Re: [asterisk-users] Realtime database function help
You can use the MYSQL function to just use an insert or update statement in
your dialplan. Look at my example below. Instead of using
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ callerid=${ARG1} and blockenabled = 1)
You could use:
exten = s,2,MYSQL(Query resultid ${connid} INSERT INTO\ callerid\
(callerid,blockenabled)\ VALUES\ ('${CALLERID(num)}', '1')\ )
I find that using the ODBC function works best for inserting data into the
MySQL databases.
Have a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc.
[globals]
realdb_host=hostnameformysqldb
realdb_user=mysqldbuser
realdb_pass=mysqldbpassword
realdb_db=mysqldbthatcontainsthevoicemailusers
[macro-checkblacklist]
; This Macro will check the blacklist table to see if the callerid of the
; caller exist and blockenabled =1 (TRUE). If the callerid is listed, then
; tell the caller they have been blacklisted and politely HangUp()
;
; ${ARG1} = CallerID of incoming call
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user}
${realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ callerid=${ARG1} and blockenabled = 1)
exten = s,3,MYSQL(Fetch fetchid ${resultid} blacklistid)
exten = s,4,MYSQL(Clear ${resultid})
exten = s,5,MYSQL(Disconnect ${connid})
exten = s,6,GoToIf($[”${blacklistid}” = “”]?7:fail,1)
exten = s,7,NoOp(${blacklistid})
; If the callerid is listed in the database, then send to blacklistednumber
; context
;
exten = fail,1,NoOp(${blacklistid})
exten = fail,2,GoTo(blacklistednumber,s,1)
[blacklistednumber]
; This is where a call will land if the macro-checkblacklist decides that
; the number should not be allowed to dial the company.
exten = s,1,Wait(2)
exten = s,2,Playback(privacy-you-are-blacklisted)
exten = s,3,Zapateller()
exten = s,4,HangUp()
On Wed, Feb 25, 2009 at 3:40 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello Everyone!
According to voip-info.org the correcy syntax for the realtime function
is:
REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read
REALTIME(family|fieldmatch|value|field) on write
It seems from the syntax that it is only possible to retrieve a full
row according to the value of only of column. This translates in SQL
language as Select * from family where fieldmath = value.
Is there any way to have more control over the realtime function?
Also, regarding the MYSQL function, I only saw documentation to query
a database. Is there any way to update a database using that
function?
Thanks!
Elliot
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[asterisk-users] Asterisk/AJAM Console
I was just looking to see if anyone knows about an open source app using the xml interface. I just started dabbling with the xml interface a little bit and it helps to look at what others are doing. I am looking for a console type app for the operator. Very simple operations like transfer, hold, status, park, etc. We are currently using the FOP, but I always have to update the fop configs to add a new button after creating/changing an extension. Our data is in a realtime DB, so I guess I could build a console that uses the realtime db and the xml interface. Anyone else in the same boat? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iPhone Sip App
Has anyone seen or know of a iphone/ipod sip client that may be in the works? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Channels Collide (Incoming Outgoing)
I have a client that is using the Sangoma A200DE with two phone lines attached. The problem is: They use their phone (Grandstream GXP2020) to dial out of the system. Instead of getting ringing, there is someone on the other end of the line that happened to dial in at the exact same moment. So now they are stuck talking with this person, instead of the one the originally called. The ZAP channels are in a dial plan context that instructs it to just dial the office phones. [zap1] exten = s,1,Dial(SIP/1001SIP/1002SIP/1003) exten = s,n,Voicemail([EMAIL PROTECTED]) Anyone know how to get around this? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom auto answer
Jerry, Did you enable Ring Answer in the phone? Look at your sip.cfg file for: alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and ringType se.rt.enabled=1 se.rt.modification.enabled=1 DEFAULT se.rt.1.name=Default se.rt.1.type=ring se.rt.1.ringer=2 se.rt.1.callWait=6 se.rt.1.mod=1/ VISUAL_ONLY se.rt.2.name=Visual se.rt.2.type=visual/ AUTO_ANSWER se.rt.3.name=Auto Answer se.rt.3.type=answer/ RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1500 se.rt.4.ringer=13 se.rt.4.callWait=6 se.rt.4.mod=1/ INTERNAL se.rt.5.name=Internal se.rt.5.type=ring se.rt.5.ringer=2 Have a look at: http://www.voicerd.org/index.php/Auto_Pickup On Mon, Apr 14, 2008 at 4:06 PM, Jerry Geis [EMAIL PROTECTED] wrote: I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call 22 and the phone rang it did not auto answer. Did I miss something? exten = 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten = 22,n,SipAddHeader(Alert-Info: Ring Answer) exten = 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten = 22,n,Set(__ALERT_INFO=Ring Answer) exten = 22,n,Set(__SIP_URI_OPTIONS=intercom=true) exten = 22,n,Dial(SIP/404) Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestions for reliable DID provider for Canada, USA and Europe
I used TelIAX for a while and was happy with the service. I used it for testing before we connected to our PRI... http://www.teliax.com On Feb 23, 2008, at 7:22 AM, Zeeshan Zakaria wrote: I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple channels per DID 3. g729 coded 4. canreinvite=yes option 5. IAX protocol Those who are already in this business, please advise me whom to go with. Is getting a virtual PRI a good solution? From their websites, they all look good so its hard to decide who is really good and will not disappear like Allo, or start giving voice quality issues. Thanks, -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HoldMusic Beep
Does anyone have a audio file they would be willing to share for on hold music? I am looking for something like the old norstar beep every few seconds. I tried 3 seconds silence, beep.wav, beep.wav. But it just didn't sound right. I need one that has a softer beep. Thanks! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum Paging Group Size?
I have done a page with at least a hundred phones before. It took about a full second for the mysql script to run and all the phones to join the conference, but worked fine. We typically only page 60 phones at once. In the coming months, I will be attempting a page with 250 phones. --- Forrest Beck http://www.shift8.biz On Jan 25, 2008, at 3:47 AM, George Pajari wrote: Has anyone experience with (or an educated guess of) the largest paging group that can be supported by the Page() command? We have an installation coming up with 110 phones -- any hope to page this entire facility? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging Recording File
I am looking to see if anyone has seen this problem before. I am
setting the MEETME_RECORDINGFILE variable in a macro, then using the r
option with the Page application to record the page. But the page is
only recorded to the file specified in MEETME_RECORDINGFILE
sometimes... Sometimes it works and sometimes it doesn't. When it
doesn't work it places the recorded file in the sounds dir with a
meetme-conf-. name. Here is my Macro.
Basically it is getting my phones that begin with a certain number
from the realtime database to create a variable with a value that ='s
SIP/6001SIP/6002SIP/6003 this is passed to the macro as ARG1
I added a System command to log the variables to a text file so I know
when the page is made, the variables are correct.
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to be paged
; ARG2 = Device for the PA system. If the user selected to
; page the PA system. That will be included.
;
exten = s,1,Set(MEETME_RECORDINGFORMAT=wav)
exten = s,2,Set(MEETME_RECORDINGFILE=custom/paging/${EPOCH})
exten = s,3,System(/bin/echo ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} $
{MEETME_RECORDINGFORMAT} ${MEETME_RECORDINGFILE} /var/log/asterisk/
pagemacro_var.log)
exten = s,4,MYSQL(Connect connid ${realdb_host} ${realdb_user} $
{realdb_pass} ${realdb_db})
exten = s,5,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\
WHERE\ name\ LIKE\ '${ARG1}%')
exten = s,6,MYSQL(Fetch fetchid ${resultid} number)
exten = s,7,GoToIf($[${fetchid} = 1]?8:10)
exten = s,8,Set(pagedevice=${pagedevice}SIP/${number})
exten = s,9,GoToIf($[${fetchid} = 1]?6:10)
exten = s,10,Set(pagedevice=${pagedevice:1})
exten = s,11,MYSQL(Clear ${resultid})
exten = s,12,MYSQL(Disconnect ${connid})
exten = s,13,GoToIf($[${ARG2} != ]?14:15)
exten = s,14,Set(pagedevice=${pagedevice}${ARG2})
exten = s,15,SIPAddHeader(Call-Info:answer-after=0)
exten = s,16,SIPAddHeader(Alert-Info: Ring Answer)
exten = s,17,NoOp(Page Recording ${MEETME_RECORDINGFILE})
exten = s,18,Set(CALLERID(all)=System Page 1010)
exten = s,19,Page(${pagedevice},r)
;On hangup, run script that will email the recording to shared
conference.
exten = h,1,System(/var/lib/asterisk/scripts/mail_lastpage ${ARG1} $
{MEETME_RECORDINGFILE})
exten = h,2,Hangup()
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Re: [asterisk-users] How to change sendmail return path
Have a look at serveremail = [EMAIL PROTECTED] and fromstring = The Asterisk PBX in voicemail.conf. On Dec 18, 2007, at 2:28 PM, shadowym wrote: Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No sound from playback and voicemail
This will also happen if there is a zap card installed and unconfigured in zaptel.conf zapata.conf. Forrest Beck [EMAIL PROTECTED] www.shift8.biz dCAP On Nov 12, 2007, at 9:46 AM, Stefan Guenther wrote: Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] exten = 99,1,ANSWER() exten = 99,2,PLAYBACK(tt-monkeys) exten = 99,3,HANGUP() The phone has access to this context, and the file exists, all codecs are allowed. I have tried to load either chan_alsa.so or chan_oss.so but it doesn't change anything. Does anyone have an idea what could be wrong? This is not the first Asterisk system that I set up, but I never had a problem like this. Asterisk is version 1.4.13 Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automating blacklists
How do you know that the call is a prank call, an not just someone
that likes calling your company alot... ?
If you just want a database of callerid's to block, here is what I
have used, I hope it helps some
My SQL table looks has 4 columns id (autoincrement), callerid,
blockenabled (to enable or disable the block), and notes.
[general]
realdb_host=localhost
realdb_user=asterisk
realdb_pass=password
realdb_db=asterisk_realtime
[pri-in]
; Conference Room Number
exten = 193,1,Answer()
exten = 193,2,Macro(checkblacklist,${CALLERID(num)})
exten = 193,3,GoTo(us-conference,s,1)
[macro-checkblacklist]
; This Macro will check the blacklist table to see if the callerid of
the
; caller exist and blockenabled =1 (TRUE). If the callerid is listed,
then
; tell the caller they have been blacklisted and politely HangUp()
;
; ${ARG1} = CallerID of incoming call
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user} $
{realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ callerid=${ARG1} and blockenabled = 1)
exten = s,3,MYSQL(Fetch fetchid ${resultid} blacklistid)
exten = s,4,MYSQL(Clear ${resultid})
exten = s,5,MYSQL(Disconnect ${connid})
exten = s,6,GoToIf($[${blacklistid} = ]?7:fail,1)
exten = s,7,NoOp(Not blocked in Blacklist)
; If the callerid is listed in the database, then send to
blacklistednumber
; context
;
exten = fail,1,NoOp(${blacklistid})
exten = fail,2,GoTo(blacklistednumber,s,1)
[blacklistednumber]
; This is where a call will land if the macro-checkblacklist decides
that
; the number should not be allowed to dial DA
exten = s,1,Wait(2)
exten = s,2,Playback(privacy-you-are-blacklisted)
exten = s,3,HangUp()
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Oct 18, 2007, at 10:25 AM, Lenz wrote:
It's not technically complex to do - you can probably use the astdb
for
that, or store all incoming numbers with timestamp in MySQL and run
something like:
SELECT count(*) 5 AS blacklisted
FROM incoming_calls
WHERE callerid = 12345
AND timestamp DATE_SUB( NOW(), INTERVAL 15 MINUTE )
you should be very well aware of the risks that can stem from such a
program - in case of bugs, or anomalous situations, you might end up
blacklisting somebody who actually needs to call in.
I hope this helps
l.
On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson
[EMAIL PROTECTED] wrote:
Hi,
I've been reading all I can on Google (and Asterisk TFOT book)
looking
for
ideas on how to implement an automated blacklist feature.
I would like to automatically blacklist a incoming number based on
timestamp
and count information.
For example, if I get a prank call from the same number 5 times
within 15
minutes, I want my dialplan to automatically blacklist this number.
Should I be looking at AGI to do something like this?
Thanks for any ideas or pointers!
--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
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Re: [asterisk-users] Paging in Asterisk
I use a mysql script to dynamically generate the page command and page about 70 phones, and I have never had a reboot problem. Sometimes there is a slight delay waiting for all the phones to join the page conference. I am using a mix of 650's, 550's, and 330's. It must only be an issue if you are using presence. Maybe I will setup presence on a couple phones and see if they reboot. Forrest Beck [EMAIL PROTECTED] http://www.shift8.biz/blog On Oct 11, 2007, at 4:34 PM, Jim Canfield wrote: Joseph Begumisa wrote: I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. Interesting. I've seen the reboot mystery mentioned before. Some have pointed to power, but this makes sense. Do you know if this is still an issue on the newer series (330,550,650) phones as well? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone using SIP trunks from Time Warner Telecom?
We use TWTelecom (NC), and the SIP trunks are available. We don't use the service however, so I can't be much help. We decided to continue using our PRI because I wanted to control echo cancellation. (6Mb EIS, 2 PRI's, and 10Mb NLAN). I will say I am extremely happy with their service. We get a couple hurricanes through the year, and their service never fails. I was told that Asterisk was supported when we looked at the service. That It was just a SIP user and Asterisk should register to it. I would assume that they would honor the QoS value in the header. since they are the next hop, they may assign there own value to the packet. They should be able to let you know, if you cal tech support. I know the QoS value is honored on our NLAN link (Point to Point between locations). I would be interested to see how you like the service, if you end up evaluating it. Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Oct 8, 2007, at 4:51 PM, Erik Anderson wrote: I am currently using a T1 PRI from TWTelecom for DID and outgoing calls, but I recently discovered that they're offering call termination/origination over SIP trunks in my area now. If they could deliver these SIP trunks to me over a guaranteed-QoS circuit, this would be of great interest to me. We're already using a DS3 circuit from TW for our internet uplink, so I'd imagine it wouldn't be difficult for them to honor the QoS flags we set on the SIP/RTP packets. Anyway - has anyone had any experience with Time Warner's SIP trunks? Officially it seems that they only support CCM and Avaya PBXs, but I'd imagine asterisk could be massaged into working just fine. Thoughts? I'm currently negotiating with our TW Sales Rep. to see if they could provision a few test DIDs on a SIP trunk so I can verify compatibility, so I *should* hopefully have answers soon for many of these questions. Thanks! -erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change verbose level
I have tried using sysconfig/asterisk but never had luck. I always just edited the safe_asterisk script. vi /usr/sbin/safe_asterisk and look for a line with -vvvc then add as many v's you want. You can also set it on the console with core set verbose 7 Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Oct 6, 2007, at 1:27 AM, Pablo Almido wrote: Hi folks, How I can change default level in asterisk from 3 level to 7level, using the script/etc/init.d/asterisk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Appliance
I am curious if anyone has used the Asterisk Appliance in an install. Where you happy with it? I have done a couple installs with 250-300 phones, but I have another one coming up that is only 7 phones. I thought I would give the Asterisk Appliance a try. Is it able to echo cancel on the FXS/FXO cards? Thanks !! Forrest Beck [EMAIL PROTECTED] www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Appliance
OK, I found the answer to my echo question (32ms). But, has anyone used it? Feelings? Forrest Beck [EMAIL PROTECTED] www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 26, 2007, at 3:07 AM, Joel Hill wrote: Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExternNotify Voicemail
Nevermind, I found the answer on the wiki: Want to run an external program whenever a caller leaves a voice mail message for a user? This is where the externnotify command comes in handy. Externnotify takes a string value which is the command line you want to execute when the caller finishes leaving a message. Note: see an example of an external notification script here. Note: This command will also run after a person who has logged into a mailbox exits the VoiceMailMain() application. The way it works is basically any time that somebody leaves a voicemail on the system (regardless of mailbox number), the command specified for externnotify is run with the arguments (in this order): context, extension, and number of voicemails in that mailbox. These arguments are passed to the program that you set in the externnotify variable. But, it would be nice to have one of the arguments be what event triggered the script. Like if it was a message was left, or some logged out of VoicemailAdmin Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 24, 2007, at 10:36 PM, Forrest Beck wrote: I have googled and can seem to find the answer to this one Does anyone here have experience with externnotify in voicemail.conf? The sample states that it will run when a message is delivered and retrieved. Does asterisk pass any arguments to the script? Thanks. Forrest Beck [EMAIL PROTECTED] www.shift8.biz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
Upgrade your kernel. Run: # uname -r if you do not see smp in the kernel version Run: # yum update kernel kernel-devel If you do see smp Run: # yum update kernel-smp kernel-smp-devel Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 25, 2007, at 10:53 AM, Tzafrir Cohen wrote: On Tue, Sep 25, 2007 at 03:22:01PM +0100, Jeng Yu wrote: Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field âownerâ specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: *** [_module_/usr/src/zaptel-1.4.5.1] Error 2 What am I missing here? Do I have to have my digium card installed first before compiling zaptel? I am running Fedora Core 5. What kernel version? uname -r -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExternNotify Voicemail
I have googled and can seem to find the answer to this one Does anyone here have experience with externnotify in voicemail.conf? The sample states that it will run when a message is delivered and retrieved. Does asterisk pass any arguments to the script? Thanks. Forrest Beck [EMAIL PROTECTED] www.shift8.biz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging MEETME_RECORDINGFILE Variable
] logger.c: -- SIP/6342-b5e0d0a0
is ringing
[Sep 21 09:18:37] VERBOSE[14309] logger.c: -- SIP/us-pa-b5e4a318
answered
[Sep 21 09:18:37] DEBUG[14309] app_meetme.c: Building dynamic
conference '177928251d'
[Sep 21 09:18:37] VERBOSE[14309] logger.c: -- Created MeetMe
conference 1021 for conference '177928251d'
[Sep 21 09:18:37] VERBOSE[14309] logger.c: Starting
recording of MeetMe Conference 177928251d into file meetme-conf-
rec-177928251d-1190380716.710.wav.
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
Begin forwarded message:
From: Forrest Beck [EMAIL PROTECTED]
Date: September 20, 2007 5:37:22 PM EDT
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Paging MEETME_RECORDINGFILE Variable
I am having a weird issue with setting the recording file for the
Page app. Here is some quick background info
I have a macro that pages all my phones:
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to be paged (6=US
Campus, 4=MS, 2=LS)
; ARG2 = Device for the PA system. If the user selected to
; page the PA system. That will be included.
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user} $
{realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip
\ WHERE\ name\ LIKE\ '${ARG1}%')
exten = s,3,MYSQL(Fetch fetchid ${resultid} number)
exten = s,4,GoToIf($[${fetchid} = 1]?5:7)
exten = s,5,Set(pagedevice=${pagedevice}SIP/${number})
exten = s,6,GoToIf($[${fetchid} = 1]?3:7)
exten = s,7,Set(pagedevice=${pagedevice:1})
exten = s,8,MYSQL(Clear ${resultid})
exten = s,9,MYSQL(Disconnect ${connid})
exten = s,10,GoToIf($[${ARG2} != ]?11:12)
exten = s,11,Set(pagedevice=${pagedevice}${ARG2})
;Add Call Info for GrandStream Phone on the PA system
exten = s,12,SIPAddHeader(Call-Info:answer-after=0)
;Add Alert-Info for all Polycom Phones
exten = s,13,SIPAddHeader(Alert-Info: Ring Answer)
exten = s,14,Set(MEETME_RECORDINGFILE=custom/paging/campuslastpage_
${RAND(1|100)})
exten = s,15,NoOp(${MEETME_RECORDINGFILE})
exten = s,16,Set(CALLERID(all)=System Page 1010)
exten = s,17,Page(${pagedevice},r)
exten = h,1,System(/var/lib/asterisk/scripts/mail_lastpage ${ARG1}
${MEETME_RECORDINGFILE})
exten = h,2,Hangup()
I call the macro with:
;Page All Phones including the PA system.
exten = 1010,1,Authenticate(12345)
exten = 1010,2,Macro(pageall,2,SIP/ls-pa)
Basically the macro goes through my sip realtime database and finds
all the phones that begin with the number 2 (my lower school
campus). The generates a variable named pagedevice that looks like
this:
SIP/2101SIP/2102SIP/2103
This part works great.
The issue I am having is setting the MEETME_RECORDINGFILE. It
should be set to an audio file in the custom sounds directory with
a random number at the end. I then use a hangup (h) extension to
execute a script (at bottom of email) to email the audio file to a
conference area in our email system (FirstClass).
What is weird is after I restart the asterisk process, this works
fine for about a week. It does exactly as it is supposed to,
creates the audio file with a random number, then the email script
delivers it. After a week or so Asterisk will stop setting the
variable MEETME_RECORDINGFILE and start placing the recordings in
the sounds directory named meetme-conf-rec.##.wav. Which is
the default is MEETME_RECORDINGFILE is not set.
Anyone seen this issue before?
Thanks!
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
#!/bin/bash
#Set some variables
USFACULTY=[EMAIL PROTECTED]
LSFACULTY=[EMAIL PROTECTED]
USFACULTY=[EMAIL PROTECTED]
MONTH=`date +%B`
DAY=`date +%d`
YEAR=`date +%Y`
HOUR=`date +%I`
MINUTE=`date +%M`
ZONE=`date +%Z`
AMPM=`date +%P`
PGSOUNDDIR=/var/lib/asterisk/sounds/
LOGFILE=/var/log/mail_lastpage.log
#Write Log
echo `date` Running script for campus $1 with file $2 $LOGFILE
#Let give asterisk time to finish creating the recordng file. Just
in Case.
sleep 10
#
#Create a temp file with our message body
#
echo Repeat Last Page /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo The attached WAV file is a copy of the last broadcast over
the phone system. /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo The page was broadcasted $MONTH $DAY, $YEAR at $HOUR:$MINUTE
$AMPM. You may play this file back if you missed the page. /tmp/
repeatpage_$1
echo /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo If you wish to mark this email as read (Remove Red Flag)
without opening the email, you may right-click (or control-click
for Mac) and left-click Mark
as Read before opening the email. /tmp/repeatpage_$1
#
#Send the email with the recorded Page attached
#
# Was it Upper School?
if [ $1 -eq 6
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
Brian, I hope this helps, it is slightly different from what you have setup. Here is what I used to connect my asterisk server to our Norstar Meridian. The Norstar T1 was setup as pri_cpe with the 24th channel as D. Check out this webpage for making a T1 crossover cable : http://www.voip-info.org/wiki/view/crossover+T1+cable You should also try making/obtaining a T1 loopback adapter and test both cards. http://www.adtran.com/adtranpx/Doc/0/FMPI9MBKGJBH39S8038BE81ID8/ CU-2abf1b83844b11d78ff20c045003.html Just curious, which cards are you using? On zapata.conf below in the norstar section, I set my timing source to be 0. This is because I am getting timing from the first span. If you are only connecting the two machines together and neither one has other T1's going to a telco. Then just set one of them as primary (1) and the other as do not use (0). span=(spannum),(timing),(LBO),(framing),(coding) Zaptel.conf -- loadzone=us defaultzone=us #PRI to TimeWarner span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #PRI to Norstar Meridian span=2,0,0,esf,b8zs bchan=25-47 dchan=48 #TDM800 Card fxoks=49-52 fxsks=53-56 Zaptel.conf -- [channels] ;Time Warner T1 (SPAN 1) context=tw-pri-in group=2 ;Echo Cancel turns on the Octastic Hardware Canceller on the card echocancel=yes ; Tell Asterisk to reset the channels every two hours resetinterval = 7200 ; Relax DTMF relaxdtmf=yes rxgain=0.0 txgain=0.0 signalling = pri_cpe switchtype = national channel = 1-23 ;Norstar T1 (SPAN 2) context=norstar group=3 signalling = pri_net channel = 25-47 Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 21, 2007, at 9:31 AM, Brian Alexander wrote: On 9/20/07, Jared Smith [EMAIL PROTECTED] wrote: I'd look at your wiring, as an HDLC error like that is usually an indication of some type of a problem at the physical layer. I tried two other cables this morning with the same results... While I was swapping cables around I unloaded asterisk and the zaptel modules. One thing I noticed once everything was hooked up and restarted is that Machine2 shows a red alarm but Machine1 shows both spans as clear of alarms... Otherwise everything looks the same as yesterday. Have any of you connected two asterisk machines by t1 crossover using pri_net/pri_cpe signaling? I am completely stumped and would love to know that some had done this and what their configuration look like. Thanks again, -Brian ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging MEETME_RECORDINGFILE Variable
I am having a weird issue with setting the recording file for the
Page app. Here is some quick background info
I have a macro that pages all my phones:
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to be paged (6=US
Campus, 4=MS, 2=LS)
; ARG2 = Device for the PA system. If the user selected to
; page the PA system. That will be included.
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user} $
{realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\
WHERE\ name\ LIKE\ '${ARG1}%')
exten = s,3,MYSQL(Fetch fetchid ${resultid} number)
exten = s,4,GoToIf($[${fetchid} = 1]?5:7)
exten = s,5,Set(pagedevice=${pagedevice}SIP/${number})
exten = s,6,GoToIf($[${fetchid} = 1]?3:7)
exten = s,7,Set(pagedevice=${pagedevice:1})
exten = s,8,MYSQL(Clear ${resultid})
exten = s,9,MYSQL(Disconnect ${connid})
exten = s,10,GoToIf($[${ARG2} != ]?11:12)
exten = s,11,Set(pagedevice=${pagedevice}${ARG2})
;Add Call Info for GrandStream Phone on the PA system
exten = s,12,SIPAddHeader(Call-Info:answer-after=0)
;Add Alert-Info for all Polycom Phones
exten = s,13,SIPAddHeader(Alert-Info: Ring Answer)
exten = s,14,Set(MEETME_RECORDINGFILE=custom/paging/campuslastpage_$
{RAND(1|100)})
exten = s,15,NoOp(${MEETME_RECORDINGFILE})
exten = s,16,Set(CALLERID(all)=System Page 1010)
exten = s,17,Page(${pagedevice},r)
exten = h,1,System(/var/lib/asterisk/scripts/mail_lastpage ${ARG1} $
{MEETME_RECORDINGFILE})
exten = h,2,Hangup()
I call the macro with:
;Page All Phones including the PA system.
exten = 1010,1,Authenticate(12345)
exten = 1010,2,Macro(pageall,2,SIP/ls-pa)
Basically the macro goes through my sip realtime database and finds
all the phones that begin with the number 2 (my lower school
campus). The generates a variable named pagedevice that looks like
this:
SIP/2101SIP/2102SIP/2103
This part works great.
The issue I am having is setting the MEETME_RECORDINGFILE. It should
be set to an audio file in the custom sounds directory with a random
number at the end. I then use a hangup (h) extension to execute a
script (at bottom of email) to email the audio file to a conference
area in our email system (FirstClass).
What is weird is after I restart the asterisk process, this works
fine for about a week. It does exactly as it is supposed to, creates
the audio file with a random number, then the email script delivers
it. After a week or so Asterisk will stop setting the variable
MEETME_RECORDINGFILE and start placing the recordings in the sounds
directory named meetme-conf-rec.##.wav. Which is the default is
MEETME_RECORDINGFILE is not set.
Anyone seen this issue before?
Thanks!
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
#!/bin/bash
#Set some variables
USFACULTY=[EMAIL PROTECTED]
LSFACULTY=[EMAIL PROTECTED]
USFACULTY=[EMAIL PROTECTED]
MONTH=`date +%B`
DAY=`date +%d`
YEAR=`date +%Y`
HOUR=`date +%I`
MINUTE=`date +%M`
ZONE=`date +%Z`
AMPM=`date +%P`
PGSOUNDDIR=/var/lib/asterisk/sounds/
LOGFILE=/var/log/mail_lastpage.log
#Write Log
echo `date` Running script for campus $1 with file $2 $LOGFILE
#Let give asterisk time to finish creating the recordng file. Just
in Case.
sleep 10
#
#Create a temp file with our message body
#
echo Repeat Last Page /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo The attached WAV file is a copy of the last broadcast over the
phone system. /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo The page was broadcasted $MONTH $DAY, $YEAR at $HOUR:$MINUTE
$AMPM. You may play this file back if you missed the page. /tmp/
repeatpage_$1
echo /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo If you wish to mark this email as read (Remove Red Flag)
without opening the email, you may right-click (or control-click for
Mac) and left-click Mark
as Read before opening the email. /tmp/repeatpage_$1
#
#Send the email with the recorded Page attached
#
# Was it Upper School?
if [ $1 -eq 6 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav -
s Recording of Last Page for Upper School $USFACULTY
fi
# Was it Middle School?
if [ $1 -eq 4 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav -
s Recording of Last Page for Middle School $MSFACULTY
fi
# How about Lower?
if [ $1 -eq 2 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav -
s Recording of Last Page for Lower School $LSFACULTY
fi
rm -rf /tmp/repeatpage_$1
rm -f $PGSOUNDDIR$2.wav
exit
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
Re: [asterisk-users] How to cancel the password check in VoicemailMain()
Actually in 1.4 the s option should be at the end. on your CLI type core show application VoiceMailMain [Synopsis] Check Voicemail messages [Description] VoiceMailMain([EMAIL PROTECTED]|options]): This application allows the calling party to check voicemail messages. A specific mailbox, and optional corresponding context, may be specified. If a mailbox is not provided, the calling party will be prompted to enter one. If a context is not specified, the 'default' context will be used. Options: p- Consider the mailbox parameter as a prefix to the mailbox that is entered by the caller. g(#) - Use the specified amount of gain when recording a voicemail message. The units are whole-number decibels (dB). s- Skip checking the passcode for the mailbox. a(#) - Skip folder prompt and go directly to folder specified. Defaults to INBOX exten = 99,n,VoiceMailMain([EMAIL PROTECTED],s) Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 19, 2007, at 12:03 PM, Mark Michelson wrote: rrgv wrote: Hi in asterisk 1.4, I need to cancel the password check and allow users enter in the mailbox without entering password. I tried this: exten = 99,1,Set(LANGUAGE()=es) exten = 99,n,VoicemailMain([EMAIL PROTECTED],s) exten = 99,n,Hangup and this: exten = 99,1,Set(LANGUAGE()=es) exten = 99,2,VoicemailMain(s) exten = 99,n,Hangup The syntax is a bit off on your VoiceMailMain call. Change it to this and see if it helps: exten = 99,n,VoiceMailMain([EMAIL PROTECTED]) In other words, put the 's' at the beginning of the argument as opposed to a separate option. Mark Michelson ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limiting Simultaneous calls
You mean in sip.conf? Look at adding to your voip providers peer/user config incominglimit, outgoinglimit or call-limit: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf --- Forrest Beck www.shift8.biz On Sep 18, 2007, at 4:26 PM, Jim Boykin wrote: Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable MoH for certain phones
You can define a new class in musiconhold.conf with an empty directory. Create a directory with nothing in it /var/lib/asterisk/moh/empty Add this class to musiconhold.conf [empty] mode=files directory=/var/lib/asterisk/moh/empty Then for the phone' entry in sip.conf add: musiconhold=empty Be sure to conect to the CLI and do sip reload On 8/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Question
You can eliminate the set CallerID line. This will just set the
variable back to itself. Asterisk will pass the callerid from one
span to the next.
You can use a GotoIF to set the callerid to something else if it is
blank or marked as Private:
exten = s,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = s,2,Set(CALLERID(num)=00)
If the CallerID number is blank go to 2 else go to 3.
I wonder if asterisk or the norstar system is holdng on to that last
callerid number on the channel?
The only time you may want to set callerid is when your Norstar dials
out through Asterisk:
[norstar]
; This context is where all incoming calls from the norstar are placed
; Basically take the call from the norstar and bridge it over to the first
; available line on the bottom of the T1 going to TimeWarner.
exten = _1900XXX,1,Playback(cannot-complete-as-dialed)
exten = _1900XXX,2,Hangup()
exten = _1X.,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = _1X.,2,Set(CALLERID(num)=511212)
exten = _1X.,3,NoOp(${CALLERID(num)})
exten = _1X.,4,Dial(${PRITRUNK}/${EXTEN},300,tD())
exten = _1X.,5,Hangup()
exten = _X.,1,GoToIf($[${CALLERID(num)} = ]?2:3)
exten = _X.,2,Set(CALLERID(num)=511212)
exten = _X.,3,NoOp(${CALLERID(num)})
exten = _X.,4,Dial(${PRITRUNK}/${EXTEN},300,)
exten = _X.,5,Hangup()
exten = i,1,Answer()
exten = i,n,Wait(1)
exten = i,n,Playback(cannot-complete-as-dialed)
exten = i,n,Playback(please-contact-tech-supt)
exten = i,n,Hangup()
On 8/9/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
hmm from what i have seen this is not supposed to be.. the info is still
there but should not be used in case of privacy..
zap show channels always show last info till a span refresh.. but the
privacy should indeed replace those with Privacy.
Maybe it could be a bug ,
On 8/9/07, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco,
Span 2 sends to my existing phone system(Nortel).
My Span1 gets sent to the context from-pri, detailed here:
[from-pri]
exten = _49XX,1,Set(CALLERID(all)=${CALLERID(all)})
exten = _49XX,2,Dial(Zap/g2/${EXTEN},,twk)
exten = _49XX,3,Congestion()
exten = _49XX,4,Set(CALLERID(all)=)
exten = _49XX,5,Hangup()
exten = _49XX,103,Congestion()
exten = _49XX,104,Set(CALLERID(all)=)
exten = _49XX,105,Hangup()
exten = h,1,Set(CALLERID(all)=)
exten = h,2,Hangup()
I'm receiving caller ID fine, and setting it on the outgoing channel the
same I received it, is my logic above wrong? Will Asterisk natively pass
through the caller ID, or is there a better way to set it?
The reason I ask, is that calls that are not coming in with CLID(blocked
or private) are showing up as the same number that was previously answered
on that channel.
Thanks.
Using Asterisk 1.4 FYI.
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Re: [asterisk-users] CDR/MySQL basic config
Adrian, What host/ip did you specify when you created the user? # mysql --user=root --password #mysql use mysql; #mysql select Host from user where User = 'asteriskcdruser' (this line is case sensitive) Does it return 127.0.0.1 or localhost. Make cdr_mysql reflect that. You should also check out cdr_odbc, asterisk can connect through an ODBC connection which in turn is a connection to the MySQL database. There seems to be more suport for the ODBC driver. Hope this helps some On 8/6/07, Adrian Marsh [EMAIL PROTECTED] wrote: Hi, I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The add-ons pack has been installed for a while, so now I'm trying to add the Mysql config. I've created a mysql database, added the grants for a user acces, and can run a mysql -u asteriskcdruser -p and can connect to the database. I've been using this as a guide: http://www.757.org/~joat/wiki/index.php/Asterisk#Viewing_CDR_Data_with_A sterisk:_CDR_Analyzer I've created cdr_mysql.conf: [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=password user=asteriskcdruser port=3306 sock=/tmp/mysql.sock userfield=1 But when I start asterisk (1.4 on my test machine), I get: == Parsing '/etc/asterisk/cdr_mysql.conf': Found [Aug 6 21:01:14] ERROR[32512]: cdr_addon_mysql.c:436 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. cdr_addon_mysql.so = (MySQL CDR Backend) [Aug 6 21:01:14] ERROR[32512]: res_config_mysql.c:627 mysql_reconnect: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. [Aug 6 21:01:14] WARNING[32512]: res_config_mysql.c:474 load_module: MySQL RealTime: Couldn't establish connection. Check debug. [Aug 6 21:01:14] NOTICE[32512]: config.c:1171 ast_config_engine_register: Registered Config Engine mysql MySQL RealTime driver loaded. res_config_mysql.so = (MySQL RealTime Configuration Driver) I'm also looking as to what CDR viewers there are available, and which people think are best. I want to view/report on the calls made within A*k. Thanks, Adrian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TimeStamp a Recording
Has anyone come up with to timestamp a Recording? I am using a pretty simple dialplan to record a audio file for a hotline. I'd like to store the date and time it was recorded somewhere, Ast DB or MySQL DB. Then when the audio file is played back to a caller, the system will say something like. This message was recorded January 14th at 10 42 pm Thanks for any ideas you may have. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G722 and Polycom 550
Has anyone found a way to enable the g722 codec as a prefered codec in the Polycom provisioning files for the 550's? I couldn't find a pref for voice.codecPref.IP_550. What needs to be put into the allow field (sip.conf) for asterisk to allow the codec? -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MYSQL Query -- PAGE
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from the MySQL query will be multiple rows of extension:
Something like:
mysql Select extension from sip where extension like '6%'
6001
6002
6003
ex
I need to put all the results into a variable that would equal something like:
SIP/6001SIP/6002SIP/6003
I have setup a couple basic MYSQL Query's for my dialplan. Mostly
just looking up a DID to Extension Mapping for setting callerid on
outbound and inbound calls.
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
Something like Set(devices=${devices}${newrow_result})
I looked at the example on
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL but that doesn't
seem to be accurate.
Thanks all!!
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Re: [asterisk-users] MYSQL Query -- PAGE
Well This seems to work.
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to be paged (US Campus=6,
LS Campus=2)
; ARG2 = Device for the PA system. If the user selected to
; page the PA system. That will be included.
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user}
${realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\
WHERE\ name\ LIKE\ '${ARG1}%')
exten = s,3,MYSQL(Fetch fetchid ${resultid} number)
exten = s,4,GoToIf($[${fetchid} = 1]?5:8)
exten = s,5,Set(pagedevice=${pagedevice}SIP/${number})
exten = s,6,NoOp(${number})
exten = s,7,GoToIf($[${fetchid} = 1]?3:8)
exten = s,8,Set(pagedevice=${pagedevice:1})
exten = s,9,NoOp(PageDevice ${pagedevice})
exten = s,10,MYSQL(Clear ${resultid})
exten = s,11,MYSQL(Disconnect ${connid})
exten = s,12,GoToIf($[${ARG2} != ]?13:14)
exten = s,13,Set(pagedevice=${pagedevice}${ARG2})
exten = s,14,Set(_ALERT_INFO=RA)
exten = s,15,Page(${pagedevice})
exten = s,16,Hangup()
On 5/8/07, Remco Post [EMAIL PROTECTED] wrote:
Forrest Beck wrote:
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from the MySQL query will be multiple rows of extension:
Something like:
mysql Select extension from sip where extension like '6%'
6001
6002
6003
ex
I need to put all the results into a variable that would equal something
like:
SIP/6001SIP/6002SIP/6003
I have setup a couple basic MYSQL Query's for my dialplan. Mostly
just looking up a DID to Extension Mapping for setting callerid on
outbound and inbound calls.
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
Something like Set(devices=${devices}${newrow_result})
I looked at the example on
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL but that doesn't
seem to be accurate.
Thanks all!!
What I've done in postgresql is to build an pl/pgsql procedure that
returns the desired dialstring. So the procedure does the select and
then concats them.
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] SIP RealTime Friends
Let me check my table Voicemail and CDR in the MySQL database works fine. sip show peers isn't giving me anything. Only the one peer I left setup in sip.conf Here Is what I get from a Dial Command: [May 4 09:14:28] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argumen On 5/4/07, Sergio (Red) [EMAIL PROTECTED] wrote: Hi, Do you know how see the peers statuses like: sip show peers but when sip peers are configured by Relatime method. Thanks 0xception escribió: yes you can use the type friend On 5/3/07, *Forrest Beck* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Con Netfono, puede hablar por telefono, de PC a PC y gratis ! Instale su Netfono desde http://www.netfono.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP RealTime Friends
Nevermind. Friday and my mind has gone home! :) I forgot the ipaddr and port setting in the table. On 5/4/07, Forrest Beck [EMAIL PROTECTED] wrote: Let me check my table Voicemail and CDR in the MySQL database works fine. sip show peers isn't giving me anything. Only the one peer I left setup in sip.conf Here Is what I get from a Dial Command: [May 4 09:14:28] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argumen On 5/4/07, Sergio (Red) [EMAIL PROTECTED] wrote: Hi, Do you know how see the peers statuses like: sip show peers but when sip peers are configured by Relatime method. Thanks 0xception escribió: yes you can use the type friend On 5/3/07, *Forrest Beck* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Con Netfono, puede hablar por telefono, de PC a PC y gratis ! Instale su Netfono desde http://www.netfono.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNow!
Type CTRL-ALT- F2 (or F3 or F4 or F5) for the different tty's. You just have to login as admin as root is disabled. You can then use su to gain root access. On 5/4/07, Ed Nuñez [EMAIL PROTECTED] wrote: Does anyone know how to gain access directly to the configuration files in AsteriskNow? I have dual NICs and need to change the binding in the config file. I believe they blocked ssh2 access by default. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compile error
The problem is that your kernel is newer than the xbus-core.c file is looking for. See: http://forums.digium.com/viewtopic.php?t=15317sid=7beaf6bfed1550f4a8676427283800c5 I just did a make menuselect and eliminated the xpp module. It is for USB Astribank, something I will never use. On 5/4/07, mail-lists [EMAIL PROTECTED] wrote: I get the following error when trying to compile zaptel on CentOS 5 kernel 2.6.18-8.1.3.el5 CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: â has no member named â make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1 make[2]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp] Error 2 make[1]: *** [_module_/root/asterisk-src/zaptel-1.2.17.1] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-8.1.3.el5-i686' make: *** [all] Error 2 I'm kind of at my wits end with this - been trying for several hours.. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zttranscode crashes server
I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built zaptel/asterisk/libpri (all latest releases as of May 3rd). I am also using the /etc/sysconfig/zaptel file to only specify the two modules I do need. wct4xxp and wctdm24xxp. I am using a TE212P and a TDM844B card. I shouldn't need the zttransode module since I don't have a codec translation card. right? To work around this I added zttranscode to RMODULES in the zaptel init script. If I don't need the zttranscode module. I may try and rebuild zaptel without it. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zttranscode crashes server
So is anyone not using the zaptel init script to load modules? Anyone using modules.conf? How an I load them at boot without using the init script? Do I just remove --ignore-install from modprobe? Thanks On 5/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, May 03, 2007 at 08:38:10AM -0400, Forrest Beck wrote: I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built zaptel/asterisk/libpri (all latest releases as of May 3rd). I am also using the /etc/sysconfig/zaptel file to only specify the two modules I do need. wct4xxp and wctdm24xxp. Strangely enough, this issue exists in 1.4, but not in 1.2. Compare: http://svn.digium.com/svn/zaptel/branches/1.2/zaptel.init http://svn.digium.com/svn/zaptel/branches/1.4/zaptel.init Note the unload_module function in 1.2 (yicks: recursive functions in bourne shell) I am using a TE212P and a TDM844B card. I shouldn't need the zttransode module since I don't have a codec translation card. right? Right. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zttranscode crashes server
sorry, I meant modprobe.conf On 5/3/07, Forrest Beck [EMAIL PROTECTED] wrote: So is anyone not using the zaptel init script to load modules? Anyone using modules.conf? How an I load them at boot without using the init script? Do I just remove --ignore-install from modprobe? Thanks On 5/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, May 03, 2007 at 08:38:10AM -0400, Forrest Beck wrote: I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built zaptel/asterisk/libpri (all latest releases as of May 3rd). I am also using the /etc/sysconfig/zaptel file to only specify the two modules I do need. wct4xxp and wctdm24xxp. Strangely enough, this issue exists in 1.4, but not in 1.2. Compare: http://svn.digium.com/svn/zaptel/branches/1.2/zaptel.init http://svn.digium.com/svn/zaptel/branches/1.4/zaptel.init Note the unload_module function in 1.2 (yicks: recursive functions in bourne shell) I am using a TE212P and a TDM844B card. I shouldn't need the zttransode module since I don't have a codec translation card. right? Right. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP RealTime Friends
I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1/E1 Configuration
You can remove the extra context in zapata. signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 98 (remove this line) context=default The way zapata works is you define options for the channels, then specify the channels the options are for. Context=default will just add this option to the next channel below. Also, If you don't want to use a group on the channel, just eliminate the line all together instead of group=0 Does ztcfg run without errors? What does zap show channels display in the asterisk CLI? Are you running asterisk as root using asterisk -vvc and it still fails to load? Does lsmod show the driver modules loaded? On 5/3/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, Can anyone please post their working T1/E1 configuration... Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if you run 'genzaptelconf' it created '/etc/asterisk/zapata-channels.conf', so please post that one also. Here is my configuration which is failing Asterisk to load... I have two cards TE405P and TDM400P: - === /etc/zaptel.conf === # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 fxsks=97 fxsks=98 fxsks=99 fxsks=100 # Global data loadzone= us defaultzone = us /etc/asterisk/zapata-channels.conf group = 1 switchtype = national signalling = pri_cpe context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_cpe context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_cpe context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 97 ; context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 98 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 99 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 100 context=default Thanking in advance... Cheers, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail on Different Server
I have two seperate systems at two different locations. Each hosts
there own voicemail for their phones.
I have thought about just having all voicemail on one server. Is the
best way to do this just through a dial app?
For example, if someone dials 1000 to check voicemail at site A. The
dialplan will be something like this on Site A:
[context-for-phones-at-one-location]
exten = 1000,1,Dial(SIP/voicemailserver/${EXTEN})
Then on Site B where the voicemail is to be stored:
[context-for-incoming-voicemail]
exten = 1000,1,Voicemail(@vmcontext)
exten = o,1,Dial(SIP/siteAserver/receptionistextension
Can anyone think of draw backs to this? One I can think of is I will
have to specify a extension to redirect 0 (for receptionist) back to
the Site A server. I will also have to redirect all directory apps to
the voicemail server.
Does anyone do this? How do you handle it?
Thanks.
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Re: [asterisk-users] Voicemail on Different Server
I've heard there are problems using NFS as a storage device.??? On 4/24/07, Anthony Rodgers [EMAIL PROTECTED] wrote: Why not export an NFS mount from one server to the other? That's what we do. CP -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Tuesday, April 24, 2007 5:28 PM To: Asterisk Users List Subject: [asterisk-users] Voicemail on Different Server I have two seperate systems at two different locations. Each hosts there own voicemail for their phones. I have thought about just having all voicemail on one server. Is the best way to do this just through a dial app? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing CallerID
I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In sip.conf I am using CallerID as their internal number. I thought of maybe adding a key for each extension to the astdb and have a Macro query the astdb. Any other ideas? Thanks. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing CallerID
Thanks. I will just use the asterisk database.
This brought up a new question though.
This is what I am using to dial out. If a key for the phone exist in
the db it will assign it the did specified. If not, just assign the
main incoming operator number.
I have a new family in the astdb named external_did. Some entries
look like this:
external_did/2503/9195551212
external_did/2505/9195551213
[macro-dialoutpstn]
exten = s,1,Set(curextension=${CALLERID(num)})
exten = s,2,Set(dbdid=${DB(external_did/${curextension})})
exten = s,3,NoOp(${curextension})
exten = s,4,NoOp(${dbdid})
exten = s,4,GoToIf($[${dbdid} = ]?5:8)
exten = s,5,Set(CALLERID(num)=9195559595)
exten = s,6,Dial(SIP/mspri/${MACRO_EXTEN:1},300)
exten = s,7,Hanup()
exten = s,8,Set(CALLERID(num)=${DB(external_did/${curextension})})
exten = s,9,NoOp(${CALLERID(num)})
exten = s,10,Dial(SIP/mspri/${MACRO_EXTEN:1},300)
exten = s,11,Hangup()
This works just fine.
Now what about my incoming calls. My incoming calls will be sent from
the telco to asterisk as the seven digit number that was dialed. So
if I have _X. in my context it will be processed as extension
9195551212.
So is there a way to lookup in the asterisk database a value and
return the key it belongs to? Because I already have the phone number
in the asterisk database set to each extension.
I know I could just create a new family and add the keys there, like so
incoming_did/9195551212/2503
incoming_did/9195551213/2504
I was just looking to see if I could save myself a step.
This may be where I will need to switch to MySQL.
On 4/19/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Thu, 19 Apr 2007, Forrest Beck said something to this effect:
I thought of maybe adding a key for each extension to the astdb and
have a Macro query the astdb. Any other ideas?
That would work, and is certainly the easiest, since you can bulk-load
the DID - extension maps via external CLI commands with a simple script.
You could also have Asterisk do MySQL dips for this information, if the
desire is to administer it from a web-based front-end. Or if there is
some sort of mathematical relationship between the extension and the DID
range, the dialplan interpreter itself is capable of fairly sophisticated
mathematical extrapolations.
-- Alex
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Re: [asterisk-users] Nagios asterisk monitoring
I have a script as well. This actually may be yours Brandon. I found
it through Google. It will just open a telnet session to the manager
interface and count the ZAP and SIP channels. You just have to call
the script through a OID in snmp.conf. Works really well with Cacti.
I will forward it and how it is setup if you like.
http://picasaweb.google.com/jonforrest.beck/AsteriskCLI/photo#5052274842733411794
On 4/11/07, Brandon Kruse [EMAIL PROTECTED] wrote:
I wrote a very extensive plugin for cacti to monitor asterisk.
It uses the manager interface to poll and get statistics for 1.4 and 1.2.
Let me know if you interested, ill post it, or email me directly.
-bkruse
voip crazy wrote:
Dear list,
I am trying to configure the nagios plugin called check_sip. I just
read the README file included with the plugin. I follow the readme
steps to configure the plugin, without success. In the nagios web
interface I can see (No output!) In the status information column. If
I run the chech_sip plugin from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
SIP 200 OK: 0.00 second response time
I do not know why If I run the plugin from the consle it works ok, but
if I run it from Nagios web interface it does not run.
Anyone are using this plugin?
Could you helpme to solve that?
Any clue will be appreciated.
Thanks for your time.
VoipCrazy
Here goes my nagios check_sip plugin configuration.
define command{
command_namecheck_sip
command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$ -w 5
}
define service{
use generic-service
host_name -PBX
service_description SIP test
check_command check_sip!sip:[EMAIL PROTECTED]
contact_groups admins
max_check_attempts 4
normal_check_interval 5
retry_check_interval1
notification_interval 240
check_period24x7
notification_period 24x7
notification_optionsc,r
}
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[asterisk-users] What is your Backup Strategy?
I was just curious to what your redundancy solution is. I have considered many options, so I thought I would share and get an idea for what others are doing. My setup is two different locations with a 10MB WLAN fiber link between the two. Each location has it's own PRI as well. I have considered and tested many options this last year or so. 1) Using hearbeat and drbd to monitor the servers. When the primary fails the backup will assign itself the virtual ip used between the two, and then mount the drbd disk which has the asterisk configs and voicemail. The biggest con to this is hearbeat just monitors a ping response either over IP or a COM port. So if the asterisk service dies, heartbeat will not fail over. Although I think there are work arounds for this. The newest version is suppose to have support for monitoring a TCP port as well 2) Have two servers with the same dialplan. One in each location. Each server has it's own TDM cards installed. Phones on Site A will register with the server on Site A, and phones on Site B will register with the server on Site B. Then using Polycom phones, they will failover to using the server not on their site, if their primary isn't available. I have setup scripts to copy the dialplan from one server to the other then reload asterisk nightly. The biggest Con to this is I have to be sure my dialplans don't get different. The user's voicemail wouldn't be available until their primary server is back up, but that's OK. 3) Having a main asterisk server and a smaller VoIP gateway at each site. The gateway is a small 14inch deep rack server with a P4 and 1Gig RAM running asterisk. It will host the TDM cards, and just handle traffic to/from the PRI. The main asterisk server will just see it as a SIP trunk. The failover here is that the polycom phones will register with the gateway if the primary server isn't available. They won't have all the features and voicemail, but at least they can dial out and get 911 if needed. What do you think? Do you have a better solution? Thanks!! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is your Backup Strategy?
We are looking at about 200 total phones with low usage. Probably only 20 or so calls at once. On 4/11/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: On 4/11/07, Forrest Beck [EMAIL PROTECTED] wrote: 2) Have two servers with the same dialplan. One in each location. Each server has it's own TDM cards installed. Phones on Site A will register with the server on Site A, and phones on Site B will register with the server on Site B. Then using Polycom phones, they will failover to using the server not on their site, if their primary isn't available. I have setup scripts to copy the dialplan from one server to the other then reload asterisk nightly. The biggest Con to this is I have to be sure my dialplans don't get different. The user's voicemail wouldn't be available until their primary server is back up, but that's OK. Now you said you had two machines with the same dialplans. What happens when you go into fail over an someone leaves a voice message and it gets stuck on the other server? I think the key here is to treat functions as a cluster. IVRs, voicemail, phone calls, etc you need to have a redundant solution for each, not just a spare or redundant asterisk server. Then again you could be working on a small scale project where what I describe its not really important. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom
I know this doesn't belong on this list but... I am looking to see if anyone is using Polycom and knows of a web based software for creating/managing the cfg files for polycom phones. I see that the AsteriskNow will add provisioning support for Polycom phones. Since it is still in beta, I was just looking to see if there was anything else out there. Thanks! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging
First off, A lot of thanks to this list. I have learned ton from reading through the posts this past year. I need some advise. I have two group of phones connected to a single server. Group1= SIP/2503SIP/2504 Group2=SIP/3501SIP/3502 I'd like to be able to dial an extension and page a certain group of phones only if ChanIsAvail returns 1. I am not sure how to go about programming this. I though to write a AGI script that reads a list of phones (one list per group), checks ChanIsAvail then Pages the phone. I will have about 60 extensions per group to Page. Will there be lag until all the phones get paged and the script finishes? Then I thought maybe a Macro in the dialplan to dial a global var of the group of phones, but that won't work. If phone isn't available, none will get paged. Has anyone done this before? I just don't know where to start. Thanks -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Paging
Forgot to mention. We are using Polycom phones on asterisk 1.4.2 I tried the allpage agi, but it checks for all SIP peers connected to the server. On 3/30/07, Forrest Beck [EMAIL PROTECTED] wrote: First off, A lot of thanks to this list. I have learned ton from reading through the posts this past year. I need some advise. I have two group of phones connected to a single server. Group1= SIP/2503SIP/2504 Group2=SIP/3501SIP/3502 I'd like to be able to dial an extension and page a certain group of phones only if ChanIsAvail returns 1. I am not sure how to go about programming this. I though to write a AGI script that reads a list of phones (one list per group), checks ChanIsAvail then Pages the phone. I will have about 60 extensions per group to Page. Will there be lag until all the phones get paged and the script finishes? Then I thought maybe a Macro in the dialplan to dial a global var of the group of phones, but that won't work. If phone isn't available, none will get paged. Has anyone done this before? I just don't know where to start. Thanks -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro Dial - External DID
I am using the sample (slightly modified) macro for dialing phones. My
extensions are in the 2000 range. Each extension has it's own
external DID. Is there any way to have the macro look up the DID to
be used for the extension and set the DID as the callerid? Maybe from
a flat file somewhere? Or is there a better way to do this???
I know I can use callerid in sip.conf, but I only want the DID used
when the call goes external.
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED]) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED]) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
[phones]
exten = _2XXX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
DID example:
2001 = 5552871701
2002 = 5552871702
2003 = 5552871703
Thanks!
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[asterisk-users] Server Recomendation
I am looking to install a system with 200 phones (polycom). There will be about 30-40 simultaneous calls. I am looking at the Dell 1950 with Quad 2.66, 2Gig RAM, Two 160 Gig SATA Drives (Mirrored with a Perc5 card), Dual Gig NIC, and RHEL 4.0. I will use two gateways for my PRI's and FXS Cards so PCI won't be used. I will probably use a small 14 2U server to handle the ZAP Cards. Does anyone for see a problem with using the 1950? Good/Bad thoughts??? Thanks! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.1
Any idea when 1.4.1 will be available. There is a bug fix in the cvs head that I need, and I don't want to run the cvs build on a production machine. Thanks... -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Billing Telephone Number (BTN)
I have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Marks SNMP HowTo
I followed Marks SNMP howto on Voip Magazine and ran into a small problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/) When asterisk is running as a non-root user (asterisk) SNMP request for for the Asterisk MIB tree return nothing. If I quit asterisk and run it as root, all is fine. Does anyone have a idea what is going on? I have never used agentX, so I am unsure of what it is doing. Does it bind to a particular port that maybe my asterisk user does not have permission to access??? Here is my snmpd.conf file: master agentx agentXPerms 0660 0550 asterisk asterisk com2sec local localhost da_public com2sec mynetwork 10.11.0.0/16 da_public com2sec dmz 172.17.0.0/16 da_public group MyROGroup any local group MyROGroup any mynetwork group MyROGroup any dmz view all included .1 access MyROGroup any noauth 0 all none none and here is res_snmp.conf [general] subagent = yes enabled = yes Thanks all.! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Marks SNMP HowTo
OK. problem solved. It was something dumb on my part. /var/agentx didn't have enough permissions to let asterisk access the socket. On 2/25/07, Forrest Beck [EMAIL PROTECTED] wrote: I followed Marks SNMP howto on Voip Magazine and ran into a small problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/) When asterisk is running as a non-root user (asterisk) SNMP request for for the Asterisk MIB tree return nothing. If I quit asterisk and run it as root, all is fine. Does anyone have a idea what is going on? I have never used agentX, so I am unsure of what it is doing. Does it bind to a particular port that maybe my asterisk user does not have permission to access??? Here is my snmpd.conf file: master agentx agentXPerms 0660 0550 asterisk asterisk com2sec local localhost da_public com2sec mynetwork 10.11.0.0/16 da_public com2sec dmz 172.17.0.0/16 da_public group MyROGroup any local group MyROGroup any mynetwork group MyROGroup any dmz view all included .1 access MyROGroup any noauth 0 all none none and here is res_snmp.conf [general] subagent = yes enabled = yes Thanks all.! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Marks SNMP HowTo
Sergio, The resource has to be compiled at install. If net-snmp is installed along with a couple other packages, then it will be installed. To see if it is there now type module show like snmp in the cli. (1.4.0) Here is a how to. http://www.voip-magazine.com/content/view/2877/ On 2/25/07, Sergio R. D'Ippolito [EMAIL PROTECTED] wrote: How can i see if snmp is running ok on mi * box ? Thanks in advance -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Forrest Beck Enviado el: Domingo, 25 de Febrero de 2007 06:14 p.m. Para: Asterisk Users List Asunto: [asterisk-users] Re: Marks SNMP HowTo OK. problem solved. It was something dumb on my part. /var/agentx didn't have enough permissions to let asterisk access the socket. On 2/25/07, Forrest Beck [EMAIL PROTECTED] wrote: I followed Marks SNMP howto on Voip Magazine and ran into a small problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/) When asterisk is running as a non-root user (asterisk) SNMP request for for the Asterisk MIB tree return nothing. If I quit asterisk and run it as root, all is fine. Does anyone have a idea what is going on? I have never used agentX, so I am unsure of what it is doing. Does it bind to a particular port that maybe my asterisk user does not have permission to access??? Here is my snmpd.conf file: master agentx agentXPerms 0660 0550 asterisk asterisk com2sec local localhost da_public com2sec mynetwork 10.11.0.0/16 da_public com2sec dmz 172.17.0.0/16 da_public group MyROGroup any local group MyROGroup any mynetwork group MyROGroup any dmz view all included .1 access MyROGroup any noauth 0 all none none and here is res_snmp.conf [general] subagent = yes enabled = yes Thanks all.! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.441 / Virus Database: 268.18.3/699 - Release Date: 23/02/2007 01:26 p.m. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.4/702 - Release Date: 25/02/2007 03:16 p.m. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automatic Dial, Play message
Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? For example, you have a database of FirstName LastName PhoneNumber Jon -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automatic Dial, Play message
Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? Just for example, you have a database of FirstName, LastName, PhoneNumber Jon, Beck, 9194713175 So it would pull each record with phone number, dial the number, when answered play a pre-recorded message. It could be used to notify parents at a school that a after school game is canceled. I appreciate any direction you can point me in. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Voicemail IMAP
For some reason Dovecot also doesn't like the // in the string either.
After removing the imapflags option in voicemail.conf, It tries
{localhost:143/imap//user=username}INBOX
I also tried it with the novalidate-cert option and dovecot chokes on
it as well...
It's almost like if there are no flags to be passed it need to remove
the trailing slash.
I will just use my hacked app_voicemail.c for now. This is a test system
Thanks.
On 1/11/07, Ray Jackson [EMAIL PROTECTED] wrote:
AFAIK: You shouldn't need to modify the code to get this work. I have:
imapflags=novalidate-cert
setup as my IMAP flags which works fine for a Courier IMAP backend.
Courier uses Maildir (not mbx) which works just fine for me?
Cheers,
Ray
Forrest Beck wrote:
OK. I needed to remove the flags from the string. So I modified
app_voicemail.c and recompiled.
It is working now by, not using the flags to login.
It logs in as: ares.school.da.org:143/imap/user=fbeck}INBOX
4587 if (strlen(authuser) 0) {
4588 snprintf(tmp, sizeof(tmp),
{%s:%s/imap/authuser=%s/%s/u
ser=%s},imapserver,imapport,authuser,imapflags,vms-imapuser);
4589 } else {
4590 snprintf(tmp, sizeof(tmp),
{%s:%s/imap/user=%s},imapserver,imapport,vms-imapuser);
In apps/ I modified app_voicemail.c line 4590 and removed /%s after
/imap.
Then I removed ,imapflags after imapport.
My next hurdle is the mailbox format. It's not mbx, and crashes
asterisk after creating the mail message.
Thanks.
On 1/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
Thanks Ray. I also noticed in some a post reply
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg169259.html
that there was a problem with the trailing options /. Is there a work
around for removing the trailing /? I too am having a problem with
it.
Doesn't Work: {ares.school.da.org:143/imap//user=fbeck}INBOX
Works: {ares.school.da.org:143/imap/user=fbeck}INBOX
Thanks again!
On 1/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
I know some of this doesn't belong on this list, but I am just
including it for problem history.
I am trying to setup IMAP Voicemail with our email server.
We are using a non-standards based groupware server called FirstClass.
The server has some built in support for IMAP.
My problem seems to be that the authuser flag is not supported.
When I use mtest in the imap toolkit to connect to the FirstClass
server, and input this as the Mailbox to connect to:
{ares.school.da.org:143/imap/authuser=admin//user=fbeck}Inbox
mtest replies:
[Trying IP address [10.11.5.253]]
[FirstClass IMAP4rev1 server v8.262 at mail.ares.school.da.org ready]
?Can't do /authuser with this server
When I try to connect to the mailbox:
{ares.school.da.org:143/imap/user=fbeck}Inbox
I am prompted with a password which is accepted.
mtest response is:
IMAP4rev1 (RFC 3501) server ares.school.da.org
Mutually-supported SASL mechanisms: CRAM-MD5
Supported standard extensions:
Access Control lists (RFC 2086)
Multiple namespaces (RFC 2342)
Extended UID behavior (RFC 2359)
Implementation identity negotiation (RFC 2971)
LIST children announcement (RFC 3348)
Supported draft extensions:
Notice I also had to eliminate the flags and authuser sections of
the URL.
So my question is.
Has anyone found a way to include the IMAP username and password in
the users mailbox string? something like 1201 =
1201,Joe,email,,imapuser=joe|imappassword=somethingsecret ?
I know all my users email passwords, as they have to change it
through us...
Thanks all.
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___
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[asterisk-users] Voicemail IMAP
I know some of this doesn't belong on this list, but I am just
including it for problem history.
I am trying to setup IMAP Voicemail with our email server.
We are using a non-standards based groupware server called FirstClass.
The server has some built in support for IMAP.
My problem seems to be that the authuser flag is not supported.
When I use mtest in the imap toolkit to connect to the FirstClass
server, and input this as the Mailbox to connect to:
{ares.school.da.org:143/imap/authuser=admin//user=fbeck}Inbox
mtest replies:
[Trying IP address [10.11.5.253]]
[FirstClass IMAP4rev1 server v8.262 at mail.ares.school.da.org ready]
?Can't do /authuser with this server
When I try to connect to the mailbox:
{ares.school.da.org:143/imap/user=fbeck}Inbox
I am prompted with a password which is accepted.
mtest response is:
IMAP4rev1 (RFC 3501) server ares.school.da.org
Mutually-supported SASL mechanisms: CRAM-MD5
Supported standard extensions:
Access Control lists (RFC 2086)
Multiple namespaces (RFC 2342)
Extended UID behavior (RFC 2359)
Implementation identity negotiation (RFC 2971)
LIST children announcement (RFC 3348)
Supported draft extensions:
Notice I also had to eliminate the flags and authuser sections of the URL.
So my question is.
Has anyone found a way to include the IMAP username and password in
the users mailbox string? something like 1201 =
1201,Joe,email,,imapuser=joe|imappassword=somethingsecret ?
I know all my users email passwords, as they have to change it through us...
Thanks all.
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[asterisk-users] Re: Voicemail IMAP
Thanks Ray. I also noticed in some a post reply
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg169259.html
that there was a problem with the trailing options /. Is there a work
around for removing the trailing /? I too am having a problem with
it.
Doesn't Work: {ares.school.da.org:143/imap//user=fbeck}INBOX
Works: {ares.school.da.org:143/imap/user=fbeck}INBOX
Thanks again!
On 1/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
I know some of this doesn't belong on this list, but I am just
including it for problem history.
I am trying to setup IMAP Voicemail with our email server.
We are using a non-standards based groupware server called FirstClass.
The server has some built in support for IMAP.
My problem seems to be that the authuser flag is not supported.
When I use mtest in the imap toolkit to connect to the FirstClass
server, and input this as the Mailbox to connect to:
{ares.school.da.org:143/imap/authuser=admin//user=fbeck}Inbox
mtest replies:
[Trying IP address [10.11.5.253]]
[FirstClass IMAP4rev1 server v8.262 at mail.ares.school.da.org ready]
?Can't do /authuser with this server
When I try to connect to the mailbox:
{ares.school.da.org:143/imap/user=fbeck}Inbox
I am prompted with a password which is accepted.
mtest response is:
IMAP4rev1 (RFC 3501) server ares.school.da.org
Mutually-supported SASL mechanisms: CRAM-MD5
Supported standard extensions:
Access Control lists (RFC 2086)
Multiple namespaces (RFC 2342)
Extended UID behavior (RFC 2359)
Implementation identity negotiation (RFC 2971)
LIST children announcement (RFC 3348)
Supported draft extensions:
Notice I also had to eliminate the flags and authuser sections of the URL.
So my question is.
Has anyone found a way to include the IMAP username and password in
the users mailbox string? something like 1201 =
1201,Joe,email,,imapuser=joe|imappassword=somethingsecret ?
I know all my users email passwords, as they have to change it through us...
Thanks all.
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[asterisk-users] Re: Voicemail IMAP
OK. I needed to remove the flags from the string. So I modified
app_voicemail.c and recompiled.
It is working now by, not using the flags to login.
It logs in as: ares.school.da.org:143/imap/user=fbeck}INBOX
4587 if (strlen(authuser) 0) {
4588 snprintf(tmp, sizeof(tmp),
{%s:%s/imap/authuser=%s/%s/u
ser=%s},imapserver,imapport,authuser,imapflags,vms-imapuser);
4589 } else {
4590 snprintf(tmp, sizeof(tmp),
{%s:%s/imap/user=%s},imapserver,imapport,vms-imapuser);
In apps/ I modified app_voicemail.c line 4590 and removed /%s after /imap.
Then I removed ,imapflags after imapport.
My next hurdle is the mailbox format. It's not mbx, and crashes
asterisk after creating the mail message.
Thanks.
On 1/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
Thanks Ray. I also noticed in some a post reply
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg169259.html
that there was a problem with the trailing options /. Is there a work
around for removing the trailing /? I too am having a problem with
it.
Doesn't Work: {ares.school.da.org:143/imap//user=fbeck}INBOX
Works: {ares.school.da.org:143/imap/user=fbeck}INBOX
Thanks again!
On 1/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
I know some of this doesn't belong on this list, but I am just
including it for problem history.
I am trying to setup IMAP Voicemail with our email server.
We are using a non-standards based groupware server called FirstClass.
The server has some built in support for IMAP.
My problem seems to be that the authuser flag is not supported.
When I use mtest in the imap toolkit to connect to the FirstClass
server, and input this as the Mailbox to connect to:
{ares.school.da.org:143/imap/authuser=admin//user=fbeck}Inbox
mtest replies:
[Trying IP address [10.11.5.253]]
[FirstClass IMAP4rev1 server v8.262 at mail.ares.school.da.org ready]
?Can't do /authuser with this server
When I try to connect to the mailbox:
{ares.school.da.org:143/imap/user=fbeck}Inbox
I am prompted with a password which is accepted.
mtest response is:
IMAP4rev1 (RFC 3501) server ares.school.da.org
Mutually-supported SASL mechanisms: CRAM-MD5
Supported standard extensions:
Access Control lists (RFC 2086)
Multiple namespaces (RFC 2342)
Extended UID behavior (RFC 2359)
Implementation identity negotiation (RFC 2971)
LIST children announcement (RFC 3348)
Supported draft extensions:
Notice I also had to eliminate the flags and authuser sections of the URL.
So my question is.
Has anyone found a way to include the IMAP username and password in
the users mailbox string? something like 1201 =
1201,Joe,email,,imapuser=joe|imappassword=somethingsecret ?
I know all my users email passwords, as they have to change it through us...
Thanks all.
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[asterisk-users] Console\DSP
I am using a extension to dial the console which has autoanswer enabled. I am getting a strange warning, has anyone seen this before? Nothing on Google, or Voip-Info [Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request: oss_request ty console data 0x0xb7851e00 dsp Call to device 'dsp' dnid '(null)' rdnis '(null)' on console from 'XX' XX Auto-answered -- Called dsp -- OSS/dsp answered SIP/mspri-usasterisk-0a119be0 Hangup on console == Spawn extension (system1, 6, 1) exited non-zero on 'SIP/mspri-usasterisk-0a119be0' It works fine, I am just concerned what the warning is for. the extension is simple exten = 6,1,Dial(console\dsp) BTW.. I am using chan_oss not alsa. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MusicOnHold Files
I was just wondering what you all are doing for music on hold files for best quality. I am not much of an expert on sound rates, bits, stereo, mono, tracks, and all that jazz. Currently I am taking music from a CD (our campus jazz band has recorded a CD), converting to WAV, using Audacity to convert the stereo tracks into mono, drop the gain to -15db, then I use sox to convert to GSM and 8 bit (by typing # /usr/bin/sox file1.wav -r 8000 -c1 file2.gsm resample -ql ) The audio while on hold is OK. I wonder if there is a way to get better audio. I noticed that asteriskguru.com has a audio conversion on their website. Thanks for any help. Forrest ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HowTO configure voice T1
PRI is just a standard used on the T1 medium. If you have a solid T1 between the locations. Why not use PRI on the T1. If the T1 is dedicated point to point between locations, then you can use PRI on the line dedicating one channel to signaling (d channel). If you can't give up the 24th channel then have a look at HDLC. I know the sangoma cards support HDLC, but I am not sure about Digium. If you choose PRI, then you just need to set one side as pri_net and the other as pri_cpe. On 1/4/07, David Gomillion [EMAIL PROTECTED] wrote: T1s can use many different signalling types. You need to find out which one is running, what the line encoding is, etc. PRI vs T1 are not the only distinctions... On 1/4/07, Mark Greene [EMAIL PROTECTED] wrote: Alright guys here is my question. What is do I need to set switchtype, and signalling to in zapata for a voice T1. This is not a PRI. I cannot say that enough. It is NOT, A, PRI. It is just a Voice T1 with 24 voice channels. There is not a D Channel. It runs from one office to another and USED to plug into two opt. 11c but now one end is going to plug into an asterisk box. - Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HowTO configure voice T1
You can get switchtype from your carrier. On 1/4/07, Forrest Beck [EMAIL PROTECTED] wrote: PRI is just a standard used on the T1 medium. If you have a solid T1 between the locations. Why not use PRI on the T1. If the T1 is dedicated point to point between locations, then you can use PRI on the line dedicating one channel to signaling (d channel). If you can't give up the 24th channel then have a look at HDLC. I know the sangoma cards support HDLC, but I am not sure about Digium. If you choose PRI, then you just need to set one side as pri_net and the other as pri_cpe. On 1/4/07, David Gomillion [EMAIL PROTECTED] wrote: T1s can use many different signalling types. You need to find out which one is running, what the line encoding is, etc. PRI vs T1 are not the only distinctions... On 1/4/07, Mark Greene [EMAIL PROTECTED] wrote: Alright guys here is my question. What is do I need to set switchtype, and signalling to in zapata for a voice T1. This is not a PRI. I cannot say that enough. It is NOT, A, PRI. It is just a Voice T1 with 24 voice channels. There is not a D Channel. It runs from one office to another and USED to plug into two opt. 11c but now one end is going to plug into an asterisk box. - Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail to email
You just specify the users email address in the voicemail.conf file, along with their mailbox number: see the sample file: [default] ; Define maximum number of messages per folder for a particular context. ;maxmsg=50 2506 = 2506,Grandstream,[EMAIL PROTECTED],,attach=yes|imapuser=fbeck 1234 = 4242,Example Mailbox,[EMAIL PROTECTED] ;4200 = 9855,Mark Spencer,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=no|[EMAIL PROTECTED]|tz=central|maxmsg=10 ;4300 = 3456,Ben Rigas,[EMAIL PROTECTED] ;4310 = -5432,Sales,[EMAIL PROTECTED] ;4069 = 6522,Matt Brooks,[EMAIL PROTECTED],,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|sayduration=yes|saydurationm=1 ;4073 = 1099,Bianca Paige,[EMAIL PROTECTED],,delete=1 ;4110 = 3443,Rob Flynn,[EMAIL PROTECTED] ;4235 = 1234,Jim Holmes,[EMAIL PROTECTED],,Tz=european 2503 = 2503,Forrest Beck,[EMAIL PROTECTED] On 1/3/07, Doug Crompton [EMAIL PROTECTED] wrote: There should be an example in your voicemail.conf Here is mine... mail is tagged from [EMAIL PROTECTED] and sent to [EMAIL PROTECTED] In voicemail.conf mailcmd=/usr/sbin/sendmail -f [EMAIL PROTECTED] [EMAIL PROTECTED] You of course would use the mailer that your system uses. I have sendmail on the same system as Asterisk. There are many other things you can define for mail but all should be in your example voicemail.conf Doug On Wed, 3 Jan 2007, Mark Greene wrote: Hey guys, I need to set up asterisk so that it sends the voicemail to the users email. I understand that I need to say attatch=yes, but what else needs to be done. I would think that somewhere I need to specify the server that it uses to send the email, etc. - Mark Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Lines Confusion
I am not sure if this is what you are looking for, but I will give it a shot. There may be a better way to do this but... I would use agent Queues for your users. Your users can log into the Queue, so that if the dialed user is not available, then it will drop the caller into a Queue for a certain amount of time. After that time period expires they will get a voicemail box for the original person called. If Mary were to have a internal extension of 2101 (external phone number 020-555-1212), and Bob has a internal extension of 2102 (external phone number 020-555-1213) then I would do something like the following: ;Context for incoming calls. VoIP/Zaptel calls will be dropped here: ;Extension is the number the caller dialed: [incoming] ;Mary's Phone Number exten = 020-555-1212,1,Goto(internalextensions,2101,1) ;Bob's Phone number exten = 020-555-1213,1,Goto(internalextensions,2102,1) ;Context for internal extensions. [internalextensions] ;m(class) replace class to specify a context in musiconhold.conf for the music to be played ;during ringing. 20 is how long to dial. ;Mary-- exten = 2101,1,Dial(SIP/2101,20,m(class)) exten = 2101,1,Queue(findsomeone|tTn|||20) exten = 2101,1,VoiceMail([EMAIL PROTECTED]) ;Bob-- exten = 2102,1,Dial(SIP/2102,20,m(class exten = 2102,2,VoiceMail([EMAIL PROTECTED]) www.asteriskguru.com provides alot of good information. Here is a tutorial on setting up agent queues. You should be able to get a agent queue setup fairly quickly. On 12/20/06, Mr Gabriel [EMAIL PROTECTED] wrote: First off, please, for the love of God, don't cremate me, if I should already know the answer to this! I've installed a small setup for an office who wanted to be able to talk to each other instead of having to rely on MSN to communicate. Weird request, I know, but hey, we do what we need to do to get paid. I installed soft phones, gave everyone an extension, and bingo, they can call and talk over their PCs happy as hell. Which brings me to my problem - they loved the system so much, that now, they want it for ALL their calls, that is their calls that involve the real world. ATM, they all have separate independent land lines, which is why they had a problem in the first place, large bills for calling each other, now they want a VoIP solution, that would have calls coming in over their broadband connection, and automatically route to each of their phones, depending on which line has called them. I just got out of a meeting with them, and what they want, goes as such... Bob has a VoIP number 020-xxx-xxx - when this number rings, the box answers the call, plays some music, while it waits for Bob to answer. This call should only go to Bobs extension. If he's not there, it routes the call to his voicemail Mary has a different number. When this number rings, it gets routed straight to her extension, in the same manner as Bobs, but if she's not available, it looks for who is, and rings their phones, and if no one answers, then goes to voice mail. Basically, there are 2 types of behaviours that they would like on their lines. My problem, is how to implement it! I'm an asterisk virgin, and getting them to be able to talk to each other across their office network and 12 extensions, took the best part of 2 hours - I don't want to have to spend a whole day working on this one. The VoIP numbers have already been purchased, and are ready to go - i just need to configure it all - Can it be done!? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No music on hold?
You should look at the asterisk-addons package. There is a addon module in the package called format_mp3 that will play your mp3 files instead of using mpg123 (which is a dead project). I just use sox to convert my mp3's to GSM with something like this: /usr/bin/sox musicfile.mp3 -r 8000 -c1 musicfile.gsm resample -ql This also puts it into 8bit mono, which sounds a little better on our phones. I have a compiled version of sox with added support for the lame encoder (mp3). They are RPM's built for CentOS/RHL. If you want them. On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote: I'm totally at a loss here. I can't get music on hold when placing someone on hold or when dialing an internal extension. When I dial an internal extension I hear ringing yet on my phone it shows little musical notes like it thinks it's hearing music. What to do! J Phil Lee Jenkins wrote: I was wondering the same thing as my MOH isn't working either in a 1.2.14 installation so I'm recompiling mpg123 as per: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat We know you obviously need to use the m flag for the caller to hear MOH when dialing an extension, but I wonder if it's required AFTER the call has been answered and then put on hold. OK, asterisk just finished compiling and my MOH is working correctly. I have also verified that you do *not* have to have m in the Dial command in order for MOH to play when placed on hold. Note that I have a command in the initial context of my dialplan that set music on hold: exten=s,1,SetMusicOnHold(default) -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial 9 to get out?
Is this what you are looking for
exten = _9.,1,Set(CALLERID(num)=3045551212)
exten = _9.,n,Dial(ZAP/g2/${EXTEN:1})
On 12/20/06, Bruce Reeves [EMAIL PROTECTED] wrote:
Look at the digit map in your Polycom configuration files. I had the same
problem and had to chage the digit map to support an extra digit when
dialing 9.
On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote:
Hi all,
Can someone point me in the right direction here. What I'd like to do
with Asterisk is a) dial a 3 digit extention (i.e. 100) on my polycom phones
and after the 3rd digit is entered, it dials that extension and b) dial 9 to
get out like older PBX systems. Since my internal extensions start with a 1
I think what happens is I enter extension 100 for example, and the phone
sits there. If I enter 1, areacode, number the moment I enter the last
digit of the number it dials the number. ALSO I'd like to be able to dial
local numbers without using 1+areacode. Note that I'm using voicepulse so
it makes sense that it isn't intelligent enough to know when a number is a
local one or not.
Thanks!
Phil
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--
Bruce
Nortex Networks
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Re: [asterisk-users] sip help for newbie
www.asteriskguru.com On 12/12/06, blackwater dev [EMAIL PROTECTED] wrote: Does anyone know of any good step by step tutorials on getting sip set up? I have asterisk installed but can't seem to figure out how to get an account set up and connect from my xTen phone so I can try the demo. The tutorials I read online seem to go into voicepulse stuff and all and I don't have an account there so am a bit lost. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk accepting calls to fast
Have a look at TIMEOUT(digit) http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout On 12/7/06, Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED] wrote: Hi, the german telco Colt Telekom has assigned the phone number block 56830-xxx to one of our customers. In the diaplan we have setup extensions like the following ones: exten = 56830910,1,Answer() exten = 56830910,2,Dial(SIP/bduerring,10,tr) exten = 56830910,3,VoiceMail,u20 exten = 56830910,4,hangup exten = 56830910,103,VoiceMail,b20 exten = 56830910,104,hangup exten = 5683091,1,Answer() exten = 5683091,2,DIAL(ZAP/g5/56830990,10,r) exten = 5683091,3,Hangup The problem now is, that sometimes (maybe when the caller doesn't hit the buttons fast enough) asterisk takes the extension for 5683091, although the 0 is still coming a little bit later. I'm not quite sure whether the delay in transferring the numbers is caused by the caller or by the telco. But is their a chance to tell asterisk to wait a little bit longer, before it starts searching the extensions.conf? Or do I have to tell the ISDN card to wait for the complete number, before it is forwarded to asterisk? Software hardware: SuSE 10.0 Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p chan_capi-0.7.0 divas4linux_EICON-106.20-1 Eicon Networks Corporation Diva Server 4BRI Rev 2 Thanks for your help hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk freezes when DNS not working: a BUG??
You can run dnsmasq on the machine for local caching of the dns names. (http://thekelleys.org.uk/dnsmasq/doc.html) and then apply this patch that will allow dnsmasq to set a minimum time to live (http://lists.thekelleys.org.uk/pipermail/dnsmasq-discuss/2005q2/000253.html). dnsmasq can be then configured to only allow localhost inquires On 12/6/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Bob, thanks for reply. The problem is all PBX are not in the same LAN and every customer wants his/her own DNS. I think I'll use /etc/hosts but the problem still remain: Asterisk shouldn't freeze during reloadthe registration should be located in another process but I think that such a change would modify too much Asterisk sip/iax applications and part of Asterisk architecture. So, I know it works that way, I accept it and I try to workaround it. Thanks Giorgio Incantalupo Bob Chiodini wrote: Giorgio, You could set up a caching name server in your local network, use it as your primary DNS server and your ISP's as a secondary. This would cache your ITSP's address(es) locally limiting your reliance on your ISP. Bob... On Wed, 2006-12-06 at 10:43 +0100, Giorgio Incantalupo wrote: Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP so I cannot use it. Is there anybody who had a problem like this? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only
I use the GoToIf:
If the SIP phone is Extension 2501 and dials out (I am using the
norstar 9 to dial out convention). BTW. ${PSTNOUT} is a global
variable for ZAP/G2.
exten = _9X.,1,GoToIf($[${CALLERIDNUM} = 2501]?2:3)
exten = _9X.,2,Set(CALLERID(num)=9195551212)
exten = _9X.,3,NoOp(${CALLERIDNUM})
exten = _9X.,4,Dial(${PSTNOUT}/${EXTEN},300,)
exten = _9X.,5,Hangup()
On 12/6/06, Rob Schall [EMAIL PROTECTED] wrote:
The way I would try to solved this would be to have a different context
for just that use. His outbound calls would set a personal caller id,
and then make the outbound call. Everyone else would use the group
context. Other than that, possibly a good Macro might take care of that.
Rob
Ron McCarthy wrote:
Yeah,
Bascailly lets say extension 2 places a outbound call, it needs to
show that persons private DID name and number, and anyone else gets
the global callerid name/number. I guess you do this via a if
statement, im trying but having a hell of a time getting it to work!
On 12/6/06, *Rob Schall* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Ron,
By source #, i assume you mean you have something like a SIP phone on
the network with the extension like 4455, and you want that to have a
different caller id when you make outgoing calls, then the rest of
the
phones on your network (the rest would show a global company number).
Based on where you put the exten, it works as either an incoming or
outgoing handle. For example, I have it setup on our network, that if
you dial out and connect to our local area code, that the callerid is
one number. If you call long distance, you would see our 1800 number.
Is this the setup you are looking for?
Rob
Ron McCarthy wrote:
Hi Rob,
Well see that would work great if I knew the numbers they would be
calling, but all I know is the source number/phone, i have no
clue who
they will be calling. Any ideals now? I wish it was that easy!
Thanks!
On 12/6/06, *Rob Schall* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Ron,
I believe you would just want to edit your extensions.conf
file so
that
the extension you want separate has its own rule set.
exten = 4567,1,Set(CALLERID(all)=000-000-)
exten = 4567,n,Dial(SIP/4567)
all other calls would just fit in like:
exten = _4.,1,Set(CALLERID(all)=111-000-)
exten = 4567,n,Dial(SIP/${EXTEN})
Hope that gives a bit of insight or puts you in the right
direction.
Rob
Ron McCarthy wrote:
Hi List,
Ive got one extension/login that when they call out from
that it
needs
to show a different name/number, and then the rest of the
phone will
have a default one. Whats the best way to do this? I know
it can be
done, just cant figure out how! Ive looked around and seem
to see no
docs on it. Any help or examples would be great on this!
Thanks!
Ron
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Re: [asterisk-users] TE110P Out fine / In Fail
30 Channels on Verizon? Is this in the US? T1 (24 channels) or E1(30 channels)? Are you dialing from the top (g1) of the group or bottom (G1)? On 12/5/06, Klaverstyn, David C [EMAIL PROTECTED] wrote: I have just installed Asterisk wit a TE110P card. I have configured 30 channels which seems to be recognised by staff and zap show channels. I can make outbound calls with exceptional call quality but inbound (receiving) calls the caller get a message saying Your call could not be connected, please check the number and try again. Nothing is displayed in the CLI. Is this a configuration problem with Asterisk or a problem with Verizon? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 4.4 + asterisk
That kernel-devel fix is just for ZAPTEL. The bug has been solved in 4.4 On 12/5/06, Vicky [EMAIL PROTECTED] wrote: I am not sure but i think that fix is for compiling zaptel not asterisk . Asterisk runs on centos with 0 problems :) On 05/12/06, varun [EMAIL PROTECTED] wrote: Thanks Karl. On Tue, 2006-12-05 at 08:20 -0500, [EMAIL PROTECTED] wrote: I have CentOS 4.4 on several boxes with Asterisk 1.2 and they run great. Have not tested conferencing yet though. Karl Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Native TDM Bridge
I have a two port TE205P Digium card. I have set everything up to
create a native zap bridge between the two spans. Everything works
perfectly except one thing. Our telco has a password that has to be
entered as soon as a long distance call is made. So if I dial a long
distance call from my meridian system, asterisk bridges the call
between two channels, my telco picks up and gives me a tone, I enter
my 3 digit password, and the call is supposed to be completed.
Instead I get a busy signal. The call is already bridged by the time
I have to punch in the telco password. This works fine If I plug my
norstar system directly into the PRI telco. Another strange issue is,
If I make a slip of the finger and I dial 1-349-555-1 with a
trailing digit all works fine.
Anyone have an idea I tried relaxing the DTMF on both PRI's as
well. Debug does't show anything either.
SPAN1 is connected to my Telco's PRI and SPAN2 is connected to a
Norstar Meridian.
Here is my zapata.conf:
[channels]
#PRI to TimeWarner defined as group 2
switchtype = national
signalling = pri_cpe
context=pstn
callerid=asreceived
resetinterval = never
group = 2
channel = 1-23
#PRI to Norstar Meridian defined as group 3
switchtype = national
signalling = pri_net
context=meridian
callerid=asreceived
resetinterval = never
group = 3
channel = 25-47
Here is my zaptel.conf:
#PRI to TimeWarner
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
#PRI to Norstar Meridian
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
loadzone=us
defaultzone=us
Here is my extensions.conf:
;
; Context for meridian incoming calls
; Check incoming call to see if callerid is already set. If not
; then set it to the Main number, and forward it out to the
; highest available channel on the TW PRI.
[meridian]
exten = _X.,1,GoToIf($[${CALLERIDNUM} = ]?2:3)
exten = _X.,2,Set(CALLERID(num)=EXCLUDEDFORLIST)
exten = _X.,3,NoOp(${CALLERIDNUM})
exten = _X.,4,Dial(${PSTNOUT}/${EXTEN})
exten = _X.,5,Hangup()
exten = i,1,Answer()
exten = i,n,Wait(1)
exten = i,n,Playback(cannot-complete-as-dialed)
exten = i,n,Playback(please-contact-tech-supt)
exten = i,n,Hangup()
;
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Re: [asterisk-users] operator console
Talk to the folks at Asteria. The have a product called Reign. It looks just like your old interface, runs off .NET as a client on the machine. http://www.asteriasgi.com/pbx/reign On 11/7/06, Stephen Wingfield [EMAIL PROTECTED] wrote: Andres, The Bicom Systems Operator Panel is probably what you are looking for. OPCOM http://www.bicomsystems.com/docs/opcom/1.0/html/ This is included with every copy of PBXware and is fully supported. If you care to register you may order a trial of PBXware with our SOHO. Regards Steve steve 'at' bicomsystems 'dot' com - Original Message - From: Andres Paglayan To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 5:27 PM Subject: [asterisk-users] operator console Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and extension, and voila. I'll be switching the company's phone system to Asterisk. I know * is way much more flexible and rich featured than the box we currently have, ...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? I don't mind using a commercial product, if the only part we have to pay for is the gui, besides, we will buying the enterprise * version Thanks a bunch, Andres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Register vs. Host=IPADDR
I am not sure if I am going to use SIP registration's or just specify the host ip address in sip.conf. Are there any pros or cons to the two? My phones will have a static IP address and won't be changed unless a admin does it. So the logical path would be to just turn off registration on the sip account (in the phone setup). Can anyone forsee a problem to this? Something I will miss out on if I had used sip registration? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel.c: Unable to request channel ZAP
What does zap show channels show? Are all the channels shown as in use? What is set in zapata.conf for resetinterval= ? If anything. I believe that resetinterval is used to reset unused channels for any channels that are left open. On 10/31/06, Asterisk [EMAIL PROTECTED] wrote: Hi All, I have one rather annoying problem...my PBX can work great for weeks, when suddenly I start receiving these messages when I try to make a zaptel call: Oct 31 13:52:47 NOTICE[15636] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) Oct 31 13:52:49 NOTICE[15648] channel.c: Unable to request channel ZAP/g1/247 I'm using Sangoma A104 card (with four E1 spans), and these problems are only occurring on the first two spans (which are connected to a legacy PBX) – the second two spans, which are connected to the Telco, work perfectly. Even more: when these messages start to occur, I can hardly initiate any call via problematic two spans (1st and 2nd), where I can with no problem initiate a new call thru the unproblematic two spans (3rd and 4th). Restart of the Asterisk is the only cure so far… Does anyone know what could possibly be the cause, or how could I troubleshot this problem? Regards. Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Characters in CLI on TTY9
When I look at TTY9 (using init.d and safe_asterisk to start the asterisk process), I am getting some strange characters. When a application is run the and the CLI shows the application executing the languange almost looks russian...?? Anyone seen this before? http://picasaweb.google.com/jonforrest.beck/AsteriskCLI ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Help
I need some help with AGI. I am unsure how it is written and works. But I have a bash command that will spit out a two digit numerical value (The temperature in the room). The bash command is: #!/bin/bash /usr/local/digitemp/digitemp-1.3/digitemp -a | tail -n1 | cut -d -f9 | cut -d . -f1 This command will spit out someting like 73. I would like to get asterisk to say these digits to the caller. How should I do this? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switchtype,Signalling,rxwink warnings
What's in zapata.conf? On 10/13/06, Remi Quezada [EMAIL PROTECTED] wrote: When I reload the asterisk I get the following warnings: Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring switchtype Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring signalling Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring rxwink Everything works fine as far as I know, I can dial and complete calls. So why am I getting this warning. Is there anyway to fix this? Thanks, Remi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom HDVoice
Has anyone used the Polycom HDvoice phone yet? I am curious if it uses a different codec. Does it actually sound any better? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as SIP Client
You could use heartbeat http://www.linux-ha.org (or ultramonkey http://ultramonkey.org). With this you set up a director that shares the load to multiple servers. You can even set it to have consistent connections so a originating IP will return the the same server. I have hearbeat running on two asterisk servers for high availability instead of load balancing. The configs are shared through drbd (www.drbd.org). Give these a look at. If you are running CentOS just type: yum groupinstall heartbeat-drbd FB On 10/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I am little confused on load balancing, when asterisk server is also a sip client. Based on these, XO Communications one of the largest US DID Provider, now offer SIP Orignation Services for wholesale. Verizon Communications One of the largest US Teleco, now offer SIP Orignation Services. That means no need for PRI card. So if I take service from them, then my asterisk server will be SIP client. Right? How can I set up my asterisk servers so that the calls originated by XO/Verizon goes to different asterisk servers based on load. Has any one does this and can share that with me. Any idea or hint will be appreciated. Thank you, -Kunal, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Cancel Cards
I am using the TE110P with the Intel 945P chipset, and I don't have any issues with compatibility. The 945P chipset is a very common chipset for the D and 4 processor. Works quite well. On 10/10/06, R.R. Libera [EMAIL PROTECTED] wrote: I´m about to acquire an E1 interface. I was reading about TE110P and hardware incompatibilities issues with some boards, servers and chipsets. I also read a lot of compliments about Sangoma Hardware (specially for E1/T1 interfaces) and I was wondering if A101 from Sagoma is a better choice (technically speaking) than Digium TE110P. I read now, on this post, an opinion about Sangoma interfaces and echo cancellation issues.. I have a PC with Asus P5LD2 board (Intel 945P chipset). I asked Digium support for how compatible is the TE110P with my box.. and they said that no incompatibility issues had been reported with the chipset I use.. BUT, they had no test TE110P with this chipset... I´m not a Sangoma or Digium fan... I´m just a newbie who don´t want to get the wrong piece of hardware. I really appreciate any advice from people with a lot of experience and skills on this topic. Thanks in advance R.R. Libera Dovid B escribió: I have never used T1 cards but as far as POTS line cards I would say that I like sangoma better. It is a little bit harder to set up but works wonders. - Original Message - From: Thomas Kenyon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 12:31 PM Subject: Re: [asterisk-users] Echo Cancel Cards Joseph wrote: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. I've got an SPA-3000 at home that is constantly crashing, echoey and is almost unusable. (The CS4660-based ATA and PA1688-based handsets have otherwise been fine, as were the the Cisco 468 and Linysys PAP2 when they were in use). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo Cancel Cards
Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 Passthrough
I want to setup a asterisk server with two T1 spans (two TE110P
cards). The server will have one card connected to the PRI and the
other will connect to our Norstar Meridian ICS system. I want to have
a very simple dial plan for the context that the PRI card will be
assigned to something like this. Note that our telecom provider sends
final three digits of the phone number:
SPAN 1
Channels 1-23
g1
context: pri_incoming
SPAN 2
Channels 25-48
g2
context: norstar_ics
[pri_incoming]
exten = _XXX,1,Dial,ZAP/g2/${EXTEN}
My questions are:
Will I need to set the callerid before routing to the next span, or
will the three digits remain intact.?
and
Has anyone tried this? and if so do you forsee any problems i will run into?
This is all theroey in my head right now, since I am awaiting the
second cards arrival.
Thanks.
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Re: [asterisk-users] IP Phones
http://www.voipsupply.com/home.php On 10/4/06, Devraj Mukherjee [EMAIL PROTECTED] wrote: Nokia E series with proper firmware upgrade :) On 10/5/06, Steve Glaus [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi List; I would like to know where I can find the IP Phones that can be used with Asterisk? Is there any link? Regards Bilal Ghayad Mobile: 00965 9849460 Office: 00965 2623174 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Any phone supporting SIP or IAX are good choices for asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail and Forwarding
I am a little stumped on this one and it may be because my brain is
ready for the weekend. I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
I tried.
exten = 2000,1,Voicemail([EMAIL PROTECTED])
this of course gives me a error that mailbox 2000 doesn't exist.
I also tried..
exten = 2000,1,Voicemail(${CALLERID(num)[EMAIL PROTECTED])
This gives the original caller his own mailbox.
Stumpped.
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[asterisk-users] Re: Voicemail and Forwarding
Nevermind. Just decided to use:
exten = _22XXX,1,Voicemail(u${EXTEN:[EMAIL PROTECTED])
On 10/6/06, Forrest Beck [EMAIL PROTECTED] wrote:
I am a little stumped on this one and it may be because my brain is
ready for the weekend. I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
I tried.
exten = 2000,1,Voicemail([EMAIL PROTECTED])
this of course gives me a error that mailbox 2000 doesn't exist.
I also tried..
exten = 2000,1,Voicemail(${CALLERID(num)[EMAIL PROTECTED])
This gives the original caller his own mailbox.
Stumpped.
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Re: [asterisk-users] Unknown signalling method 'pri_cpe'
build libpri. On 10/3/06, Eugeniy Khvastunov [EMAIL PROTECTED] wrote: yusuf пишет: Eugeniy Khvastunov wrote: Hello! Why Asterisk tell: Unknown signalling method 'pri_cpe' Why the asterisk does not know such signaling method? [chan_zap.so] = (Zapata Telephony) Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601 setup_zap: Unknown signalling method 'pri_cpe' Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601 setup_zap: Unknown signalling method 'pri_cpe' Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10226 setup_zap: Signalling must be specified before any channels are. Oct 3 13:04:02 WARNING[5823]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Oct 3 13:04:02 WARNING[5823]: loader.c:554 load_modules: Loading module chan_zap.so failed! Ouch ... error while writing audio data: : Broken pipe I think its because you dont have libpri installed. Install libpri, then try! After installation libpri I need to reinstall asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Gateway
Not that server model, right now we have a dell server for testing (which puts the server cost ata round $2000). I am hoping to get one in to see if it will play nice with the TDM cards. This appealed to me because of it's small rackable form factor and cheap price. For that price I can have a cold/hot spare. I will post again if I have luck. On 9/26/06, Kevin Kiely [EMAIL PROTECTED] wrote: Forrest, I noticed your post on the mailing list and was curious if you had used that server before with asterisk with any TDM cards in it? Kevin -Original Message- From: Forrest Beck [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 26, 2006 1:50 PM To: Asterisk Users List Subject: [asterisk-users] SIP Gateway I am thinking of using a mini atx 1u server with a digium zaptel (wcte11xp) installed to act as a SIP gateway. This way any of my asterisk servers can forward calls to any gateway (seperated by about 3miles of fiber). Has anyone else tried this? I would just load a basic asteisk config and zaptel with something like CentOS 4.4 ServerCD. Here is the hardware I am thinking of. http://www.abmx.com/1u-short-depth-rack-mount-server-p-256.html It seems like this would be alot cheaper than getting a pre-built sip gateway from VOX. Any input is greatly appreciated. Forrest ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.9/456 - Release Date: 9/25/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP Gateway
To rich for my blood. Googled it. Looks like it is about $12000, I hope to stay in the $1500 range. We are but mearly a private school. On 9/29/06, James [EMAIL PROTECTED] wrote: I use the Lucent MAX TNT. They are cheap, will do up to 24 T1's, have 12 fans and I've never had one fail. I also can't remember the last time that I had to reboot on of them. G.711 G.729 is built in. James Taylor 1-903-691-0069 - Original Message - From: Forrest Beck [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Tuesday, September 26, 2006 12:50 PM Subject: [asterisk-users] SIP Gateway I am thinking of using a mini atx 1u server with a digium zaptel (wcte11xp) installed to act as a SIP gateway. This way any of my asterisk servers can forward calls to any gateway (seperated by about 3miles of fiber). Has anyone else tried this? I would just load a basic asteisk config and zaptel with something like CentOS 4.4 ServerCD. Here is the hardware I am thinking of. http://www.abmx.com/1u-short-depth-rack-mount-server-p-256.html It seems like this would be alot cheaper than getting a pre-built sip gateway from VOX. Any input is greatly appreciated. Forrest ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Gateway
I am thinking of using a mini atx 1u server with a digium zaptel (wcte11xp) installed to act as a SIP gateway. This way any of my asterisk servers can forward calls to any gateway (seperated by about 3miles of fiber). Has anyone else tried this? I would just load a basic asteisk config and zaptel with something like CentOS 4.4 ServerCD. Here is the hardware I am thinking of. http://www.abmx.com/1u-short-depth-rack-mount-server-p-256.html It seems like this would be alot cheaper than getting a pre-built sip gateway from VOX. Any input is greatly appreciated. Forrest ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with Tieing Outbound calls to Zap Channels
Setting the callerid should be passed from asterisk through the zap channel. Have a look at CALLERID http://www.voip-info.org/wiki-Asterisk+cmd+SetCallerID I can speak for a 23channel PRI, not sure about analog. On 9/22/06, Kevin Steil [EMAIL PROTECTED] wrote: I would like to tie outbound calls from specific extensions to specific zap channels...I have multiple clients in an executive suite and would like to be able to tie lets say extension 1234 to Zap Channels 1 and 2 and extension 5678 to channels 3 and 4 and so on... This so that their caller ID show up properly on outbound calls.. Thanks Kevin J. Steil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA with wireless client
This suggestion may be sort of a hodge podge setup, but you could use something like a airport express, which has wireless bridging built in. Connected directly to a ATA On 9/22/06, Brian Candler [EMAIL PROTECTED] wrote: Sorry, one other equipment query: does anyone know of an ATA with wireless hardware which can act as a *client* to another wireless network? The Linksys units have an integrated wireless access point, but I want something which will work as a client onto an existing wireless network - so you can install ATAs around a building without additional LAN cabling. An ATA with integrated Homeplug (powerline carrier networking) would be another option, but again I can't find such a thing. Any suggestions? Many thanks, Brian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
