Anybody faced this issue...? Also what's the significance of rtcachefriends=yes in sip.conf, if i enable this error frequency reduces, but error occurs eventually. -Sumanth ___ -- 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] T1 problem (call using a .call file)
Nope, I always dial 1 + 10 digits for all my numbers. It works on all numbers when I am using my phone (Analogue or IP) but when I do it using a .call file it does not work on some numbers mostly. That is the weirdest thing I have ever seen. I tried different codecs in the call file, I still get the PROGRESS with cause code 127 On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg dbackeb...@gmail.comwrote: On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno tipas...@gmail.com wrote: I have a weird problem with call using my T1 card. I can make calls fine using my analog and IP phones, but when I try to initiate a call using a .call file, I get the following error -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 127 received it happens on certain numbers I dial, but if I dial that same number with an ip or analog phone that use the T1 channel, the call is going through normally. Anybody knows why? Are you doing anything silly with prefixing or short-circuit dialing? in other words.. You dial 8 for an outside line, then 1+10 digits and you're forgetting to do that for some numbers? ___ -- 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] Asterisk talking to mysql
Hi, I have 2 instances of asterisk running and taking to common mysql database via ODBC connection. I am facing some issue while running bulk call (around 100 calls at a time),like few call gets error out of 100, i am suspecting that SQL query is failing and the error i get in one the call failure is .. [*Mar 12 03:29:55] WARNING[27752] chan_sip.c: sip_xmit of 0xb76a27fc (len 446) to 114.105.116.121:26991 returned -1: Address family not supported by protocol* *[Mar 12 03:29:56] WARNING[27752] chan_sip.c: sip_xmit of 0xb76a27fc (len 446) to 114.105.116.121:26991 returned -1: Address family not supported by protocol* *[Mar 12 03:29:58] WARNING[27752] chan_sip.c: sip_xmit of 0xb76a27fc (len 446) to 114.105.116.121:26991 returned -1: Address family not supported by protocol* Anybody faced this issue...? Also what's the significance of rtcachefriends=yes in sip.conf, if i enable this error frequency reduces, but error occurs eventually. -Sumanth ___ -- 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] Controlling BLF Leds ...
Is there a way to set/clear a BLF LED on a phone from the dialplan? I want to use one as an indicator of some state in the PBX - in this case it's night mode but I can think of other applications. I have BLFs working just fine for normal stuff, just wonderin if I can use them for more! Cheers, Gordon ___ -- 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] Controlling BLF Leds ...
Gordon Henderson wrote: Is there a way to set/clear a BLF LED on a phone from the dialplan? I want to use one as an indicator of some state in the PBX - in this case it's night mode but I can think of other applications. I have BLFs working just fine for normal stuff, just wonderin if I can use them for more! Cheers, Gordon I think this is what you want: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate -Dave ___ -- 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] Video phone crashing meetme on asterisk 1.4.
Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe) Options: (17246|d) Video is at 192.168.1.1 port 17280 Audio is at 192.168.1.1 port 14574 Adding codec 0x4 (ulaw) to SDP Adding codec 0x20 (h264) to SDP cain*CLI --- Reliably Transmitting (NAT) to 72.55.182.118:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 72.55.182.118;branch=z9hG4bKf848.70b5f975.0;received=72.55.182.118 Via: SIP/2.0/UDP 75.119.236.164:1036;rport=1036;branch=z9hG4bKd57525869b922403 Record-Route: sip:72.55.182.118;lr=on From: sip:omnitronik.5...@proxy.voip.omnity.biz;tag=9222f0071b85e960 To: sip:8...@proxy.voip.omnity.biz;tag=as07220da7 Call-ID: bd57fc656f213...@192.168.215.9 CSeq: 16492 INVITE User-Agent: OmniVOIP-1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:8...@192.168.1.1 Content-Type: application/sdp Content-Length: 258 v=0 o=root 10487 10487 IN IP4 192.168.1.1 s=session c=IN IP4 192.168.1.1 b=CT:384 t=0 0 m=audio 14574 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 17280 RTP/AVP 99 a=rtpmap:99 H264/9 a=sendrecv [Mar 18 11:55:46] WARNING[13512]: channel.c:3300 ast_request_with_uniqueid: No channel type registered for 'zap' -- Created MeetMe conference 1023 for conference '17246' -- SIP/system.117-b6c0f708 Playing 'conf-onlyperson' (language 'fr') cain*CLI --- SIP read from 72.55.182.118:5060 --- ACK sip:8...@192.168.1.1 SIP/2.0 Record-Route: sip:72.55.182.118;lr=on Via: SIP/2.0/UDP 72.55.182.118;branch=z9hG4bKf848.70b5f975.2 Via: SIP/2.0/UDP 75.119.236.164:1036;rport=1036;branch=z9hG4bK87f6056385b67360 From: sip:omnitronik.5...@proxy.voip.omnity.biz;tag=9222f0071b85e960 To: sip:8...@proxy.voip.omnity.biz;tag=as07220da7 Contact: sip:omnitronik.5...@75.119.236.164:1036 Proxy-Authorization: Digest username=omnitronik.5015, realm=proxy.voip.omnity.biz, algorithm=MD5, uri=sip:8...@proxy.voip.omnity.biz, qop=auth, nc=0001, cnonce=9f83c0053191ebd0, nonce=49c11a7bbb1996940c0ef180cf394bc0f6df19d52a90, response=3e6fdd871c5dc5741f85efdad5ebf3c9 Call-ID: bd57fc656f213...@192.168.215.9 CSeq: 16492 ACK User-Agent: Grandstream GXV3000 1.1.3.29 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 - --- (14 headers 0 lines) --- [Mar 18 11:55:50] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 726 Buffer size: 320 [Mar 18 11:55:50] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 726 [Mar 18 11:55:50] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 930 Buffer size: 320 [Mar 18 11:55:50] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 930 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1291 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 979 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 979 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1172 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1265 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1261 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1191 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1384 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1030 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 993 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 993 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 998 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 998 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1083 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1158 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1279 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 876 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 876 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 875 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320
[asterisk-users] Video phone crashing meetme on asterisk 1.4.
Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. When the video phone calls the extension, it negociates the codecs : Capabilities: us - 0x24 (ulaw|h264), peer - audio=0x380907 (g723|gsm|ulaw|g726|g729|h263|h263p|h264)/video=0x38 (h263|h263p|h264), combined - 0x24 (ulaw|h264) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 75.119.236.164:5004 Peer video RTP is at port 75.119.236.164:5006 Looking for 8500 in from-openser (domain proxy.mytld.biz) list_route: hop: sip:72.55.182.118;lr=on So we see in this SDP packet that Asterisk has negociated ulaw/h264. Now it goes into my extension and calls my AGI script. My AGI script stuff and eventually spits this out : EXEC MEETME 1234|d Asterisk console says : -- AGI Script Executing Application: (MeetMe) Options: (17246|d) Video is at 72.55.182.117 port 17842 Audio is at 72.55.182.117 port 19702 Adding codec 0x4 (ulaw) to SDP Adding codec 0x20 (h264) to SDP Than it sends out a SIP packet to the video phone : Contact: sip:8...@192.168.200.1 Content-Type: application/sdp Content-Length: 258 v=0 o=root 10487 10487 IN IP4 192.168.200.1 s=session c=IN IP4 192.168.200.1 b=CT:384 t=0 0 m=audio 19702 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 17842 RTP/AVP 99 a=rtpmap:99 H264/9 a=sendrecv After that I see : [Mar 18 11:55:50] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 726 Buffer size: 320 [Mar 18 11:55:50] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 726 [Mar 18 11:55:50] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 930 Buffer size: 320 [Mar 18 11:55:50] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 930 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1291 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 979 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 979 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1172 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1265 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1261 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1191 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1384 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1030 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 993 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 993 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 998 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 998 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1083 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1158 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1279 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 876 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 876 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 875 Buffer size: 320 [Mar 18 11:55:51] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 875 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 745 Buffer size: 320 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 745 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1138 Buffer size: 320 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 902 Buffer size: 320 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 320 Buffer size: 902 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1315 Buffer size: 320 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1252 Buffer size: 320 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1097 Buffer size: 320 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1122 Buffer size: 320 [Mar 18 11:55:52] NOTICE[13512]: app_meetme.c:1918 conf_run: Audio bytes: 1188 Buffer size: 320 [Mar 18 11:55:52] NOTICE[13512]:
Re: [asterisk-users] Controlling BLF Leds ...
On Wed, 18 Mar 2009, Dave Fullerton wrote: Gordon Henderson wrote: Is there a way to set/clear a BLF LED on a phone from the dialplan? I want to use one as an indicator of some state in the PBX - in this case it's night mode but I can think of other applications. I have BLFs working just fine for normal stuff, just wonderin if I can use them for more! Cheers, Gordon I think this is what you want: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate Possibly, but: Introduced in Asterisk 1.6 as DEVICE_STATE(), with a backport available for 1.4. Ah well. When I move out of 1.2 I'll give it a go. Cheers, Gordon ___ -- 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] Global h exten
Hi all, Is there something like a global h exten, that gets called on every hang up, no matter what exten? Thanks, Gabriel ___ -- 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] Global h exten
On Wed, 18 Mar 2009, Gabriel Ortiz Lour wrote: Is there something like a global h exten, that gets called on every hang up, no matter what exten? (no matter what context) Nope -- but it sounds like a great idea. I do it this way... I define an h template: [h](!) exten = h,1,goto(finish-call,h,1) And then every context references the template: [block-me](digit-timeout,h,i,max-timeout,pound-main,s) 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] Global h exten
On Wed, Mar 18, 2009 at 11:57 AM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 18 Mar 2009, Gabriel Ortiz Lour wrote: Is there something like a global h exten, that gets called on every hang up, no matter what exten? (no matter what context) Nope -- but it sounds like a great idea. I do it this way... I define an h template: [h](!) exten = h,1,goto(finish-call,h,1) And then every context references the template: [block-me](digit-timeout,h,i,max-timeout,pound-main,s) That's an elegant way to do it, another in AEL, would be to define a context with an h-exten, and include it in your other contexts... Just beware, that the h-exten is NOT ALWAYS called; the in the channel/peer role world, the h-exten is usually called on just the channel in the channel role. The parking manager doesn't run the h-exten if a channel hangs up while parked. And channel and peer roles can sometimes get a bit confused in transfer scenarios. The truth of each sentence above will surely change with new releases... murf -- Steve Murphy ParseTree Corp ___ -- 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] Good phone near $125
On Mon, Mar 16, 2009 at 20:26, Marc Charbonneau timebandit...@gmail.com wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet, support PoE and works with 2.5mm headset. $110 at voipsupply Or the Polycom 320 -- same phone as the 330, both have PoE support, 320 has 1 Ethernet port, 330 has two ports (built in switch) Now that I have to reply to your message, may I suggest telephonydepot.com. They have the 330 for $106 and the 320 for $83. FWIW VoIP supply are horrible (and overpriced.) They took 6 months to RMA a phone, and even then they didn't do what I requested. ___ -- 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] Good phone near $125
I concur on TelephonyDepot.com. I really like the Grandstream GXP2000 @ about $95, and the Budgetone 200 at about $48 bucks (No POE on the 200) but dual Ethernet ports. Personally I stopped using the SNOM360 and use the GXP2000 with a headset. Both of those Grandstreams support 2.5 mm headset ports. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Joakimsen Sent: Wednesday, March 18, 2009 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Good phone near $125 On Mon, Mar 16, 2009 at 20:26, Marc Charbonneau timebandit...@gmail.com wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet, support PoE and works with 2.5mm headset. $110 at voipsupply Or the Polycom 320 -- same phone as the 330, both have PoE support, 320 has 1 Ethernet port, 330 has two ports (built in switch) Now that I have to reply to your message, may I suggest telephonydepot.com. They have the 330 for $106 and the 320 for $83. FWIW VoIP supply are horrible (and overpriced.) They took 6 months to RMA a phone, and even then they didn't do what I requested. ___ -- 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] Good phone near $125
I've worked with VoIP Supply several times in the past. I've been very pleased with their service. And if you compare the prices of the two phones you mention: Polycom IP 320 330 a difference of 109.94 vs 106 and 84.95 vs 83 seems to disperse the allegation of being overpriced. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Joakimsen Sent: Wednesday, March 18, 2009 3:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Good phone near $125 On Mon, Mar 16, 2009 at 20:26, Marc Charbonneau timebandit...@gmail.com wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet, support PoE and works with 2.5mm headset. $110 at voipsupply Or the Polycom 320 -- same phone as the 330, both have PoE support, 320 has 1 Ethernet port, 330 has two ports (built in switch) Now that I have to reply to your message, may I suggest telephonydepot.com. They have the 330 for $106 and the 320 for $83. FWIW VoIP supply are horrible (and overpriced.) They took 6 months to RMA a phone, and even then they didn't do what I requested. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: Re: DAHDI or Zaptel doesn't compile against 1.4.24]
Tzafrir Cohen a écrit : On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote: Hi, We installed the latest 1.4.24 on a test machine and can't get zaptel nor dahdi compile. It's a Linux Debian Etch. Errors we have: keewi:/usr/src/dahdi-linux-2.1.0.4# make make -C /lib/modules/2.6.18-custom.2/build ARCH=i386 SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.1.0.4/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 » WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers is missing; modules will have no dependencies and modversions. CC [M] /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.o In file included from /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi-base.c:38: /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_config.h:23:27: error: linux/version.h: Aucun fichier ou répertoire de ce type This is plain wrong. Your source tree is bad. What kernel version do you want to build dahdi against? What kernel version do you use? uname -a |d...@keewi:/usr/src/dahdi-linux-2.1.0.4$ uname -a |Linux keewi 2.6.18-custom.2 #1 Sat Nov 29 22:07:56 CET 2008 i686 |GNU/Linux | |Well, it seems that something is wrong with my kernel tree. I will |build |a new kernel from sources. That was it. Now everything is fine. Thanks to you and to John have point the direction to look for. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to receive faxes
Hi,
I'm experiencing a quite strange behavior while trying to receive faxes
through Asterisk (either directly through app_rxfax or with spandsp +
hylafax).
Config:
HFC quad BRI card (3 T0 connected to the card)
Asterisk 1.4.21
asterisk-app-fax 0.0.20070624-2
hylafax 2:4.4.4-10.1
libpri 1.4.2
libspandsp3 0.0.4pre16
/etc/zaptel.conf
loadzone=fr
defaultzone=fr
span=1,0,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
/etc/asterisk/zapata.conf
[channels]
language=fr
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = unknown
prilocaldialplan = dynamic
nationalprefix = 0
internationalprefix = 00
echocancel=yes
rxgain=1.1
txgain=1.1
group = 1
context=zaptel-in
channel = 1-2
channel = 4-5
channel = 7-8
/etc/asterisk/extensions.conf:
exten = 0258,1,Answer()
exten = 0258,n,Set(FAXFILE=/var/spool/asterisk/fax/fax-0258.tif)
exten = 0258,n,rxfax(${FAXFILE})
exten = 0258,n,NoOp()
exten = 0258,n,Hangup
Here is the asterisk cli output:
-- Accepting voice call from '' to '0258' on channel 0/1, span 1
-- Executing [0...@zaptel-in:1] NoOp(Zap/1-1, ) in new stack
-- Executing [0...@zaptel-in:2] Set(Zap/1-1, CALLERID(num)=) in
new stack
-- Executing [0...@zaptel-in:3] Goto(Zap/1-1,
default|0491140258|1) in new stack
-- Goto (default,0258,1)ff o
-- Executing [0...@default:1] Goto(Zap/1-1, 0258|1) in new stack
-- Goto (default,0258,1)
-- Executing [0...@default:1] Answer(Zap/1-1, ) in new stack
-- Executing [0...@default:2] Set(Zap/1-1,
FAXFILE=/var/spool/asterisk/fax/fax-0258.tif) in new stack
-- Executing [0...@default:3] RxFAX(Zap/1-1,
/var/spool/asterisk/fax/fax-0258.tif) in new stack
-- Channel 0/1, span 1 got hangup request, cause 16
-- Hungup 'Zap/1-1' channel
The resulting tiff file is nearly empty.
-rw-r--r-- 1 asterisk asterisk 8 2009-03-18 22:31
/var/spool/asterisk/fax/fax-0258.tif
Do anyone have a clue about this issue ?
What is cause 16 about ?
Thanks
Laurent
___
-- 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] Voicemail config help - require password
How do you require a password for a voicemail box? I have been searching all day, and can't find any type of security setting for voicemail. I am looking for some what to have some minimum security like no blanks, can't be the same as the extension, can't be sequential numbers or repeated numbers. I know that not all of these options may exist, but there has to be a way to require a password? If not, does anyone have an external script that does some of the above before I write one? Also, is there a way to retain deleted messages for a length of time before they are purged? We currently have that feature on our production VM server that I am trying to replicate. Thanks! -Jonathan ___ -- 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] Recent changes in chan_mobile need testing!
Greetings chan_mobile users, I have just merged my refactor of chan_mobile into asterisk-addons trunk and now the code needs testing. The changes I have made should improve the stability and reliability of the code and should also improve audio quality. Error reporting should be improved as well. I have tested most of the code with my blackberry device, but I need more people to test the code with different devices. Specifically, I was unable to test SMS support, so anyone able to test that functionality would be greatly appreciated. If you can, please test the latest version of chan_mobile (rev 815 of asterisk-addons/trunk) and report any problems you find to the bug tracker. Thanks. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] work around the 64 pickupgroups limit
On 17/03/2009 9:10 a.m., Doug wrote:
This looks great! A few questions...
in the standard extension macro we add a line:
Is this in extensions.conf?
Yeah, we have a macro (which is in the default extensions.conf) which we
add that line to.
exten = s,n,Set(_PICKUPMARK=${DB(pickupgroup/${ARG1})})
Where ARG1 is the extension about to be called (i.e. 201)
When someone dials 29 to pickup:
exten = 29,1,Pickup(${DB(pickupgroup/${CALLERID(number)})}...@pickupmark)
Would this also be in extensions.conf?
Yep, we have a context which we put applications in that users can use
(i.e. voicemail, echo test, pickup etc - this is added there)
So to make extension 201 in pickup group 1 just do:
asterisk -rx 'database put pickupgroup 201 1'
So this is a command line argument. Can this
be automated? Whenever we do a reload, can
this be stored?
The database lasts across restarts - so once you put it in, it is in.
You can do it from the commandline, Asterisk Manager, or even from
extensions.conf. For example you could make it that if someone dials
1234x, their pickup group is set to x.
I.E.
exten = _1234x,1,Set(${DB(pickupgroup/${CALLERID(number)})}=${EXTEN:4})
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
___
-- 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] 428 Loop Detected
It's so uncommon for me fxs and fxo cards and based on the reference of sip.conf files and accounts i totally missed last paragraph where it was mentioned only analogue lines and fxs (phone). my appologies. E1 and BRIs and sip trunks have been overloading my last month of work. cheers, -- Marco Mouta 2009/3/16 Steve Totaro stot...@totarotechnologies.com: Again, if I am interpreting this correctly, he is not using SIP. A four port card 2fxo/2fxs means to me that he is not using SIP at all. If by card, you mean some kind of SIP gateway, then I misunderstood and the problem, but seeing DAHDI channels leads me to believe that SIP is not required and actually causing your problems. SIP is a protocol for VoIP, DAHDI/Zaptel is TDM (analog POTS in this case)... If you had a SIP device, it would be connected to the data network, not a phone line. Can you just plug your phone into a regular landline jack and get dialtone? If so, forget SIP for now. Comment out or delete all your sip.conf peers since you are not using SIP. Change your dialplan to not (Dial/SIP but (Dial/DAHDI/1,10) and the correct channel to your FXS port that the phone is connected to. Thanks, Steve Totaro On Sun, Mar 15, 2009 at 9:20 PM, Marco Mouta marco.mo...@gmail.com wrote: Hi, problem is that you are saying that phone in sip.conf is at the same ip address of your asterisk box so you are dialing into a loop to your self asterisk box [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 what you need is: [phone] type=friend context=phone1 secret=g00dpazzwerd dtmfmode=rfc2833 host=dynamic ;configuring your codecs (i don't know what else you have configured, just preventing audio for you) disallow=all allow=ulaw allow=alaw allow=gsm Dial sip/phone is enough too.. [from-pstn] ;include = default exten = s,1,Dial(SIP/phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup hope it helps. don't forget to asterisk reload on cli. Looking forward to hearing from you. cheers -- Marco Mouta On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-internal default 2 from-internal default 3 from-pstn default 4 from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Voicemail config help - require password
On Wednesday 18 March 2009 17:02:33 Jonathan Thurman wrote: How do you require a password for a voicemail box? I have been searching all day, and can't find any type of security setting for voicemail. I am looking for some what to have some minimum security like no blanks, can't be the same as the extension, can't be sequential numbers or repeated numbers. I know that not all of these options may exist, but there has to be a way to require a password? If not, does anyone have an external script that does some of the above before I write one? Also, is there a way to retain deleted messages for a length of time before they are purged? We currently have that feature on our production VM server that I am trying to replicate. Thanks! Starting in 1.6.2, yes. The setting within voicemail.conf that controls this is externpasscheck. This setting controls the name of a script used for password validation. The script should output the value 'VALID', plus any additional details you want in the log if the password meets the specified rules and 'FAILURE' (followed by any additional details) otherwise. -- 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] Manager API Originate CDR Problem, all is NO ANSWER
On 18/03/2009 9:58 p.m., MaxGao wrote: hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten = s,1,Answer() exten = s,n,Wait(10) exten = s,n,Hangup() when the phone 888 pick up , it will come to callout context, after hangup, one cdr generate, but the cdr disposition is still NO ANSWER ... and the billsec is 0, Although i can use duration instead, is this a bug ? or how to set it right?? Do you get two CDR entries for that call? Any errors? Any uniqueid on CDR? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Voicemail config help - require password
On 19/03/2009, Jonathan Thurman jthurma...@gmail.com wrote:
Also, is there a way to retain deleted messages for a length of time
before they are purged? We currently have that feature on our
production VM server that I am trying to replicate. Thanks!
Could this be done with a simple nightly cron job? Something like
find /var/lib/asterisk/messages [I forget the path] -name *.wav -mtime
+90 -exec rm {} \;
That'll delete any WAV files older than 90 days. Mind you, there might
be an index file that needs editing as well (haven't used the VM
system for a while) so a script may be needed...
Andrew
--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
-- Bill Garrett
___
-- 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] queued?
Any idea what this means? And why they are different? - Extension Changed 22142[default] new state Idle for Notify User 31001 (queued) Extension Changed 22142[default] new state Idle for Notify User 30060 - I have googled and searched, and can't find anything on this subject. Does anyone have an suggestions? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 problem (call using a .call file)
This has to be a bug, because I dont know what else to try here On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno tipas...@gmail.com wrote: Nope, I always dial 1 + 10 digits for all my numbers. It works on all numbers when I am using my phone (Analogue or IP) but when I do it using a .call file it does not work on some numbers mostly. That is the weirdest thing I have ever seen. I tried different codecs in the call file, I still get the PROGRESS with cause code 127 On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg dbackeb...@gmail.comwrote: On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno tipas...@gmail.com wrote: I have a weird problem with call using my T1 card. I can make calls fine using my analog and IP phones, but when I try to initiate a call using a .call file, I get the following error -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 127 received it happens on certain numbers I dial, but if I dial that same number with an ip or analog phone that use the T1 channel, the call is going through normally. Anybody knows why? Are you doing anything silly with prefixing or short-circuit dialing? in other words.. You dial 8 for an outside line, then 1+10 digits and you're forgetting to do that for some numbers? ___ -- 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] Manager API Originate CDR Problem, all is NO ANSWER
在2009-03-19?06:53:56，Matt?Riddell?li...@venturevoip.com?写道： On?18/03/2009?9:58?p.m.,?MaxGao?wrote: ?hi,?all ??asterisk?1.4.24?,?zaptel?1.4.10.1?,?E1 ??Manager?API?Action?: ?Action:?Originate ?Channel:?ZAP/G1/888 ?Callerid:?12345678 ?Context:?callout ?Exten:?s ?Priority:?1 ?extensions.conf ?[callout] ?exten?=??s,1,Answer() ?exten?=??s,n,Wait(10) ?exten?=??s,n,Hangup() ?when?the??phone??888??pick?up?,?it?will?come?to?callout?context,?after?hangup,?one?cdr?generate,?but?the?cdr?disposition?is?still?NO?ANSWER?...?and?the?billsec?is?0,?Although?i?can?use?duration?instead,?is?this?a?bug???or?how?to?set?it?right?? Do?you?get?two?CDR?entries?for?that?call???Any?errors???Any?uniqueid?on?CDR? i get one CDR , that's right , but the disposition? field of cdr is NO?ANSWER , the callee and the caller both answer the call , why the cdr is still NO?ANSWER ? and the billsec field is still zero thanks a lot. --? Kind?Regards, Matt?Riddell Director ___ http://www.venturevoip.com?(Great?new?VoIP?end?to?end?solution) http://www.venturevoip.com/news.php?(Daily?Asterisk?News?-?html) http://www.venturevoip.com/newrssfeed.php?(Daily?Asterisk?News?-?rss) ___ -- 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] Manager API Originate CDR Problem, all is NO ANSWER
On 19/03/2009 2:17 p.m., MaxGao wrote: ??2009-03-19?06:53:56??Matt?Riddell?li...@venturevoip.com??? On?18/03/2009?9:58?p.m.,?MaxGao?wrote: ?hi,?all ??asterisk?1.4.24?,?zaptel?1.4.10.1?,?E1 ??Manager?API?Action?: ?Action:?Originate ?Channel:?ZAP/G1/888 ?Callerid:?12345678 ?Context:?callout ?Exten:?s ?Priority:?1 ?extensions.conf ?[callout] ?exten?=??s,1,Answer() ?exten?=??s,n,Wait(10) ?exten?=??s,n,Hangup() ?when?the??phone??888??pick?up?,?it?will?come?to?callout?context,?after?hangup,?one?cdr?generate,?but?the?cdr?disposition?is?still?NO?ANSWER?...?and?the?billsec?is?0,?Although?i?can?use?duration?instead,?is?this?a?bug???or?how?to?set?it?right?? Do?you?get?two?CDR?entries?for?that?call???Any?errors???Any?uniqueid?on?CDR? i get one CDR , that's right , but the disposition? field of cdr is NO?ANSWER , the callee and the caller both answer the call , why the cdr is still NO?ANSWER ? and the billsec field is still zero thanks a lot. You need to change your email client to send plain text emails. I'm sure you can see the mess this has made. So you have debugging turned on in logger.conf, and reloaded Asterisk and have no errors writing to CDR? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Manager API Originate CDR Problem, all is NO ANSWER
oh, i am sorry, plain text : hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten = s,1,Answer() exten = s,n,Wait(10) exten = s,n,Hangup() when the phone 888 pick up , it will come to callout context, after hangup, one cdr generate, but the cdr disposition is still NO ANSWER ... and the billsec is 0, Although i can use duration instead, is this a bug ? or how to set it right?? Do you get two CDR entries for that call? Any errors? Any uniqueid on CDR? i get one CDR , that's right , but the disposition field of cdr is NO ANSWER , the callee and the caller both answer the call , why the cdr is still NO ANSWER ? and the billsec field is still zero thanks a lot. ___ -- 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] what is the effect of high LBO settings?
On Mon, Mar 2, 2009 at 3:07 PM, Brandon B. bran...@brellsystems.com wrote: On Fri, Feb 27, 2009 at 7:49 PM, Jared Smith jsm...@digium.com wrote: As I understand it, the LBO is effectively an attenuation value, with a higher number meaning less attenuation. This way, you don't get too hot of a signal with a short cable, or two low of a signal on long cable. Just how far is your Asterisk box from the demarcation point? This system is connected to a CSU in the same room that provides the physical T1 line. I've always set the LBO setting at 0 for this because I've never had a long line to deal with. Since 0 works for me, I'm going to assume it's the correct setting with the demarc point (i.e. the Paradyne CSU) in the same room -- right? It's slightly confusing with settings 5,6,7 labelled CSU and no description as to when to use those levels. Could you provide any suggestion for when levels 5,6,7 would be appropriate? From what you say an LBO setting of 5 would boost the signal level, which could be hot. Is there any chance this would cause the card to fail after a while? It appears this site just had 4 port Digium card fail today. Turns out this problem was not the card, but another hardware issue. The hard disk with Reiserfs eventually cratered and took the system down, and the entire filesystem appears to be unrecoverable. I've changed the LBO settings from 5 to 0, and the system is working fine. If anyone has anything to add regarding the effect of different LBO settings I think it might be helpful to have this documented. Brandon B. ___ -- 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] Voicemail config help - require password
On Wed, Mar 18, 2009 at 4:18 PM, Andrew Furey andrew.fu...@gmail.com wrote:
On 19/03/2009, Jonathan Thurman jthurma...@gmail.com wrote:
Also, is there a way to retain deleted messages for a length of time
before they are purged? We currently have that feature on our
production VM server that I am trying to replicate. Thanks!
Could this be done with a simple nightly cron job? Something like
find /var/lib/asterisk/messages [I forget the path] -name *.wav -mtime
+90 -exec rm {} \;
That'll delete any WAV files older than 90 days. Mind you, there might
be an index file that needs editing as well (haven't used the VM
system for a while) so a script may be needed...
Andrew
The cron job is a great idea for purging the files. However, what I
am looking for is when a user deletes the message, it gets moved into
a Delete items directory for lack of better terms. So the user has
X number of days before the message is purged to undelete it through
the voicemail menu. I am using the plain file system voicemail. Is
something like this supported with any of the voicemail modules?
-Jonathan
___
-- 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] AstLinux 0.6.4 available for upgrade
The AstLinux Team is happy to announce that AstLinux 0.6.4 is available. All users of AstLinux are encouraged to upgrade since this release fixes the recently reported security vulnerability in Asterisk 1.4.23.1 Right now a mix up on the Sourceforge site is preventing us from uploading full install versions, but current users of 0.6.2 or 0.6.3 can upgrade to 0.6.4 by using either the 'upgrade-run-image' script from the command line or the upgrade firmware option in the web interface. New versions of the full install will be available as soon as possible on the Sourceforge site. Changes: Asterisk 1.4.24 is included which fixes several bugs and at least one security issue Asterisk-gui was updated to svn 4618 netsnmp was updated to 5.3.2.3 The web interface was upgraded to add several features/improvements An arno-upgrade-firewall script was added to break this away from an init change. This won't really affect users of 0.6.x until they move to 0.7.x which uses a newer version of Arno's firewall. When the time comes, we'll explain the importance. A serial number file was added to trace the version of the firewall config files. To upgrade from the command line: 1). upgrade-run-image check http://mirror.astlinux.org/firmware 2). upgrade-run-image upgrade http://mirror.astlinux.org/firmware 3). reboot as instructed To upgrade from the web interface: 1). Navigate to the system tab on the web interface 2). Select check for new, select the confirm box, and click the Firmware button. 3). Select upgrade with new, select the confirm box, and click the Firmware button. 4). Reboot by checking that confirm button and clicking Reboot. Enjoy The AstLinux Team ___ -- 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 crashed!!!
Hi All, I have a working asterisk 1.4.23.1 on server. OS: Centos 5.2 Suddenly asterisk has stopped to process calls crashed. I found that asterisk has generated coredumps. I have restarted asterisk it started to work as expected without any issue. Would you please help me out to troubleshoot the cause of crash? Please checkout following link, I have uploaded coredump backtraces there. http://pastebin.com/m5480bcb8 Please provide me help regarding this. Thanks in advance. Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
2009/3/17 Marc Charbonneau timebandit...@gmail.com I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet, support PoE and works with 2.5mm headset. $110 at voipsupply While the Polycom IP-330s do work reasonably well, my first impressions with them was of a cheap light plasticy handset, while that might be a preference issue, all the other IP phones I've used (Cisco, Linksys and Avaya) had more substantial handsets that I found more comfortable to use personally and were preferred by the user base... That said, the Polycom's do the job and for their price, you get a decent amount of value... I personally probably wouldn't choose them for myself as the handset is an issue for me, but, in a reasonable size deployment where cost is a factor, they aren't bad phones... 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
