Re: [Asterisk-Users] equivalent to SetvarIf ?
On 11/21/05, Wilson Pickett [EMAIL PROTECTED] wrote:
Is there a syntax I can use to set a variable based on the evaluation
of an expression? I need something that will work in 1.0.9 and 1.2.
Isn't this what you're looking for:
set(VARIABLE=$[NULL${something}=NULL]})
I'm not quite sure I understand that. However using a regex works.
But I'm getting an error that I halfway understand and don't know how
to fix.
Set(something=800111)
This works:
Set(var2=$[${something} : ([1-9])])
This doesn't, giving me an 'invalid repetition count(s)' error:
Set(var2=$[${something} : ([1-9]{2,10})])
Anyone know what's wrong with my syntax?
Chris
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Re: [Asterisk-Users] Realtime Problems
scott wrote: Hi Thank you for your reply. I have tried various definitions in the sipusers table but none seem to be working :-( I have attached mey structure and content export below for your attention. You should have a look at this page : http://www.asteriskguru.com/tutorials/realtime_pgsql.html. -- Benoit Merouze Network Software Developer at IPercom [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recall button using tdm400 Australia
Paul == Paul Liew [EMAIL PROTECTED] writes: Paul You are correct - rxflash and flash in zapata does the Paul equivalent, but I should also have said in my earlier post Paul that you need to drop the max pulse time (for pulse Paul dialling) to be less than the hook flash timing. Default Paul settings for max pulse is 150ms, which inteferes with Paul Australian hook flash of 100ms. - It does work, as it is Paul running in our setup here. Arhhh... That makes sense. I suspect you think it is misinterpreting the flash as a pulse used in pulse dialing. Later: I set pulsedial=no in zapata.conf, it doesn't help. That leaves: Paul You need to set rxflash and flash as max and min times for Paul the hookflash to work. I am sorry, you lost me here? You mean set rxflash to the max and flash to the min time? What times should I use? Currently I have: pulsedial=no flash=100 rxflash=100 -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Select multiple columns from MYSQL cmd...
In article [EMAIL PROTECTED],
Ben Higley [EMAIL PROTECTED] wrote:
I have read on the wiki the many howto's to select data using the MYSQL
command. I would like to select multiple columns from a table using the
MYSQL command, however, it will only fetch one at a time.
You just need to provide multiple variables in the Fetch command to
receive the columns.
I have tried the code to select using the GOTO(3) - (refereneced in the
wiki) - to fetch if more data, however, i would have to keep track of a
counter, and if the counter is now =2, then that column variable needs to
be set with the value that came out of the database.
Does someone have some code that does this process? Or are you all using
an AGI script?
I'm not familiar with the wiki example, but here is an extract from the
extensions.conf of one of my systems that illustrates the technique, by
fetching each inserted record again to write to a backup file:
exten = h,1,MYSQL(Connect conn localhost username password database)
exten = h,2,MYSQL(Query res ${conn} 'INSERT INTO calls(callerid,calltime,ddi)
VALUES(\'${CALLERIDNUM}\',NOW(),\'${DDI}\')')
exten = h,3,MYSQL(Query res ${conn} 'SELECT call_id,callerid,calltime,ddi FROM
calls WHERE call_id=LAST_INSERT_ID()')
exten = h,4,MYSQL(Fetch fid ${res} call_id callerid calltime ddi)
exten = h,5,MYSQL(Clear ${res})
exten = h,6,MYSQL(Disconnect ${conn})
exten = h,7,System(/bin/echo ${call_id}','${callerid}','${calltime}','${ddi}
/tmp/calls.csv)
Hope this helps!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Re: problem with registration of SIP phone
I managed to isolate the problem a bit more, maybe it will help to find a solution:The problem with the phones is not the initial registration, but the re-registration process.When I create a new extension the phone registers ok, but when the same phone tries to re-register it fails. On 11/22/05, Asterisk User [EMAIL PROTECTED] wrote: I'm runing [EMAIL PROTECTED] beta6 and I have a I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart or after long idle time the phones can't register and I get this in the server:Transmitting (no NAT) to 10.1.1.152:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.1.152;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152From: sip: [EMAIL PROTECTED];tag=12e8dd0080754148To: sip: [EMAIL PROTECTED];tag=as2383b1df Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip: [EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=54cb5290 Content-Length: 0After a few server restarts and/or phone restarts the phone registers ok. Any ideas why ?Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech recognition or TTS with Asterisk?
On 11/18/05 12:55 John Todd said the following: affordable, which probably means $50 or less I suspect. This would be a native Linux environment for all components. Again, while I have no when, oh when, will folk like these support use downtrodden freebsd folk ? :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outbound sip proxy
Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] channel_find_locked
I've been playing around with AgentCallbackLogin, etc. Now I get this message --- Nov 22 17:31:45 WARNING[1889]: channel.c:784 channel_find_locked: Avoided initial deadlock for '0x 8135b00', 10 retries! --- whenever a user tries to dial into the system. Restarting asterisk and even rebooting doesn't help. I reduced extensions.conf, agents.conf queues.conf to the absolute minimum with no effect. show channels prints: --- Channel Location State Application(Data) 0 active channels 0 active calls --- How can I get rid of this message? Greetings, Marcus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Problems
The host column must contain 'dynamic' not your IP. UPDATE sip_users SET host = 'dynamic' WHERE name = '114'; *CLI sip show peers Name/username Host Dyn Nat ACL Port Status 114/114 80.xxx.xxx.xxx D 5060 Unmonitored 1 sip peers [1 online , 0 offline] Just try it. Interested in Open Source Asterisk Realtime. The best examples and real life installations can be found at: http://astbill.com-- Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultants http://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIPAstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Menu Tree Delay
I have a fairly simple menu structure, three options branch to submenus. There is a long (several seconds) delay between pressing a key and getting the next menu. This happens on 2 out of 3 of my menus for no apparent reason. I am kind of at a loss as to what to look at. Any suggestions would be appreciated. I am using Asterisk 1.2, CentOS 4.2. 2.6ghz machine with 1gb of RAM. -Kerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP - ! No D-channels available!
We've had no problems for a few weeks running Asterisk CVS-D2005.10.28.07.54. However, this morning, we're getting users complaining that they were cut off - and I found these in the logs. This has happened 5 times this morning, and there is an entry in the log at the appropriate time. Is this a BT issue ? EuroISDN with a TE405P. Julian. Nov 22 09:53:29 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:53:31 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:53:37 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:53:37 WARNING[8519] app_dial.c: Unable to forward voice Nov 22 09:53:37 WARNING[8505] app_dial.c: Unable to forward voice Nov 22 09:54:34 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:54:34 WARNING[8678] app_dial.c: Unable to forward voice Nov 22 09:54:37 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:54:38 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk not picking up calls.
On Mon, 2005-11-21 at 21:48 +, Mark Ackroyd wrote: All, I thought I'd post the answer to this, After I found what the problem was. It was the cable from the TDM card to the phone socket. I used one that came with an old modem and it worked a charm :-) I've had that problem very often here in France with various kit, I was testing a Clipcomm CG-410, it exhibited all sorts of strange behaviour until I changed the connecting cable. It reminded me of Animal Farm - two wires good four wires bad -. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PAP2 and double ringback tone
Hi, I have a problem with double ringback tone - outgoing connections to PSTN. I do not use 'r' option in Dial function so I expect to hear 'real' sounds from pstn provider. But PAP2 generates extra ringback tone itself! How to get rid of that? Regards, L ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: oh323 channel disappears
First of all, thank you for your answer, the only that does not claim to
not restart the box !
Asterisk is the last stable version via cvs, not cvs head
show version:
Asterisk CVS-v1-0-10/31/05-17:43:16 built by [EMAIL PROTECTED] on a i686
running Linux
So it was the last stable version on 31 of October;
Also other components were taken via CVS;
cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds
about oh323, these are the instructions that I assembled and followed,
reading around;
cd /root
wget
http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/pwlib-Mimas_patch2-src-tar.gz
wget
http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/openh323-Mimas_patch2-src-tar.gz
cd /usr/src
wget
http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/download/asterisk-oh323-0.6.7.tar.gz
cd /root
tar zxvf pwlib-Mimas_patch2-src-tar.gz
tar zxvf openh323-Mimas_patch2-src-tar.gz
mv pwlib_Mimas_patch2 pwlib
mv openh323_Mimas_patch2 openh323
cd /usr/src
tar zxvf asterisk-oh323-0.6.7.tar.gz
PWLIBDIR=/root/pwlib
export PWLIBDIR
OPENH323DIR=/root/openh323
export OPENH323DIR
LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib
export LD_LIBRARY_PATH
modify the file:
vi /etc/ld.so.conf
and add in it::
/root/pwlib/lib
/root/openh323/lib
then:
ldconfig
cd /root/pwlib
./configure make clean make opt make install ldconfig
cd /root/openh323
./configure make clean make opt make install ldconfig
cd /usr/src/asterisk-oh323-0.6.7
modify Makefile according to the directories:
vi /usr/src/asterisk-oh323-0.6.7/Makefile
PWLIBDIR=/root/pwlib
OPENH323DIR=/root/openh323
make make install ldconfig
chown /usr/lib/asterisk/modules/asterisk . -R
chgrp /usr/lib/asterisk/modules/asterisk . -R
chown asterisk /usr/local/lib -R
chgrp asterisk /usr/local/lib -R
chmod 777 /root
chown asterisk /root/pwlib -R
chgrp asterisk /root/pwlib -R
chown asterisk /root/openh323 -R
chgrp asterisk /root/openh323 -R
the only thing I am absolutely not hayy to did was that chmod 777 /root;
I think that it should be not necessary at all, I did it becouse asterisk
run as asterisk user, and peraphs i thought some problems aboutr
accessing pwlib or oh323;
I have an heavily stressed system, but I have a couple of hours of almost
no traffic (people sleep sometimes...)
To shut down asterisk, killing a maximum 1 or 2 phones and than reboot (
only restart gracefully or now is not sufficient to re-live the oh323
channel)
is a bad thing, but is better than drop 5,000 phones 5 hours later.
Why not only reboot ? becouse if you shurdown asterisk BEFORE rebooting,
the cdr is updated correctly with the last phnes running.
I tried to reboot a box WITHOUT exiting from asterisk, and the running
conversetion (with more then 2000 billsec) was not recorded in the cdr
I am using the g729 codec ( I bought a 30 channels license from Digium).
So, what to say... ah, you also need my oh323,conf file: here it is.
asterisk02:/etc/asterisk # cat oh323.conf
;
; Configuration file of OpenH323 channel driver
;
;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=1
tcpEnd=2
;
; Configure the UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
; rtp.conf
;
udpStart=1
udpEnd=2
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
; lowdelay, throughput, reliability, mincost, none
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=100
inboundMax=100
simultaneousMax=100
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=2
;crlThreshold=30
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only the trace info for OpenH323 is logged in libTraceFile.
;
[Asterisk-Users] Codec that quality does not get affect *much* against packet loss
I think I have heard in the past that someone mentioned to me there is a codec that does not getting affected much because of packet loss. Is there such thing? Sam ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_icd anyone? on 1.2?
I also have never found anybody running an Asterisk system using app_icd. Maybe app_queue is now after all flexible enough to be used in most cases. Anybody else using different apps for Asterisk call centre applications? l. On Mon, 21 Nov 2005 20:30:33 +0100, Waldo Rubinstein [EMAIL PROTECTED] wrote: I've asked the same question in several occasions in the past and never received a response. I figured this project was dead and stop pursuing using it. - Waldo On Nov 21, 2005, at 12:17 PM, Lenz wrote: Well, this is interesting - is anybody actually using app_icd out there? :-) l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone parked in your Asterisk?
this is very welcome as i need to keep track of agent status using the SNOM BLF Alexander Lopez wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Olle E Johansson Sent: Monday, November 21, 2005 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone parked in your Asterisk? Alexander Lopez wrote: Does it hold state information for any channel? Even ZAP, IAX, etc!!! If it does, Olle, you have just placed us one step closer to being able to emulate a Key system!!! This fix is very focused on parking. Previous to this fix, we can check device status in chan_agent, chan_iax2 and chan_sip. show channeltypes tell you which channels in your Asterisk that support device status notification. The ability to visually see parking lots has been asked for, and I created this by adding device status notification in chan_local (does an extension exist or not in the active dialplan?) and a notification system in res_features whenever parking adds or removes an extension - i.e. parks a call. It is a strange form of abstraction, but it works :-) /O ___ Since the state gets updated every time an event happens. Event being placing or receiving a call. Would it be a good idea to add this to the IsChanAvail application (such as IsExtenAvail)?? That would give us a channel independent application to update the notification system. It would put more work into the dialplan but it could be handled by a macro for the lazy... Or am I just not understanding the notification idea. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.0 AddOn's compile error with MySQL 5.0.15
On Monday 21 November 2005 23:49, Rainer Maier wrote: Hi all, I want to compile asterisk's newest version with mysql's newest version, but I ran into a big problem. At compile time for asterisk-addons-1.2.0 I get the following errors: make -- snip -- cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -I/usr/local/mysql/include/mysql -c -o res_config_mysql.o If you didn't do mv /usr/src/asterisk-1.2.0 /usr/src/asterisk that might be your problem. But cannot be seen because of your --snip--. Further down I don't see either CFLAGS+=-I../asterisk-1.2.0 which the other way around. Hope that helps. benchev res_config_mysql.c res_config_mysql.c: In function 'realtime_mysql': res_config_mysql.c:117: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'realtime_multi_mysql': res_config_mysql.c:224: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'update_mysql': res_config_mysql.c:313: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'config_mysql': res_config_mysql.c:376: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c: In function 'realtime_mysql_status': res_config_mysql.c:648: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:650: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:652: warning: incompatible implicit declaration of built-in function 'snprintf' res_config_mysql.c:656: warning: incompatible implicit declaration of built-in function 'snprintf' cc -shared -Xlinker -x -o res_config_mysql.so res_config_mysql.o -lmysqlclient -lz-L/usr/local/mysql/lib -L/usr/local/mysql/lib/mysql sv5000:/usr/src/asterisk-addons-1.2.0# Now the details: I wanted to set up a plain asterisk computer without any more programms. I set up a plain debian sarge system and installed kernel 2.6.14.2. I downloaded, unpacked mysql-5.0.15 under /usr/src/mysql-5.0.15. Then I put the link /usr/src/mysql to this directory. I compiled and installed mysql successfully. I then downloaded asterisk-1.2.0.tar.gz and unpacked it to /usr/src/asterisk-1.2.0 I compiled and installed it successfully with make, make install and make-samples. I then downloaded asterisk-addons-1.2.0.tar.gz and unpacked it to /usr/src/asterisk-addons-1.2.0 I tried to compile and had the problem that the compiler did not find the mysql includes an libs. I had to modify Makefile first. First I added this directory to the MODS, CFLAGS and MLFLAGS. It would be nice to have them in the next update. Afterwards the compiler stopped with the above error's. Is there a new 'snprintf' version used ? Do you have a solution for that ? At the end are the compiler etc. versions. Makefile at /usr/src/asterisk-addons-1.2.0 --- - --- V MODS+=$(shell if [ -d /usr/local/mysql/include ] || [ -d MODS+MODS+/usr/local/mysql/include/mysql ] || [ -d /usr/include/mysql ] MODS+|| [MODS+-d /usr/local/include/mysql ] || [ -d MODS+/usr/local/mysql/include ] || [ -d /opt/mysql/include ]; then echo MODS+cdr_addon_mysql.so app_addon_sql_m ysql.so res_config_mysql.so; MODS+fi) CFLAGS+=$(shell if [ -d /usr/local/mysql/include ]; then echo -I/usr/local/mysql/include; fi) CFLAGS+=$(shell if [ -d /usr/local/mysql/include/mysql ]; then echo -I/usr/local/mysql/include/mysql; fi) --- CFLAGS+=$(shell if [ -d /usr/include/mysql ]; then echo CFLAGS+-I/usr/include/mysql; fi) =$(shell if [ -d CFLAGS+/usr/local/include/mysql ]; then echo CFLAGS+-I/usr/local/include/mysql; fi) =$(shell if [ -d CFLAGS+/opt/mysql/include/mysql ]; then echo CFLAGS+-I/opt/mysql/include/mysql; fi) MLFLAGS= MLFLAGS+=$(shell if [ -d /usr/lib/mysql ]; then echo -L/usr/lib/mysql; MLFLAGS+fi) =$(shell if [ -d /usr/lib64/mysql ]; then echo MLFLAGS+-L/usr/lib64/mysql; fi) =$(shell if [ -d /usr/local/mysql/lib ]; then echo -L/usr/local/mysql/lib; fi) MLFLAGS+=$(shell if [ -d /usr/local/mysql/lib/mysql ]; then echo -L/usr/local/mysql/lib/mysql; fi)--- MLFLAGS+=$(shell if [ -d /usr/local/lib/mysql ]; then echo MLFLAGS+-L/usr/local/lib/mysql; fi) =$(shell if [ -d MLFLAGS+/opt/mysql/lib/mysql ]; then echo -L/opt/mysql/lib/mysql; fi) Details for compiler and libs: dpkg -l | grep gcc ii gcc 4.0.2-1 The GNU C compiler ii gcc-3.3-base 3.3.6-7 The GNU Compiler Collection (base package) ii gcc-4.0 4.0.2-2 The GNU C compiler ii gcc-4.0-base 4.0.2-2 The GNU Compiler Collection (base package) ii libgcc1 4.0.2-2 GCC support library dpkg -l | grep libssl-dev ii
RE: [Asterisk-Users] Using Long Distance Operators
Yes, something like below should do what you want.
Exten = _90.,1,Dial(ZAP/g1/7980${EXTEN:2})
_
From: Carlos Prieto [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 22, 2005 1:20 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Using Long Distance Operators
Hi !
I'm in a project of deploying an Asterisk server instead of a Panasonic
PBX on a customer.
Actually, they use a different operator from their PRI supplier for long
and international calls. In the Panasonic PBX, is saved that when a user
tries to reach a long distance number, they dial 9 (for getting dial
tone) and then they dial the number: 0 for long distance and
00 for international calls; in the first case, the 0 is
replaced for the operator number, we say 789, so the Panasonic really
dials: 789-0-. for long distance calls, and 789-00- for
international calls. So, the users don't dial the 789.
Is there a way to implement that dial plan in the Asterisk?
Thanks in advance for the help.
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[Asterisk-Users] sip routing
Hello, Can we configure asterisk in order to send sip requests to a outbound proxy when asterisk get AOR of users agents with an private ip ? Asterisk AOR:[EMAIL PROTECTED] ip | | sip proxy/nat box---user agent 192.168.0.0/24 Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on AMD64
Hi, I've been using it on both P4 and AMD64 (32 and 64 bit). Performance is about the same. We also didn't have any special compile or usage problems. David Lowes Mark Quitoriano wrote: anyone tried using asterisk on AMD64? how's the performance is better than p4? -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 error: Ouch ... error while writing audio data: : Broken pipe
On Fri, Nov 18, 2005 at 10:22:23AM -0600, Kevin P. Fleming wrote: Leo Burd wrote: Any ideas about what is going on? Yes. You didn't read the warnings prominently displayed at the end of 'make install' about removing old modules from /usr/lib/asterisk/modules. Does that include the 729 codec modules, or can they stay there for 1.2? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call waiting issue
A simple sql command will do this. -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 22, 2005 1:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Call waiting issue Whenever I restart Asterisk, I then have to go to each phone and dial *70 to turn call waiting back on so that the multiple lines on the phones will ring through instead of getting a busy when the user is only on a single call. Is there a simple way to have call waiting be On by default? -Kerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spandsp/rxfax/txfax Asterisk 1.2stable - problems loading the modules
Hi all, today I installed asterisk 1.2stable and than spandsp-0.0.2pre21 with rxfax txfax. After I restart the asterisk and get the following errors: [app_rxfax.so] WARNING[6340]: loader.c:325 __load_resource: /usr/lib/ asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler WARNING[6340]: loader.c:554 load_modules: Loading module app_rxfax.so failed! [app_txfax.so] WARNING[6311]: loader.c:325 __load_resource: /usr/lib/ asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_info WARNING[6311]: loader.c:554 load_modules: Loading module app_txfax.so failed! I am running Asterisk on fedora core 4 - all works great (Asterisk, app_conference and others...), but tx/rxfax failed :( Now I found the following message on http://www.asteriskguru.com/ tutorials/spandsp.html: // START // 2) If you receive a message like the following: [app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_info Oct 5 12:05:24 WARNING[14665]: loader.c:543 load_modules: Loading module app_txfax.so failed! Ouch ... error while writing audio data: : Broken pipe When you execute the asterisk -vvvc command and the Asterisk crashes when you try to execute the safe_asterisk command, then very probably you have the following problem: The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist. The solution is to find the location of this symlink and to delete it manually. Usually it is in the /usr/lib/ directory. /// STOP /// - But I only have spandsl-0.0.2 installed, and the libs are in /usr/ local/lib, see: -rw-r--r-- 1 root root 946280 18. Nov 22:02 libspandsp.a -rwxr-xr-x 1 root root 822 18. Nov 22:02 libspandsp.la lrwxrwxrwx 1 root root 19 18. Nov 22:02 libspandsp.so - libspandsp.so.0.0.1 lrwxrwxrwx 1 root root 19 18. Nov 22:02 libspandsp.so.0 - libspandsp.so.0.0.1 -rwxr-xr-x 1 root root 738959 18. Nov 22:02 libspandsp.so.0.0.1 - Does anybody have the same problems? Best regards and thx for help! Dominik Simon. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk to Fax Server
Title: Message There is a problem with Avaya that DS1 cards are nor recognizing incoming FAX. Using unified messaging I must answer the call and if I hear a Faxpulse I have to transferthe call to UM. I want Asterisk do this job. Recognize fax and send directly to UM. In my previous mail I used the term "Fax server"to make it simple Thanks Arcady -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy KuoSent: Monday, November 21, 2005 8:38 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk to Fax ServerWhat Fax server are you using? On 11/21/05, Arcady Litmanovich [EMAIL PROTECTED] wrote: Hi I'm looking for following solution: Asterisk is connected to PSTN by Digium or some another card which has Fax Detection If incoming call is a fax I woud like to transfer it to External Fax server by SIP or H323 for getting a Fax. If incoming call is a voice to direct it to another trunk. Is it possible to make it on Asterisk? If yes which E1 card is preferable? Thanks in advance Arcady ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi_cm-0.6.1: ISDN1: too much voice to send for NCCI=0x10101
Answering myself here. It turned out that the machine already had kernelcapi installed and was doing some weird things with the modules. I removed it and reinstalled isdn-utils. All is now well! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up FXO in router
Paul,
Thanx for your suggestions, but no luck ths far.On 11/22/05, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Once when setting up a SIP based mobile phone
gateway, I had to use
(SIP/${EXTEN)@rupert) and set up an entry in sip.conf for rupert.
This lets you use passwords, etc.
Worth a try, if nothing else.
PaulH
- Original Message -
From:
Gary
Stark
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 22, 2005 1:35
PM
Subject: Re: [Asterisk-Users] Setting up
FXO in router
Many thanx Paul,Yes, I am having fun with this
stuff...If I set this to exten = _9.,1,Dial(SIP/[EMAIL PROTECTED])
I
get the following messages in the log file and on the CLI console.channel.c: Channel allocation failed: Can't create alert pipe!chan_local.c: Unable to allocate channel structure(s)app_dial.c: Unable to create local channel for call forward to '
Local/[EMAIL PROTECTED]' (cause = 0)If I try exten = _9.,1,Dial(IAX2/[EMAIL PROTECTED]
)I then get chan_iax2.c: Rejected connect attempt from 192.168.0.5, who was trying to reach 's@'
chan_iax2.c: Call rejected by 192.168.0.5: No authority foundI'm getting close, but need some assistance for that last few yards. :)
On 11/22/05, [EMAIL PROTECTED]
[EMAIL PROTECTED]
wrote:
Some of the sip gear just needs a DIAL(SIP/[EMAIL PROTECTED]
) to make it dial
out.
later,
PaulH
-
Original Message -
From:
Gary Stark
To:
asterisk-users@lists.digium.com
Sent:
Tuesday, November 22, 2005 12:09 PM
Subject:
[Asterisk-Users] Setting up FXO in router
G'day y'all.I have a mostly working asterisk
installation, and attached to my LAN is a a Netcomm NB5W
router/gateway, which includes two FXS ports and one FXO port.I
have the FXS ports configured via the internal web configuration on the
router, and they're happily working as extensions from my asterisk server
(albeit with a couple of minor hang-up detection issues) but I now want to
use the FXO port on the router as well, but I cannot see anything,
anywhere, that tells me how I can start on this aspect of the
configuration of the whole system.Can somebody please oint me at
some documentation that is relevant to this?Thanx in advance for
any and all assistance.-- g.Gary
Stark[EMAIL PROTECTED]
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-- g.Gary Stark[EMAIL PROTECTED]
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listAsterisk-Users@lists.digium.com
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UNSUBSCRIBE or update options visit:
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http://lists.digium.com/mailman/listinfo/asterisk-users-- g.Gary Stark[EMAIL PROTECTED]
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Re: [Asterisk-Users] Problems with fax failing when bridged across TDM400Pvers E
Hi,
What I do not understand is how dropped packets prevent the fax from
working. Faxes are designed to adopt to noise on the line by reducing
their connection speed. It seems like their is something else besides
packet loss going on here. Also why would the board work for receiving
faxes but not sending???
Thanks
Rich Adamson wrote:
In my case, we only get a small number of faxes, so I outsourced it to
a company with an 800 number. Faxes are received in pdf form via email
which is much better for us anyway. The few times I need to send a fax,
I simply unplug one of four pstn lines and use it with the fax.
Would be nice if the TDM could handle modem calls.
Hi,
Okay I tried what you suggested and bummer I get 99.987793%. So how do
you handle faxes.. We want to use a fax machine and make the line
available for voice when the fax is not used which is not much. We use a
trunk dial setup for this which works great. I am all ears there must be
a way to fix this??
Thanks
Rich Adamson wrote:
Try running /usr/src/zaptel/zttest -v and see what you get for results.
If you get something like this:
Opened pseudo zap interface, measuring accuracy...
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
where the intervals are less then 100%, then the problem is the same
issue we've all been fighting for some time. The less then 100% is
indicative of 'missed frames' of data from the TDM card to the zaptel
drivers (eg, pci bus).
Fax modem transmissions will never succeed if that number is less then
100%. I can't get 100% out of my system regardless of what I've tried,
including two different distro's, two different MBs, etc.
Rich
H,
We have a fax machine connected to a FXS modules on TDM400P card. There
is an FXO module connected to a pots line. We can receive faxes okay but
we seem to be having trouble sending them. The connection is bridged
between the appropriate ZAP channels but it just hangs there. The remote
fax answers and the fax machine indicates that it is connecting - but
then we get transmission errors and the fax fails. I had tuned the line
to get rid echos using ./ztmonitor (works great by the way) could I have
tuned it too much or should I tune each channel separately (you can see
my rxgain and txgain settings in zapata.conf)? Here are my files.
zapata.conf
[trunkgroups]
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=14.0
txgain=4.0
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
immediate=no
faxdetect=both
context=incoming-home
signalling=fxs_ks
group=1
channel = 1,2
context=local
signalling=fxo_ks
group=2
channel = 3
context=longdistance
signalling=fxo_ks
group=3
channel = 4
***
extensions.conf
[general]
#include macros.incl
[incoming-home]
exten = s,1,Goto(extensions-home,100,1)
exten = t,1,Goto(extensions-home,100,1)
exten = i,1,Goto(extensions-home,100,1)
[extensions-home]
include = parkedcalls
;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer()
exten = 100,2,Queue(extensions-home|trn|||120)
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 590,1,Dial(ZAP/3,20)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin(,,@extensions-home)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;exten = i,1,Voicemail(s300)
;exten = t,1,Voicemail(s300)
exten = fax,1,Dial(ZAP/4,20)
exten = fax,2,Congestion
exten = fax,102,Congestion
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911
include = extensions-home
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
[globals]
OUTBOUNDTRUNK=Zap/G1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
***
modules.conf
modules]
autoload=yes
;
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload = app_intercom.so
;
; Explicitly load the chan_modem.so early on to be sure
; it loads before any of the chan_modem_* 's afte rit
;
noload = chan_modem.so
noload = chan_modem_aopen.so
noload = chan_modem_bestdata.so
noload =
Re: [Asterisk-Users] Codec that quality does not get affect *much* against packet loss
I think you are thinking of iLBC: http://www.voip-info.org/wiki-iLBC Be aware that this codec is known to be pretty CPU intensive to accomplish its compression. - PedroOn 11/22/05, Sam Tam [EMAIL PROTECTED] wrote: I think I have heard in the past that someone mentioned to me there is acodec that does not getting affected much because of packet loss.Is there such thing?Sam___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Menu Tree Delay
Kerry Garrison wrote: I have a fairly simple menu structure, three options branch to submenus. There is a long (several seconds) delay between pressing a key and getting the next menu. This happens on 2 out of 3 of my menus for no apparent reason. I am kind of at a loss as to what to look at. Any suggestions would be appreciated. The problem is that asterisk does not know if it needs to wait for additional digits so is waiting for a timeout. When someone dials 3, are they done or could they also dial 30 or 301? The way to get rid of this wait is to make sure any other numbers in this context begin with a different digit. Only have 1 option 1 2 option 2 3 option 3 and not additional lines like 301 extension 301 Extensions in the same context would begin with a 4, 5, 6, 7, 8, 9. Don Pobanz ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.10
I noticed that asterisk.org now has asterisk and zaptel downloads for version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9 version number. Just wondering for those using the 1.0.x versions of asterisk instead of the 1.2 versions - will libpri, addons and sounds be updated to match the 1.0.10 version or will 1.0.9 be the final release of those packages? - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: problem with registration of SIP phone
And one more update that may help to find a solution to this problem. If I run asterisk -rx reload the registration works fine until the next re-registration and then I have the same error again Is there some solution for this problem exept runnning asterisk -rx reload all the time ? On 11/22/05, Asterisk User [EMAIL PROTECTED] wrote: I managed to isolate the problem a bit more, maybe it will help to find a solution:The problem with the phones is not the initial registration, but the re-registration process.When I create a new extension the phone registers ok, but when the same phone tries to re-register it fails. On 11/22/05, Asterisk User [EMAIL PROTECTED] wrote: I'm runing [EMAIL PROTECTED] beta6 and I have a I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart or after long idle time the phones can't register and I get this in the server:Transmitting (no NAT) to 10.1.1.152:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.1.152;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152From: sip: [EMAIL PROTECTED];tag=12e8dd0080754148To: sip: [EMAIL PROTECTED];tag=as2383b1df Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip: [EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=54cb5290 Content-Length: 0After a few server restarts and/or phone restarts the phone registers ok. Any ideas why ?Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp/rxfax/txfax Asterisk 1.2stable -
Dominik Simon wrote: Hi all, today I installed asterisk 1.2stable and than spandsp-0.0.2pre21 with rxfax txfax. After I restart the asterisk and get the following errors: - But I only have spandsl-0.0.2 installed, and the libs are in /usr/ local/lib, see: -rw-r--r-- 1 root root 946280 18. Nov 22:02 libspandsp.a -rwxr-xr-x 1 root root 822 18. Nov 22:02 libspandsp.la lrwxrwxrwx 1 root root 19 18. Nov 22:02 libspandsp.so - libspandsp.so.0.0.1 lrwxrwxrwx 1 root root 19 18. Nov 22:02 libspandsp.so.0 - libspandsp.so.0.0.1 -rwxr-xr-x 1 root root 738959 18. Nov 22:02 libspandsp.so.0.0.1 Your output shows that you have 0.0.1 installed. Delete the older version of spandsp and re-install 0.0.2 Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setting caller ID with Voicepulse
Due to some change I've been unable to identify, my Asterisk box is no longer successfully passing caller ID to the called party with calls placed through Voicepulse. This worked just fine until recently. Also, identical code functions correctly (caller ID arrives) when the call is sent via Junction Networks. I could post a fragment of extensions.conf, but before I do, I wonder if any other users of Voicepulse might want to check for problems. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk 1.0.10
Note - looks like the answer to this was posted out of *date* sequence on asterisk.org (it is below the 1.2.0 release notice): direct from asterisk.org homepage: Version 1.0.10 has been released of Asterisk and Zaptel. Libpri, Asterisk-addons, and Asterisk-sounds contain no changes, so they have not been updated. It is very likely that this will be the final release of the 1.0 branch of Asterisk. Users are strongly encouraged to begin upgrading to version 1.2. Thanks! On 11/22/05, Pedro [EMAIL PROTECTED] wrote: I noticed that asterisk.org now has asterisk and zaptel downloads for version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9 version number. Just wondering for those using the 1.0.x versions of asterisk instead of the 1.2 versions - will libpri, addons and sounds be updated to match the 1.0.10 version or will 1.0.9 be the final release of those packages? - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP debugging tools - Suggestions experience?
Hi, I previously posted a problem with my Zyxel P2000Wv2 wireless SIP phones and agent logins. In order to solve this problem I am looking at SIP debugging tools but I have limited experience with them. Some of the visual tools will not work as they require a software SIP phone to use and since my problem only occurs when the Zyxel phone is used and not a software SIP phone that will not work. I looked at asterisks 'sip debug' but I have not found good information about interpreting this output. Can anybody with experience in doing this make some suggestions. Also would this be something that should be posted on the developers forum??? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 error: Ouch ... error while writing audio data: : Broken pipe
On 11/22/05, Michael George [EMAIL PROTECTED] wrote: On Fri, Nov 18, 2005 at 10:22:23AM -0600, Kevin P. Fleming wrote: Leo Burd wrote: Any ideas about what is going on? Yes. You didn't read the warnings prominently displayed at the end of 'make install' about removing old modules from /usr/lib/asterisk/modules. Does that include the 729 codec modules, or can they stay there for 1.2? It probably depends on how old those modules are. The current modules available for download have had significant optimizations made to them by Kevin back in late September/eary October so if you haven't picked them up already, I'd advise you to do that for that reason alone. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] open letter
Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to register Zyxel WIFI Phone as SIP Client to Asterisk
Hi, I do not know if you got a reply to your questions already but I found that only the version 2 of this phone with the latest firmware works. There was a bug in the fireware where only numerical characters could be used to log in. Alpha numeric will not work unless the firmware is upgraded. Hope this helps Kanuri, Seshu (Company IT) wrote: Folks! I have this expensive gizmo Zyxel-2000 WIFI Wireless Phone that can run SIP protocol. I have configured this to my Asterisk as a SIP client but cannot register at the server. I have a basic configuration entry in sip.conf and I am running it having the client connected with a Dynamic DHCP address. My Asterisk server is running fine and it has several SIP and IAX2 clients. No problem there I have used the following options in sip.conf as trial and error in various combinations nat=yes host=dynamic canreinvite=no defaulthost=xx.xx.xx.xx Asterisk sees the phone trying to connect but it cannot authenticate with the Login/Pass Does anyone have a working configuration for the Phone as well as sip.conf entry? If not any suggestions Thanks Seshu Kanuri Morgan Stanley | Technology 1633 Broadway | Floor 19 New York, NY 10019 Phone: +1 212 537-2849 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 6/8/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bad Lines - What can the phone company do?
We suffer with some bad CO lines in the Seattle Redmond area. To compensate our gains have been tuned 10 rx and 2 tx. We have also had to add a 3 second wait to outgoing calls because many times the front of the number gets missed by the telco. Is there anything we can request from the phone company? They have checked our lines (probably just for tone) and say there is nothing wrong. Thanks! -Justin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bad Lines - What can the phone company do?
Claim that emergancy health equipment does not function, that will put them in action. Better yet tell them that 911 is not captured! On 11/22/05, Justin Selleck [EMAIL PROTECTED] wrote: We suffer with some bad CO lines in the Seattle Redmond area. To compensate our gains have been tuned 10 rx and 2 tx. We have also had to add a 3 second wait to outgoing calls because many times the front of the number gets missed by the telco. Is there anything we can request from the phone company? They have checked our lines (probably just for tone) and say there is nothing wrong. Thanks! -Justin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Users] open letter
Let me get this straight All you are doing is registering the devices with SER (below you have mentioned asterisk, and then you say they goto ser) Once they are registered to ser you wish to send them to asterisk...is this correct If so, this does not seem to hard, NAT ius dealt with in ser, I use mediaproxy, others may use nathelper, so before you send to asterisk take care of NAT issues in SER and then send to asterisk. Paste config, in pastebin, and also a ngrep of the call debug. Iqbal harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digital Assitant Help
I am the network administrator for a small school in Michigan. We are currently using an older proprietary pbx system and are trying very hard to get away from this one vender lock in. I have set up an asterisk server using the version 1.2 of asterisk. Our current system uses mailboxes and extensions for teachers to dial a * (ex. *6913) enter a password and then record the homework that they have assigned for the week. Then when a parent calls from the outside they are redirected through a prompt system to the correct class and they are redirected to mailbox 6913 without the ability to record a message there. Even if a teacher wanted to they cant edit this from the outside for security reasons. Also in this system is an automated school closing system along the same lines where the superintendent dials a number, puts in a password, and chooses an option for school closing reasons 1. Fog delay for 2 hours, 2. Snow closing and this message is automatically taken off the system and the default message is restored at 12:00am. I am very new to asterisk and any help would be appreciated. Johnathan Falk Network Administrator Clinton Community Schools 1-517-442-9622 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bad Lines - What can the phone company do?
--- Justin Selleck [EMAIL PROTECTED] wrote: We suffer with some bad CO lines in the Seattle Redmond area. To compensate our gains have been tuned 10 rx and 2 tx. We have also had to add a 3 second wait to outgoing calls because many times the front of the number gets missed by the telco. Is there anything we can request from the phone company? They have checked our lines (probably just for tone) and say there is nothing wrong. Thanks! -Justin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sounds like you need an OPX gain module to automatically adjust volume and ring back. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digital Assitant Help
On Tue, 2005-11-22 at 09:57 -0500, Johnathan Falk wrote: I am the network administrator for a small school in Michigan. We are currently using an older proprietary pbx system and are trying very hard to get away from this one vender lock in. I have set up an asterisk server using the version 1.2 of asterisk. Our current system uses mailboxes and extensions for teachers to dial a * (ex. *6913) enter a password and then record the homework that they have assigned for the week. Then when a parent calls from the outside they are redirected through a prompt system to the correct class and they are redirected to mailbox 6913 without the ability to record a message there. Even if a teacher wanted to they can’t edit this from the outside for security reasons. Also in this system is an automated school closing system along the same lines where the superintendent dials a number, puts in a password, and chooses an option for school closing reasons “1. Fog delay for 2 hours, 2. Snow closing” and this message is automatically taken off the system and the default message is restored at 12:00am. I am very new to asterisk and any help would be appreciated. Don't use voicemail. Use Record() to record the messages for later Playback(). Authenticate() may be the right solution for your 'password'. HTH. --roger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unwated outgoing Zap channel briding
Hi, I have a problem with our office PBX where outgoing FXO Zap channels get bridged and i cannot receive or make any phonecalls. First I disabled flash function and we are using # sign to do transfers between internal lines but it still happends from time to time. So is there a way to specify that certains channels lets say 1-4 should not be bridged togather EVER ?? regards m. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] REPOST:How do you get a sound to play to caller on answer?
I tried this dial command to get a sound to play to the caller on answer. I have even tried to use the LIMIT_CONNECT_FILE option with no success. As can be seen below the start_sound variable shows 'UNDEF'. Are there some other settings I have missed out, eg. file location, type etc. The sound file is in GSM format. SIP/providername/002345678|42|HL(2658:61000:3:LIMIT_CONNECT_FILE=soundfile) -- Limit Data: -- timelimit=2658 -- play_warning=61000 -- play_to_caller=yes -- play_to_callee=no -- warning_freq=3 -- start_sound=UNDEF -- warning_sound=timeleft -- end_sound=UNDEF Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Serusers] open letter
You lost me here. Was that a question or a statement? I might not be able to help, since my SER usage is totally diffent, but let me see if I got this right: - You want the SER to forward REGISTER messages to the Asterisk. - The user agents use private IP addresses. - You want the SER to perform NAT? (I'm guessing here) How a session between two user agents behind nat could keep in the path That is the question Since you a talking of a session, do you talk of calls now? yes Could you perhaps post the parts of ser.cfg that deal with register requests? I added this in register block rewritehostport(nxs.yi.org:5050); t_relay_to_udp(nxs.yi.org,5050); 5050 = asterisk port Regards, Stefan ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slightly OT - Anyone know of an external ringer compatible with Cisco phones
Have an application where Cisco phones are being used in a noisy environmentlooking for some type of external ringer or amplifier so users can hear the phones ringing over the background noise. Anyone familiar with such a device? Thanks, -- Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with fax failing when bridged across TDM400Pvers E
On Tuesday 22 November 2005 08:54, Chuck Bunn wrote: What I do not understand is how dropped packets prevent the fax from working. Faxes are designed to adopt to noise on the line by reducing their connection speed. It seems like their is something else besides packet loss going on here. Also why would the board work for receiving faxes but not sending??? Faxes are designed to work around the noise and other signal problems inherent in analog telephony. VOIP introduces an entirely different set of noise factors that fax machines are frankly ill-equipped to deal with. Jitter and dropped packets are the biggest of these issues. Now which version of Zaptel are you running (not Asterisk, Zaptel) -- was it part of the Asterisk 1.0.x series, are you running 1.2 or CVS HEAD? It is a possibility that this issue is fixed now in CVS HEAD (possibly 1.2 as well, I haven't been keeping track), so it is advisable to try the latest version of Zaptel. You do not need to upgrade to the latest Asterisk in order to use the latest Zaptel. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Users] open letter
Let me get this straight All you are doing is registering the devices with SER (below you have mentioned asterisk, and then you say they goto ser) No to asterisk. Asterisk should handle INVITE, REGISTER via ser. SER should handle IM/presence Once they are registered to ser you wish to send them to asterisk...is this correct If so, this does not seem to hard, NAT ius dealt with in ser, I use mediaproxy, others may use nathelper, so before you send to asterisk take care of NAT issues in SER and then send to asterisk. Paste config, in pastebin, and also a ngrep of the call debug. Iqbal harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users . ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Bad Lines - What can the phone company do?
Andrew Latham [EMAIL PROTECTED] wrote: Claim that emergancy health equipment does not function, that will put them in action. Better yet tell them that 911 is not captured! I'm going to have to remember that one! Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Server Side AgentCallbackLogin
Here's what I'm trying to do.. We have a small system, there are only two of us. We both do sales and we both do support. We like Queues better than music on hold with a bunch of dials happening in the background to try our phones, then cells, etc. Problem is, we don't like the idea of having to login to a queue and are wondering if there is a way to force/automatically log agents into a queue without having to do anything on the phone; have it be server side that is. I'm thinking some sort of cron job that runs every minute or five to make sure all expected agents (my partner and I) are in the queue and if not, log us in. The extentions we use to enter the queue are find-me extensions so if we aren't at our desks, calls will hit our cells. Like I said, we know we can do this by doing some excessive dialplan authoring, but we'd rather use the pre-build Queues -- they do everything we need/want, except the autologin part. Anyone know how we can solve this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Menu Tree Delay
Ok that's a big DUH on my part. And since most people like to have 1xx or 2xx for extensions, this is going to be a continuing problem. If you have a large menu, you are going to quickly run out of digits. Otherwise, is there some trick I can use to move between contexts to avoid this problem? -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don Pobanz Sent: Tuesday, November 22, 2005 6:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Menu Tree Delay Kerry Garrison wrote: I have a fairly simple menu structure, three options branch to submenus. There is a long (several seconds) delay between pressing a key and getting the next menu. This happens on 2 out of 3 of my menus for no apparent reason. I am kind of at a loss as to what to look at. Any suggestions would be appreciated. The problem is that asterisk does not know if it needs to wait for additional digits so is waiting for a timeout. When someone dials 3, are they done or could they also dial 30 or 301? The way to get rid of this wait is to make sure any other numbers in this context begin with a different digit. Only have 1 option 1 2 option 2 3 option 3 and not additional lines like 301 extension 301 Extensions in the same context would begin with a 4, 5, 6, 7, 8, 9. Don Pobanz ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Master Telephone
I am the network administrator for a small school in Michigan. We are currently using an older proprietary pbx system and are trying very hard to get away from this one vender lock in. I have set up an asterisk server using the version 1.2 of asterisk. Our current system has a master telephone used by the head secretary that can transfer anyones calls and just generally handle all phone redirection. Kind of like a head receptionist. The one rule we have about our telephone system is that during school hours a person must answer. Our superintendent refuses to have a machine answer if there are people working. So the head secretary must either redirect the calls to someone who is there or take the call herself. How can we accomplish this with asterisk? Also if it cant be accomplished through programming what special hardware would we be required to purchase. Johnathan Falk Network Administrator Clinton Community Schools 1-517-442-9622 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp/rxfax/txfax Asterisk 1.2stable -
Hi Doug, hi list, I installed spandsp-0.0.2 were the libspandsp.so.0.0.1 are included, now I installed die spandsp-0.0.3 an you see: the same problem - and now there is the libspandsp.so.0.0.2: [app_txfax.so]Nov 22 16:15:38 WARNING[29448]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_info Nov 22 16:15:38 WARNING[29448]: loader.c:554 load_modules: Loading module app_txfax.so failed! [EMAIL PROTECTED] /]# cd /usr/local/lib [EMAIL PROTECTED] lib]# ll -rw-r--r-- 1 root root 1143730 22. Nov 16:06 libspandsp.a -rwxr-xr-x 1 root root 822 22. Nov 16:06 libspandsp.la lrwxrwxrwx 1 root root 19 22. Nov 16:06 libspandsp.so - libspandsp.so.0.0.2 lrwxrwxrwx 1 root root 19 22. Nov 16:06 libspandsp.so.0 - libspandsp.so.0.0.2 -rwxr-xr-x 1 root root 873774 22. Nov 16:06 libspandsp.so.0.0.2 Whats going wrong? I cant see the mistake... Best regards, Dominik Am 22.11.2005 um 15:10 schrieb Doug Lytle: Dominik Simon wrote: Hi all, today I installed asterisk 1.2stable and than spandsp-0.0.2pre21 with rxfax txfax. After I restart the asterisk and get the following errors: - But I only have spandsl-0.0.2 installed, and the libs are in /usr/ local/lib, see: -rw-r--r-- 1 root root 946280 18. Nov 22:02 libspandsp.a -rwxr-xr-x 1 root root 822 18. Nov 22:02 libspandsp.la lrwxrwxrwx 1 root root 19 18. Nov 22:02 libspandsp.so - libspandsp.so.0.0.1 lrwxrwxrwx 1 root root 19 18. Nov 22:02 libspandsp.so.0 - libspandsp.so.0.0.1 -rwxr-xr-x 1 root root 738959 18. Nov 22:02 libspandsp.so.0.0.1 Your output shows that you have 0.0.1 installed. Delete the older version of spandsp and re-install 0.0.2 Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP Installation
Has anyone had any success installing AMP 1.10 on a Asterisk 1.2.0. If so can anyone shed some light on how to install it? I am looking for an install or someone sort of script to run the installation and I can t see it. Any assistance would be appreciated. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA verse Wildcard TDM400P
I fed up with X100P clone card and want to spurge for a better solution. I do not need a router or firewall within this device and really just need basic features. I am considering ATA adapters such as the Sipura 3000, Cisco ATA, Grandstream 488 or a Digium Wildcard TDM400P with one FXO. Does anyone have direct experience with both ATAs and the TDM400P with one FXO port they would share? Kind of Pros and Cons on both solutions. You may reply off list. I appreciate any suggestions and assistance. Thanks, Chip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Users] open letter
okay, so ALL your users are registering to asterisk...is that correct. If so the problem is howto accept users from behind a NAT into asterisk, or am I confusing things further. If the above are true, where is SER in this, or are users hitting SER and you are sending the REGISTER from ser into asterisk. Iqbal harry gaillac wrote: Let me get this straight All you are doing is registering the devices with SER (below you have mentioned asterisk, and then you say they goto ser) No to asterisk. Asterisk should handle INVITE, REGISTER via ser. SER should handle IM/presence Once they are registered to ser you wish to send them to asterisk...is this correct If so, this does not seem to hard, NAT ius dealt with in ser, I use mediaproxy, others may use nathelper, so before you send to asterisk take care of NAT issues in SER and then send to asterisk. Paste config, in pastebin, and also a ngrep of the call debug. Iqbal harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users . ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call waiting issue
I'm not following, must be too tired. Are you saying that on startup I could run a SQL command that toggles everyone's call waiting status? -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, November 22, 2005 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call waiting issue A simple sql command will do this. -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 22, 2005 1:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Call waiting issue Whenever I restart Asterisk, I then have to go to each phone and dial *70 to turn call waiting back on so that the multiple lines on the phones will ring through instead of getting a busy when the user is only on a single call. Is there a simple way to have call waiting be On by default? -Kerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Slightly OT - Anyone know of an external ringer compatible with Cisco phones
Hey Cory, I havent come across a voip ring amplifier or visual indicator. Here are some amplifiers and visual inidicators for an environment using ATAs: http://www.soundbytes.com/Merchant2/merchant.mvc?Screen=CTGYStore_Code=SBCategory_Code=PhoneRingAmplifier Omar A. SabekOn 11/22/05, Cory Andrews [EMAIL PROTECTED] wrote: Have an application where Cisco phones are being used in a noisyenvironmentlooking for some type of external ringer or amplifier sousers can hear the phones ringing over the background noise.Anyonefamiliar with such a device? Thanks,--Cory J AndrewsPartner / Purchasing+++VOIPSupply.com - Everything you need for VOIP454 Sonwil DriveBuffalo, NY 14225+++tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22f - 716.630.1548e - [EMAIL PROTECTED]AIM - b2Cory___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Users] open letter
okay, so ALL your users are registering to asterisk...is that correct. Correct via ser as outbound sip proxy If so the problem is howto accept users from behind a NAT into asterisk, or am I confusing things further. the problem is in contact field. when user agents send register we have in sip hf Contact sip:[EMAIL PROTECTED] So asterisk store this AOR and try to contact agent via nat box instead of SER If the above are true, where is SER in this, or are users hitting SER and you are sending the REGISTER from ser into asterisk. SER is an outbound sip proxy which handle IM presence nat Harry One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private | ---- |-- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with fax failing when bridged across TDM400Pvers E
Hi, I am running Asterisk 1.2 and zaptel 1.2 with the latest Digium board version. Thanks Andrew Kohlsmith wrote: On Tuesday 22 November 2005 08:54, Chuck Bunn wrote: What I do not understand is how dropped packets prevent the fax from working. Faxes are designed to adopt to noise on the line by reducing their connection speed. It seems like their is something else besides packet loss going on here. Also why would the board work for receiving faxes but not sending??? Faxes are designed to work around the noise and other signal problems inherent in analog telephony. VOIP introduces an entirely different set of noise factors that fax machines are frankly ill-equipped to deal with. Jitter and dropped packets are the biggest of these issues. Now which version of Zaptel are you running (not Asterisk, Zaptel) -- was it part of the Asterisk 1.0.x series, are you running 1.2 or CVS HEAD? It is a possibility that this issue is fixed now in CVS HEAD (possibly 1.2 as well, I haven't been keeping track), so it is advisable to try the latest version of Zaptel. You do not need to upgrade to the latest Asterisk in order to use the latest Zaptel. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which is Better!
Which FXO gateway is better and has better sound quality. AudioCodes? Or Mediatrix. Thanks for your input ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best Communications Line for VoIP
We are putting in an Asterisk VoIP solution and was wondering what the best communications medium would be for this implementation. We are going to need 20 telephone lines in/out of our business. We currently have a data T1. Could we put another data T1 to use for Asterisk, or would it be better to put in a Voice T1 or a PRI line? Also, when we do put this T1 or PRI line in, what would be the best equipment to use with the Asterisk box? Any other recommendations would be appreciated? Thank you, Jyran Glucky Advisory Programmer BlueWare, Inc. Strategic HealthWare Solutions 3060 W. 13th Street Cadillac, MI 49601 Phone: (231) 779-0224 ext. 111 Fax: 231-779-1002 mailto:[EMAIL PROTECTED] http://www.blueware.net DID YOU KNOW? BlueWare is the Grand Prize Winner of the 2005 IBM Beacon Award BEST DB2 (Document Management) Application Worldwide. BlueWare Market Share for Hospital Document Management Systems is in 25 states in the US. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.10
Can anyone point me to the changelog for 1.0.10? Craig - Original Message - From: Pedro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 22, 2005 10:04 PM Subject: [Asterisk-Users] Asterisk 1.0.10 I noticed that asterisk.org http://asterisk.org now has asterisk and zaptel downloads for version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9 version number. Just wondering for those using the 1.0.xversions of asterisk instead of the 1.2 versions - will libpri, addons and sounds be updated to match the 1.0.10version or will 1.0.9 be the final release of those packages? - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Slightly OT - Anyone know of an external ringer compatible with Cisco phones
On 11/22/05, Cory Andrews [EMAIL PROTECTED] wrote: Have an application where Cisco phones are being used in a noisy environmentlooking for some type of external ringer or amplifier so users can hear the phones ringing over the background noise. Anyone familiar with such a device? What about customized ringtone alarm like ? The other idea is to plug something in between handset cable so you can detect voltage that i belive is sent over the cable to the nice red light on the handset itself. How many devices like that would u need ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP Installation
There is alot of documentation available if you looked on their website. http://aussievoip.com.au/tiki-index.php?page=1.10.008-Installation On 11/22/05, Goran Donev [EMAIL PROTECTED] wrote: Has anyone had any success installing AMP 1.10 on a Asterisk 1.2.0. If so can anyone shed some light on how to install it? I am looking for an install or someone sort of script to run the installation and I can 't see it. Any assistance would be appreciated. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial() and j option: What is correct?
Kevin Hanson wrote: I thought I read on the list some time ago that the default for 'priorityjumping' is 'yes' so that upgrading to 1.2 won't break old dialplans. Can anyone confirm or deny? That is absolutely correct; unless the [general] section of your extensions.conf contains 'priorityjumping=no' (or the equivalent), then your system will continue to use priority jumping as before. The sample config file, though, does contain this directive, so new users will default to having it turned off. In the post-1.2 development branch we will also be changing the default in the code itself, so that when 1.4 is released then it will be the default for everyone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Server Side AgentCallbackLogin
Hi,
Got here and you will see an example of an automated login.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin
Also the AgentCallbackLogin can be passed parameters automatically when
the extension is dialed.
exten = 801,1,AgentCallbackLogin(${EXTEN:1},,[EMAIL PROTECTED])
Thanks
Jason Lixfeld wrote:
Here's what I'm trying to do.. We have a small system, there are
only two of us. We both do sales and we both do support. We like
Queues better than music on hold with a bunch of dials happening in
the background to try our phones, then cells, etc. Problem is, we
don't like the idea of having to login to a queue and are wondering
if there is a way to force/automatically log agents into a queue
without having to do anything on the phone; have it be server side
that is. I'm thinking some sort of cron job that runs every minute
or five to make sure all expected agents (my partner and I) are in
the queue and if not, log us in. The extentions we use to enter the
queue are find-me extensions so if we aren't at our desks, calls will
hit our cells.
Like I said, we know we can do this by doing some excessive dialplan
authoring, but we'd rather use the pre-build Queues -- they do
everything we need/want, except the autologin part. Anyone know how
we can solve this?
___
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[Asterisk-Users] Re: [Users] open letter
Okay almost there :-) So UA --- asterisk --- SER --- UA is that it harry gaillac wrote: okay, so ALL your users are registering to asterisk...is that correct. Correct via ser as outbound sip proxy If so the problem is howto accept users from behind a NAT into asterisk, or am I confusing things further. the problem is in contact field. when user agents send register we have in sip hf Contact sip:[EMAIL PROTECTED] So asterisk store this AOR and try to contact agent via nat box instead of SER If the above are true, where is SER in this, or are users hitting SER and you are sending the REGISTER from ser into asterisk. SER is an outbound sip proxy which handle IM presence nat Harry One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private | ---- |-- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI and AUTOHANGUP
Comeo'n AGI guys.. Please say something. Hi, Using AUTOHANGUP, I can force a call duration within a time limit. I would like to playback a message before 1 minute of autohangup. How can I accomplish it? Would anybody please give me right direction. Thanks, You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Master Telephone
Hi Johnathan - I am the network administrator for a small school in Michigan. We are currently using an older proprietary pbx system and are trying very hard to get away from this one vender lock in. I have set up an asterisk server using the version 1.2 of asterisk. Our current system has a master telephone used by the head secretary that can transfer anyone's calls and just generally handle all phone redirection. Kind of like a head receptionist. The one rule we have about our telephone system is that during school hours a person must answer. Our superintendent refuses to have a machine answer if there are people working. So the head secretary must either redirect the calls to someone who is there or take the call herself. How can we accomplish this with asterisk? Also if it can't be accomplished through programming what special hardware would we be required to purchase. Asterisk is flexible enough that you can do almost anything with it in terms of dialplan. We also have an always answer policy at my company. We do this with multiple-line sip phones (Polycom IP601). If multiple calls come in, our receptionists answer each one, and put the calls on hold as necessary until they can go back and deal with them all. You could also set something up with queues, so that each person is guaranteed to eventually talk to your receptionist, but they may have to wait a minute or two before actually speaking with her. You could play hold music and/or a greeting during this time. - Noah Miller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digital Assitant Help
Johnathan, I am also located in michigan. Maybee there is a way i can help with this project. I currently use asterisk for alot of custom applications. let me know i'll send you my phone number outside the list Johnathan Falk wrote: I am the network administrator for a small school in Michigan. We are currently using an older proprietary pbx system and are trying very hard to get away from this one vender lock in. I have set up an asterisk server using the version 1.2 of asterisk. Our current system uses mailboxes and extensions for teachers to dial a * (ex. *6913) enter a password and then record the homework that they have assigned for the week. Then when a parent calls from the outside they are redirected through a prompt system to the correct class and they are redirected to mailbox 6913 without the ability to record a message there. Even if a teacher wanted to they can't edit this from the outside for security reasons. Also in this system is an automated school closing system along the same lines where the superintendent dials a number, puts in a password, and chooses an option for school closing reasons 1. Fog delay for 2 hours, 2. Snow closing and this message is automatically taken off the system and the default message is restored at 12:00am. I am very new to asterisk and any help would be appreciated. Johnathan Falk Network Administrator Clinton Community Schools 1-517-442-9622 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] International Dialing Code
Those came from astbill. I will make the changes and reupload, I have gotten a few more changes as well.. Thanks :) On Mon, 2005-11-21 at 22:03 -0800, Innocent Evil wrote: Lots of country have wrong prefix. Andorra,376 should be 1376 Angola,244should be 1244 Antarctica,6721 should be 1672 http://en.wikipedia.org/wiki/List_of_country_calling_codes have good calling codes, but they are not complete and not downloadable :-( You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Sun, 20 Nov 2005 09:27:09 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] International Dialing Code On Sun, 2005-11-20 at 15:20 -0200, Hermann Wecke wrote: Innocent Evil wrote: I am trying to download a list of international dialing codes. Would anybody please post a link to get it Google IS your friend. Did you try? http://www.0xdecafbad.com has one in the first few links of the main page/articles page. In asterisk dialplan format as well as csv for use in LCR scripts or whatever. :) There are 2 known issues though, I just havent bothered yet. vienna austria is a bit off, and someone is sending me updates to israel sometime (to make it more complete and verify for errors). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] VoIPJet Support Contact -We have US unrestricted termination for .095
Doug/Peter/Others, You do realize that you've all just violated your Terms of Service for VoipJet right? Read: https://www.voipjet.com/tos.php Now, go down to near the middle where it says: NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE, THIS INCLUDES BUT IS NOT LIMITED TO, END USERS. CUSTOMERS MAY NOT DISCLOSE USE OF OR PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL, ACCOUNTING AND OTHER DOCUMENTS AND COMMUNICATIONS UNLESS SPECIFICALLY REQUIRED TO DO SO BY LAW. RATES QUOTED TO CUSTOMERS AND RATES PAID BY CUSTOMERS ARE STRICTLY CONFIDENTIAL AND MAY NOT BE SHARED WITH ANY OTHER PERSON OR LEGAL ENTITY. So, if you tell anyone you have voipjet, use voipjet, or complain on these lists that you are having problems with voipjet, you have just legally violated your standing with VoipJet and could be prosecuted and involved in a lawsuit... Now really, do you want to deal with a company like that? Sent: Friday, November 04, 2005 12:43 PM To: Commercial and Business-Oriented Asterisk Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-biz] VoIPJet Support Contact I have not had any issues with Voipjet when I have needed to use them. But there again, I have them, voicepulse and teliax that I use in progression of price in case one fails. Robert On Fri, 4 Nov 2005 14:51:08 -0500 Matt [EMAIL PROTECTED] wrote: This is exactly why I left voipjet for calleveryone.com Calleveryone actually gives me better rates, and there is someone to talk to. I don't know when the guy running voipjet is A) going to realize people aren't going to use a company run as a hobby, and B) people want better support then what he is giving. On 11/4/05, Peter Bowyer [EMAIL PROTECTED] wrote: I don't get a reply from that address either :-( Peter On 04/11/05, Doug Hamilton [EMAIL PROTECTED] wrote: Does anyone have a contact for support from VoIPjet apart from [EMAIL PROTECTED] Have tried the so called fastsupport email address and had nothing in 24 hours. Have been down for a day. Doug Hamilton Prodosec Inc ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.8/161 - Release Date: 11/03/2005 ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA verse Wildcard TDM400P
I personally prefer a single-box solution with a TDM400 (although I am one of the rare people who haven't had problems with the X100Ps I have put in). My office uses an SPA-3000 on a phone line that has call forward on busy to an iax.cc DID line, therefore I speak from experience with both types of solutions. Since I just happen to prefer a single box solution, I normally recommend going with the TDM400. In those cases where it is too inconvenient to get a phone to the server, the SPA-3000 works just fine. I have been running this setup for several months and it has been flawless. So...in my opinion, its more of a matter of what is most appropriate for your needs. If you have convenient access to a phone line, and may expand past a single phone line, then the TDM400 makes more sense. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cpSent: Tuesday, November 22, 2005 7:25 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] ATA verse Wildcard TDM400P I fed up with X100P clone card and want to spurge for a better solution. I do not need a router or firewall within this device and really just need basic features. I am considering ATA adapters such as the Sipura 3000, Cisco ATA, Grandstream 488 or a Digium Wildcard TDM400P with one FXO. Does anyone have direct experience with both ATAs and the TDM400P with one FXO port they would share? Kind of Pros and Cons on both solutions. You may reply off list. I appreciate any suggestions and assistance. Thanks, Chip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Server Side AgentCallbackLogin
why don't you just build your cells into the queues and setup the queue to ringall. Jason Lixfeld wrote: Here's what I'm trying to do.. We have a small system, there are only two of us. We both do sales and we both do support. We like Queues better than music on hold with a bunch of dials happening in the background to try our phones, then cells, etc. Problem is, we don't like the idea of having to login to a queue and are wondering if there is a way to force/automatically log agents into a queue without having to do anything on the phone; have it be server side that is. I'm thinking some sort of cron job that runs every minute or five to make sure all expected agents (my partner and I) are in the queue and if not, log us in. The extentions we use to enter the queue are find-me extensions so if we aren't at our desks, calls will hit our cells. Like I said, we know we can do this by doing some excessive dialplan authoring, but we'd rather use the pre-build Queues -- they do everything we need/want, except the autologin part. Anyone know how we can solve this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I need suggestions for on equipment
Sometime this winter we want to move our company to asterisk from a very old comdial executech phone system. At this point I have a system setup at home that we've been using for several months. I've tried the grandstream bt101 but have had problems keeping it working - some days the message waiting indicator works sometime it doesn't and caller id display is questionable also I'd like to try the gxp2000, I've heard it can auto answer. I also have a sipura 841 but the speaker phone is terrible on that, and I haven't found a way to setup auto answer for paging or transfer announcement. So I need suggestions on phone that can auto answer for paging - to let people know a call is waiting for them. They also need a good caller id display. Also we are out in the country so a T1 is out of the question price wise. We currently have 4 phone lines and would like to increase that to possibly 8. What hardware would you suggest to connect up to 8 phone lines into the asterisk server? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with fax failing when bridged across TDM400Pvers E
Andrew Kohlsmith wrote: Faxes are designed to work around the noise and other signal problems inherent in analog telephony. VOIP introduces an entirely different set of noise factors that fax machines are frankly ill-equipped to deal with. Jitter and dropped packets are the biggest of these issues. Jitter and dropped packets usually translate into either periods of silence (and I believe that this is more common) or periods of synthesized audio. The latter will mean in corrupted data and the former will often result in a premature carrier drop detection. If a receiver prematurely detects a carrier drop in Phase C fax image data then it *must* wait around up to perhaps 60 seconds for the sender signals to return. Many fax machines (and more particularly fax programs) are not this tolerant. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE : [Serusers] Re: [Users] open letter
Just one thing, Register the Uas to asterisk also as outbound proxy. Asterisk will register to SER all the Uas. We use this design: Ua --Asterisk(NAT)-- Ser(public Ip)-- where do you want to go It works perfectly. Maybe I miss something? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Iqbal Envoyé : mardi 22 novembre 2005 16:52 À : harry gaillac Cc : [EMAIL PROTECTED]; asterisk-users@lists.digium.com; users@openser.org Objet : [Serusers] Re: [Users] open letter Okay almost there :-) So UA --- asterisk --- SER --- UA is that it harry gaillac wrote: okay, so ALL your users are registering to asterisk...is that correct. Correct via ser as outbound sip proxy If so the problem is howto accept users from behind a NAT into asterisk, or am I confusing things further. the problem is in contact field. when user agents send register we have in sip hf Contact sip:[EMAIL PROTECTED] So asterisk store this AOR and try to contact agent via nat box instead of SER If the above are true, where is SER in this, or are users hitting SER and you are sending the REGISTER from ser into asterisk. SER is an outbound sip proxy which handle IM presence nat Harry One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private | ---- |-- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com . ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Master Telephone
First, you have to configure your zapata.conf sip.conf to support your hardware (see http://www.voip-info.org/wiki/index.php?page=Asteriskand read http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip,this last is a must-read one) After that, you have to see if all incoming calls are done within work hours: [incoming] exten = s,1,GotoIfTime(9:00-15:00,mon-fri,*,*?accepted,1,10) exten = s,2,thefunctionyouwantdoneifisnthourtime [accepted] exten = 1,1,Dial(thephoneyouwantasmaster) ; add here what you want if that phone is in use as you see, you have no need for new hardware and programming is really simple ;) - Original Message - From: Johnathan Falk To: Asterisk-Users@lists.digium.com Sent: Tuesday, November 22, 2005 4:22 PM Subject: [Asterisk-Users] Master Telephone I am the network administrator for a small school in Michigan. We are currently using an older proprietary pbx system and are trying very hard to get away from this one vender lock in. I have set up an asterisk server using the version 1.2 of asterisk. Our current system has a master telephone used by the head secretary that can transfer anyones calls and just generally handle all phone redirection. Kind of like a head receptionist. The one rule we have about our telephone system is that during school hours a person must answer. Our superintendent refuses to have a machine answer if there are people working. So the head secretary must either redirect the calls to someone who is there or take the call herself. How can we accomplish this with asterisk? Also if it cant be accomplished through programming what special hardware would we be required to purchase. Johnathan FalkNetwork Administrator Clinton Community Schools 1-517-442-9622 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which is Better!
We have tried both but given up hope about them. So now we only use Quintum DX series. Amazing machine Anders Svensson Bobas Communication From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Goran Donev Sent: den 22 november 2005 16:41 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which is Better! Which FXO gateway is better and has better sound quality. AudioCodes? Or Mediatrix. Thanks for your input ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] installing Asterisk from source
Is there a way to install Asterisk from source and not stomp on your already existing Asterisk installation? I don't see a configure script and it looks like it's trying to find stuff in /etc/asterisk and in /usr/lib/asterisk and probably other places. - Jeremy Jones ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RE : [Serusers] Re: [Users] open letter
I do it in reverse do all registration in SER since that is was it was designed for, and then pass to asterisk, and in 1.2 asterisk it has a slew of new features to help with SIP methods, having said that I havent got round to testing any :-) iqbal Olivier Taylor wrote: Just one thing, Register the Uas to asterisk also as outbound proxy. Asterisk will register to SER all the Uas. We use this design: Ua --Asterisk(NAT)-- Ser(public Ip)-- where do you want to go It works perfectly. Maybe I miss something? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Iqbal Envoyé : mardi 22 novembre 2005 16:52 À : harry gaillac Cc : [EMAIL PROTECTED]; asterisk-users@lists.digium.com; users@openser.org Objet : [Serusers] Re: [Users] open letter Okay almost there :-) So UA --- asterisk --- SER --- UA is that it harry gaillac wrote: okay, so ALL your users are registering to asterisk...is that correct. Correct via ser as outbound sip proxy If so the problem is howto accept users from behind a NAT into asterisk, or am I confusing things further. the problem is in contact field. when user agents send register we have in sip hf Contact sip:[EMAIL PROTECTED] So asterisk store this AOR and try to contact agent via nat box instead of SER If the above are true, where is SER in this, or are users hitting SER and you are sending the REGISTER from ser into asterisk. SER is an outbound sip proxy which handle IM presence nat Harry One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private | ---- |-- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com . ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail configuration
Hello,
I have my SIP clients registered with names, and I want to implement the
voicemail in my Asterisk.
I have these lines to redirect the call to the voicemail:
exten = pereira,1,Answer
exten = pereira,2,Wait(1)
exten = pereira,3,VoiceMail(u${EXTEN})
exten = pereira,4,Playback(vm-goodbye)
exten = pereira,5,Hangup
But how do I force this rule to be applied to all calls? instead of
writing these 5 lines for each of my clients ?
If I used numbers, I could do _ ... but how do I write the rule for
client names?
Thanks
Joao Pereira
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Re: [Asterisk-Users] Slightly OT - Anyone know of an external ringer compatible with Cisco phones
I was looking for something off the shelf, this is a one off application, and limited in scope I think they have about a dozen or so handsets in a noisy area they need to beef up the ring volume or present some visual indicator on an incoming call. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory izo wrote: On 11/22/05, Cory Andrews [EMAIL PROTECTED] wrote: Have an application where Cisco phones are being used in a noisy environmentlooking for some type of external ringer or amplifier so users can hear the phones ringing over the background noise. Anyone familiar with such a device? What about customized ringtone alarm like ? The other idea is to plug something in between handset cable so you can detect voltage that i belive is sent over the cable to the nice red light on the handset itself. How many devices like that would u need ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA verse Wildcard TDM400P
--Original Message Text--- From: cp Date: Tue, 22 Nov 2005 10:25:20 -0500 I fed up with X100P clone card and want to spurge for a better solution. I do not need a router or firewall within this device and really just need basic features. I am considering ATA adapters such as the Sipura 3000, Cisco ATA, Grandstream 488 or a Digium Wildcard TDM400P with one FXO. Does anyone have direct experience with both ATAs and the TDM400P with one FXO port they would share? Kind of Pros and Cons on both solutions. You may reply off list. I appreciate any suggestions and assistance. Thanks, Chip In my personal experience both are not good options. The SPA-3000 had audion issues...sound level too low. Echo issues when gains settings increased. The TDM400 is very sensitive to the motherboard/IRQ management issues. There is however a procedural solution that I arrived at in frustration. Sign up for a DID with an ITSP the call forward your POTS line to that number. Works beautifully. Not hardware required. You can drop the iPBX completely if needed by simply defeating the call forwardign and keeping a single analog phone handy. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Server Side AgentCallbackLogin
Hello Jason, if the system is so simple, why don't you connect the queue straight to a couple of you terminals, i.e. not to Agent/101 but to SIP/214. This way you have no login/logout. Yours, l. On Tue, 22 Nov 2005 16:20:50 +0100, Jason Lixfeld [EMAIL PROTECTED] wrote: Here's what I'm trying to do.. We have a small system, there are only two of us. We both do sales and we both do support. We like Queues better than music on hold with a bunch of dials happening in the background to try our phones, then cells, etc. Problem is, we don't like the idea of having to login to a queue and are wondering if there is a way to force/automatically log agents into a queue without having to do anything on the phone; have it be server side that is. I'm thinking some sort of cron job that runs every minute or five to make sure all expected agents (my partner and I) are in the queue and if not, log us in. The extentions we use to enter the queue are find-me extensions so if we aren't at our desks, calls will hit our cells. Like I said, we know we can do this by doing some excessive dialplan authoring, but we'd rather use the pre-build Queues -- they do everything we need/want, except the autologin part. Anyone know how we can solve this? -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Virtual Modems Revisited
I brought this up a while back and althought there are pieces that interface * into Fax Telephony applications, there hasn't been something that works with plain old analog modems. Then I found this piece of code. From my initial tests it looks solid, but I have no clue in how to interface this into asterisk. I thought I would put this link up for other people to comment and try. http://fabrice.bellard.free.fr/linmodem.html Out of the box it works with soundcards. I've been battling jack and alsa for a week trying to get them to play nice just to reroute the audio but I'm out of time in this regard. So I thought I would toss it up and see what other people can come up with. Happy Holidays! Don ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bad Lines - What can the phone company do?
On 11/22/05, Justin Selleck [EMAIL PROTECTED] wrote: We suffer with some bad CO lines in the Seattle Redmond area. To compensate our gains have been tuned 10 rx and 2 tx. We have also had to add a 3 second wait to outgoing calls because many times the front of the number gets missed by the telco. Is there anything we can request from the phone company? They have checked our lines (probably just for tone) and say there is nothing wrong. Did they physically come out and run tests at the demarc? Verizon is usually pretty good about doing that if you ask. Verizon's front line support is clueless, but their onsite techs are pretty good. I had them come out to my place and they gave me all the stats on the line when they were there. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP - ! No D-channels available!
I have had the same issue with a PRI connected from asterisk to an avaya system, it first worked fine, but then started doing this, what is happening is that the D-channel is getting reset for some reason (I have no clue why, but I was able to reproduce it between the avaya, when CID Name was longer than 15 characters, but I know that is not the problem and it is still happening, so I'm assuming it's an avaya problem). Other then that I have no clue how to troubleshoot it, you should have BT test the line for you to see if they can see something wrong with it. On 11/22/05, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: We've had no problems for a few weeks running Asterisk CVS-D2005.10.28.07.54. However, this morning, we're getting users complaining that they were cut off - and I found these in the logs. This has happened 5 times this morning, and there is an entry in the log at the appropriate time. Is this a BT issue ? EuroISDN with a TE405P. Julian. Nov 22 09:53:29 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:53:31 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:53:37 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:53:37 WARNING[8519] app_dial.c: Unable to forward voice Nov 22 09:53:37 WARNING[8505] app_dial.c: Unable to forward voice Nov 22 09:54:34 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:54:34 WARNING[8678] app_dial.c: Unable to forward voice Nov 22 09:54:37 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Nov 22 09:54:38 WARNING[27920] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] International Dialing Code
May I recommend www.numberingplans.com as a resource for checking international dial codes and indeed doing a reverse lookup to find out about a number. We have used this as a resource in the past. Regards Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Tuesday, November 22, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] International Dialing Code Those came from astbill. I will make the changes and reupload, I have gotten a few more changes as well.. Thanks :) On Mon, 2005-11-21 at 22:03 -0800, Innocent Evil wrote: Lots of country have wrong prefix. Andorra,376 should be 1376 Angola,244should be 1244 Antarctica,6721 should be 1672 http://en.wikipedia.org/wiki/List_of_country_calling_codes have good calling codes, but they are not complete and not downloadable :-( You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Sun, 20 Nov 2005 09:27:09 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] International Dialing Code On Sun, 2005-11-20 at 15:20 -0200, Hermann Wecke wrote: Innocent Evil wrote: I am trying to download a list of international dialing codes. Would anybody please post a link to get it Google IS your friend. Did you try? http://www.0xdecafbad.com has one in the first few links of the main page/articles page. In asterisk dialplan format as well as csv for use in LCR scripts or whatever. :) There are 2 known issues though, I just havent bothered yet. vienna austria is a bit off, and someone is sending me updates to israel sometime (to make it more complete and verify for errors). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bad Lines - What can the phone company do?
Also.. All that is required of the phone company is a minimum line quality, anything else is at their pleasure. And if you want to push them a little call up and enter the option to cancel your service. That is the *fastest* way to get to people who can actually do something for you as I found out. Those are the people who instantly scheduled a tech to come out and gave me some discounts also. They put their best people in the customer retention department. Of course that assumes there is a viable option in your area that you can hold over their head. Over in Kirkland Comcast offers full phone service, and for the couple of hundred a month I pay to Verizon they were more than willing to work with me to keep my business. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of If no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2) with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-biz] VoIPJet Support Contact -We have US unrestricted termination for .095
On 22/11/05, Matt [EMAIL PROTECTED] wrote: Doug/Peter/Others, You do realize that you've all just violated your Terms of Service for VoipJet right? Read: https://www.voipjet.com/tos.php Now, go down to near the middle where it says: NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE, THIS INCLUDES BUT IS NOT LIMITED TO, END USERS. CUSTOMERS MAY NOT DISCLOSE USE OF OR PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL, ACCOUNTING AND OTHER DOCUMENTS AND COMMUNICATIONS UNLESS SPECIFICALLY REQUIRED TO DO SO BY LAW. RATES QUOTED TO CUSTOMERS AND RATES PAID BY CUSTOMERS ARE STRICTLY CONFIDENTIAL AND MAY NOT BE SHARED WITH ANY OTHER PERSON OR LEGAL ENTITY. So, if you tell anyone you have voipjet, use voipjet, or complain on these lists that you are having problems with voipjet, you have just legally violated your standing with VoipJet and could be prosecuted and involved in a lawsuit... Now really, do you want to deal with a company like that? On a point of order - all I said was that I couldn't get a response from their support address. At no time did I say (nor am I now) that I use their service. Am I allowed to say that I don't use it anymore? -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Server Side AgentCallbackLogin
On 11/22/05, Jason Lixfeld [EMAIL PROTECTED] wrote: Here's what I'm trying to do.. We have a small system, there are only two of us. We both do sales and we both do support. We like Queues better than music on hold with a bunch of dials happening in the background to try our phones, then cells, etc. Problem is, we don't like the idea of having to login to a queue and are wondering if there is a way to force/automatically log agents into a queue without having to do anything on the phone; have it be server side that is. I'm thinking some sort of cron job that runs every minute or five to make sure all expected agents (my partner and I) are in the queue and if not, log us in. The extentions we use to enter the queue are find-me extensions so if we aren't at our desks, calls will hit our cells. Add static members into the queue in your queues.conf entry. You can use Local channels to find your follow-me [EMAIL PROTECTED] Like: [myqueue] music = default strategy = ringall timeout = 20 member = Local/[EMAIL PROTECTED] -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.6.3 Polycom Firmware?
Has anyone tried the newest Polycom firmware? The release notes indicate they have added support for a new BLA draft. TIA, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to register Zyxel WIFI Phone as SIP Clientto Asterisk
My p2000w Works with asterisk. Here is the sip.conf entry [1006] type= friend subscribecontext = all-local accountcode = 1006 amaflags= default username= 1006 secret = whatever host= dynamic language= en dtmfmode= rfc2833 callerid= Zyxel WiFi toestel 1006 qualify = yes nat = no canreinvite = no mailbox = [EMAIL PROTECTED] disallow= all regards, Joash -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Tuesday, November 22, 2005 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to register Zyxel WIFI Phone as SIP Clientto Asterisk Hi, I do not know if you got a reply to your questions already but I found that only the version 2 of this phone with the latest firmware works. There was a bug in the fireware where only numerical characters could be used to log in. Alpha numeric will not work unless the firmware is upgraded. Hope this helps Kanuri, Seshu (Company IT) wrote: Folks! I have this expensive gizmo Zyxel-2000 WIFI Wireless Phone that can run SIP protocol. I have configured this to my Asterisk as a SIP client but cannot register at the server. I have a basic configuration entry in sip.conf and I am running it having the client connected with a Dynamic DHCP address. My Asterisk server is running fine and it has several SIP and IAX2 clients. No problem there I have used the following options in sip.conf as trial and error in various combinations nat=yes host=dynamic canreinvite=no defaulthost=xx.xx.xx.xx Asterisk sees the phone trying to connect but it cannot authenticate with the Login/Pass Does anyone have a working configuration for the Phone as well as sip.conf entry? If not any suggestions Thanks Seshu Kanuri Morgan Stanley | Technology 1633 Broadway | Floor 19 New York, NY 10019 Phone: +1 212 537-2849 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. --- - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- - No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 6/8/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I need suggestions for on equipment
You could use a Digium TDM2422B, which has 8FXS and 8FXO, and leaves you 8 ports past that for future FXS or FXO expansion. That card, with a normal Asterisk rackmount or tower server, and a mini patch panel and amphenol cable I would think would do the trick. For phones, I would suggest the Linksys SPA-941 or something from Polycom or Snom. Get yourself a decent APC, Tripplite or other battery backup unit as well. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory Tim Litwiller wrote: Sometime this winter we want to move our company to asterisk from a very old comdial executech phone system. At this point I have a system setup at home that we've been using for several months. I've tried the grandstream bt101 but have had problems keeping it working - some days the message waiting indicator works sometime it doesn't and caller id display is questionable also I'd like to try the gxp2000, I've heard it can auto answer. I also have a sipura 841 but the speaker phone is terrible on that, and I haven't found a way to setup auto answer for paging or transfer announcement. So I need suggestions on phone that can auto answer for paging - to let people know a call is waiting for them. They also need a good caller id display. Also we are out in the country so a T1 is out of the question price wise. We currently have 4 phone lines and would like to increase that to possibly 8. What hardware would you suggest to connect up to 8 phone lines into the asterisk server? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
