[asterisk-users] Answer Machine detection
I use Asterisk mainly in the IVR world and sometimes do a few outbound based campaigns. The horrible subject of Answer Machine Detection as lifted its head again. To my knowledge there are 3 ways to deal with it. 1) When the call is answered, you please a message something like There is an important message for you please press 1 and you detect the Live/Machine state from that. 2) You listen for the Hello? using backgrounddetect then a pause. If there is little or no pause assume it's an answer machine else it's a live 3) You listen for the tone or beep given off by the answer machine at the end of the announcement. At the moment prefer 1, it's easy and quick to implement. Using method 2 is tricky you need to play with various settings and I only usually get a 60% success rate with it. Option 3, I have never seen an asterisk based solution for this one, or know that they exist. Does anyone have any snippets or even a commercial solution to get option 2 success rate to the 90% mark? or anyone know of some that has an asterisk solution for option 3?. Failing that is there an option 4 or 5? Thanks, 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
[asterisk-users] Sending back a ring signal, SIP 180 i think.
Guys,
I am trying to set up a SIP to PSTN gateway, and I have a client sending in
SIP traffic, once on the asterisk machine, I do something like this.
[outbound-sip]
exten = _XXX.,1,NoOP(${CDR(accountcode)})
exten = _XXX.,2,Dial(ZAP/g1/${CDR(accountcode)}${EXTEN})
This works, except that the client wants to bill only when the call is
finally connected on the zap line, and to his SIP equipment he needs to see
a 180 or 183 SIP signal back until the call on the zap line is connected.
I am a bit stuck at this point :-(. Anyone know how I can do this?
Thanks,
Mark
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[Asterisk-Users] Sangoma card A101 Card troubles.
Hiya all, I have had no end of trouble trying to get my A101 E1 card working on a new asterisk installation. The sangoma tech people have ignored my emails about this. All the installation of wanpipe seems to go ok, and zaptel. it all installed compiled and does all the wanpipe hwprobe exactly as documented. Asterisk compiled ok, but when I run it give me Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10614 setup_zap: Unknown signalling method 'pri_cpe' Jun 27 11:12:18 ERROR[10528]: chan_zap.c:10239 setup_zap: Signalling must be specified before any channels are. Am I right in thinking that's it's something to do with libpri? 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
RE: [Asterisk-Users] PHP UnixODBC MS SQl 2000
I used the configure option '--with-mssql' after freeTDS is installed. http://uk.php.net/manual/en/ref.mssql.php Fatal error: Call to undefined function: odbc_connect() in /var/www/html/odbctest.php on line 3 ___ --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] CallerID
Here in the UK on pri, setting the callerid to 0, withholds it. I am trying to set CIDNum to nothing, but my outgoing PRI controlled by another PBX seems to fill in something when asterisk does not.. If I set a number either in the sip channel for the phone, or from extensions.con, it is realized.. If I try to leave them blank, or even Not Defined, the main number of the pri gets sent out.. I am trying to debug a glitvh in or software and I need to be able to make a test call with unknown (blank callerid).. I can successfully set it to private, but that is not the same.. ___ --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] Dialing status detection
Hiya, the disconnected number and fax are fairly easy to detect, as the behavior on these connections is consistent. Answering machines and Human is a bit trickier. You can : 1) do a Background detect for noise to try and pick the hello, then a pause that happens when a human answers. 2) Play a there is a message for you, to hear press 1. This is much more accurate, but not very friendly, as most people just put the phone down. 3) Listen for the answering machine beep. You start by assuming it's a human, and if you hear a answering machine style beep, restart a machine answered style voice file. All the these methods have their pro's and con's. If you can come up with a better and more accurate way, then slap and a patent on it and try sell it. rantrandom disclaimer about how crap software patent are here/rant Mark Anyone has hints to share about dialing result detection. By that I mean the ability to detect what answered: - Human - Answering machine - Fax - Disconnected number. ___ --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: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. Works perfectly on mine. Using Digium Digital boards (ZAP - UK Mobile). Asterisk version 1.2.1 with a few patches. 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
RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky
I noticed one thing: I got courtesytone = beep in my features.conf If I took it off, I got no sound. That's one sorted out :-) Do you have this on it? Do you have a global DYNAMIC_FEATURES = monitor in extensions.conf ? Yes. ___ --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] hyperthreading and zaptel
Finally, I decided to turn hyperthreading back on, and everything is back to normal. Unless there is somewhere in CentOS 4.3 that has the processor count hardcoded from the install, I am baffled by this. Was it on when * and zaptel was compiled?. Maybe the compiler produced HT optimized binaries. Just a thought? 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
RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky
All sorted now. The features timeout needs to be quite high on mobiles. After a few tests, it works perfectly. thanks for your help :-) 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
RE: [Asterisk-Users] DTMF detection when outgoing call to mobile phones
From my recent problem on this sort of thing, I'd suggest you set the timeout to around 1500ms in the feature.conf file. This is of course if your using the DTMF digit's to activate any of the features. also make the devices both sides of the call are using the same DTMF mode. Mark Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. ___ --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 Option wW picking up the *1 is a bit flaky
When I hit *1 in my system, I got a beep to let me know that the recording started. Is this not happenning to you? No ! , it doesn't. Most of the time it doesn't pick up the *1. ___ --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 Option wW picking up the *1 is a bit flaky
I have exactly the same problem - attended transfer (*2) is the same - sometimes asterisk just generates the DTMF tome for * followed by the number instead of interpreting the command. Are there any DTMF configuration settings that can be tweaked? How does Asterisk decide is a key sequence is a command or needs to be transferred as DTMF on the line? I have several applications on the same server that use a lot of DTMF key sequences to move around the system and this works flawlessly. ___ --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: RE: [Asterisk-Users] Dial Option wW picking up the *1 is abitflaky
I have been messing about with this all day. Below is a debug and verbose 99 log of me calling into the system (landline) and being connected out to my mobile. at the start of the log I am pressing #1 on my mobile. I have the record set up to kick in on #1. As you can see this request is ignored. I have wW set in the dial command, at the bottom of the call I am pressing #1 on the landline phone and it kicks in first time. I reckon there is a bug somewhere. -- Attempting native bridge of Zap/32-1 and Zap/1-1 May 3 19:35:33 DEBUG[22030]: chan_zap.c:4602 zt_read: DTMF digit: 1 on Zap/1-1 May 3 19:35:33 DEBUG[22030]: channel.c:3281 ast_generic_bridge: Got DTMF on channel (Zap/1-1) May 3 19:35:33 DEBUG[22030]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels Zap/32-1 and Zap/1-1 May 3 19:35:33 DEBUG[22030]: res_features.c:1333 ast_bridge_call: Timed out for feature! -- Attempting native bridge of Zap/32-1 and Zap/1-1 May 3 19:35:34 DEBUG[22030]: chan_zap.c:4325 __zt_exception: Exception on 51, channel 32 May 3 19:35:34 DEBUG[22030]: chan_zap.c:3517 zt_handle_event: Got event Dial Complete(9) on channel 32 (index 0) May 3 19:35:34 DEBUG[22030]: chan_zap.c:1382 zt_enable_ec: Echo cancellation already on May 3 19:35:35 DEBUG[22030]: chan_zap.c:4602 zt_read: DTMF digit: # on Zap/1-1 May 3 19:35:35 DEBUG[22030]: channel.c:3281 ast_generic_bridge: Got DTMF on channel (Zap/1-1) May 3 19:35:35 DEBUG[22030]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels Zap/32-1 and Zap/1-1 May 3 19:35:35 DEBUG[22030]: res_features.c:990 ast_feature_interpret: Feature interpret: chan=Zap/32-1, peer=Zap/1-1, sense=2, features=16 May 3 19:35:35 DEBUG[22030]: res_features.c:1458 ast_bridge_call: Set time limit to 500 -- Attempting native bridge of Zap/32-1 and Zap/1-1 May 3 19:35:36 DEBUG[22030]: chan_zap.c:4602 zt_read: DTMF digit: 1 on Zap/1-1 May 3 19:35:36 DEBUG[22030]: channel.c:3281 ast_generic_bridge: Got DTMF on channel (Zap/1-1) May 3 19:35:36 DEBUG[22030]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels Zap/32-1 and Zap/1-1 May 3 19:35:36 DEBUG[22030]: res_features.c:1333 ast_bridge_call: Timed out for feature! -- Attempting native bridge of Zap/32-1 and Zap/1-1 May 3 19:35:36 DEBUG[22030]: chan_zap.c:4325 __zt_exception: Exception on 51, channel 32 May 3 19:35:36 DEBUG[22030]: chan_zap.c:3517 zt_handle_event: Got event Dial Complete(9) on channel 32 (index 0) May 3 19:35:36 DEBUG[22030]: chan_zap.c:1382 zt_enable_ec: Echo cancellation already on May 3 19:35:37 DEBUG[22030]: chan_zap.c:4602 zt_read: DTMF digit: # on Zap/1-1 May 3 19:35:37 DEBUG[22030]: channel.c:3281 ast_generic_bridge: Got DTMF on channel (Zap/1-1) May 3 19:35:37 DEBUG[22030]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels Zap/32-1 and Zap/1-1 May 3 19:35:37 DEBUG[22030]: res_features.c:990 ast_feature_interpret: Feature interpret: chan=Zap/32-1, peer=Zap/1-1, sense=2, features=16 May 3 19:35:37 DEBUG[22030]: res_features.c:1458 ast_bridge_call: Set time limit to 500 -- Attempting native bridge of Zap/32-1 and Zap/1-1 May 3 19:35:38 DEBUG[22030]: chan_zap.c:4602 zt_read: DTMF digit: 1 on Zap/1-1 May 3 19:35:38 DEBUG[22030]: channel.c:3281 ast_generic_bridge: Got DTMF on channel (Zap/1-1) May 3 19:35:38 DEBUG[22030]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels Zap/32-1 and Zap/1-1 May 3 19:35:38 DEBUG[22030]: res_features.c:1333 ast_bridge_call: Timed out for feature! -- Attempting native bridge of Zap/32-1 and Zap/1-1 May 3 19:35:38 DEBUG[22030]: chan_zap.c:4325 __zt_exception: Exception on 51, channel 32 May 3 19:35:38 DEBUG[22030]: chan_zap.c:3517 zt_handle_event: Got event Dial Complete(9) on channel 32 (index 0) May 3 19:35:38 DEBUG[22030]: chan_zap.c:1382 zt_enable_ec: Echo cancellation already on May 3 19:35:40 DEBUG[22030]: chan_zap.c:4602 zt_read: DTMF digit: # on Zap/32-1 May 3 19:35:40 DEBUG[22030]: channel.c:3281 ast_generic_bridge: Got DTMF on channel (Zap/32-1) May 3 19:35:40 DEBUG[22030]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels Zap/32-1 and Zap/1-1 May 3 19:35:40 DEBUG[22030]: res_features.c:990 ast_feature_interpret: Feature interpret: chan=Zap/32-1, peer=Zap/1-1, sense=1, features=16 May 3 19:35:40 DEBUG[22030]: res_features.c:1458 ast_bridge_call: Set time limit to 500 -- Attempting native bridge of Zap/32-1 and Zap/1-1 May 3 19:35:40 DEBUG[22030]: chan_zap.c:4602 zt_read: DTMF digit: 1 on Zap/32-1 May 3 19:35:40 DEBUG[22030]: channel.c:3281 ast_generic_bridge: Got DTMF on channel (Zap/32-1) May 3 19:35:40 DEBUG[22030]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels Zap/32-1 and Zap/1-1 May 3 19:35:40 DEBUG[22030]: res_features.c:990 ast_feature_interpret: Feature interpret: chan=Zap/32-1, peer=Zap/1-1, sense=1, features=16 -- User hit '#1' to record call. filename: wav|auto-1146684940-870751-s|m
[Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky
I have dial through application, that uses the wW options on the dial command. However it's seems to be really hit or miss if asterisk picks up the *1 and starts the recording. It can take 3 or 4 attempts before I can see from the console that's it's started recording. A user just on the call not able to see the console has no chance of knowing if it was started or not. does anyone know of any other area's of the system that I can tweak that would make this a bit less flakey. Thanks, 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
RE: [Asterisk-Users] Get sysdate + 5 minutes
Grab the UNIX timestamp and add 5*60 to it.
In my application I want to have the sysdate + 5 minutes.
I know that the sysdate is in the variable ${DATTIME}
But now I want to now how I get the sysdate + 5 minutes into a variable?
Doe's anybody knows the answer?
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RE: [Asterisk-Users] Call recording
Have a look at the Dial command, http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial the w and W option allow you to start recording at any time with the *1 keypress. Is there a way to record a call conversation starting in the middle of the call? I know I can recording whole conversation with mixmonitor, but I prefer only recording certain part of the conversation. Thnx. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grabbing the billsec and duration after a hangup.
Hello,
I am wondering if someone has got any ideas that can help solve this
problem.
I have a dial plan that you call into, and depending on certain conditions
it calls out on a number grabbed from a database.
Something like this :
exten = s,n,Do something
exten = s,n,Do something else
exten = s,n,Dial(ZAP/g1/${OUTBOUND},${timeout})
I need to log the time the person was connected to $(OUTBOUND) , these are
duration and billsec in the CDR's
So at hangup I do something like this.
exten = h,1,DeadAGI(cdr-
outlogger.php|${CDR(start)}|${OUTBOUND}|${CDR(channel)}|${CDR(duration)}|$
{CDR(billsec)}|${CDR(disposition)}|${CDR(accountcode)})
Trouble is duration and billsec are *ALWAYS* 0 (zero), as if they have not
been loaded with the values, even though the channel is hung up.
Anyone got any ideas on how I can access ${CDR(duration)} and
${CDR(billsec)} in the hangup extension?
Thanks, hope I explained that well enough.
Mark
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RE: [Asterisk-Users] Grabbing the billsec and duration after a hangup.
The reason for it being 0 is because as long as you sit on the h extension the call is not yet done, therefore asterisk has no clue what those valuse are. If you use the h extension then you are messing up the CDR. So how can I tell it the call is complete and give the CDR values? Is it just not possible? 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
RE: [Asterisk-Users] PHPAGI
Here is a simple mysql snippet in php. Straight from the PHP manual.
http://www.php.net
$link = mysql_connect('HOST', 'UID', 'PASS') or die (Could not connect);
mysql_select_db('DATABASE_NAME') or die (Could not select database);
$query = SELECT * FROM table;
$result = mysql_query($query) or die(Query failed);
while ($line = mysql_fetch_array($result))
{
var_dump($line)
}
mysql_free_result($result);
mysql_close($link);
anybody have a little example on how do that???
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RE: [Asterisk-Users] PHPAGI
i have installed [EMAIL PROTECTED] i need install more soft? odbc etc...?, i think if [EMAIL PROTECTED] have inside the phpmyadmin i dont need more installed, this is true? I don't use [EMAIL PROTECTED] , but if phpmyadmin is installed you be pretty sure that you have all you need to connect to a mysql database. 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
RE: [Asterisk-Users] phpagi
$dbconn = mysql_connect(127.0.0.1,vladimir,vladimir);
snip
kinda looks ok, but I would seriously consider putting as much code as you
can into the dialplan rather than forking a PHP script for all the user
interaction.
For example, I use asterisk to handle loads of fax 2 email accounts. on 4
E1's. Each E1 goes to
[incoming]
exten = _XXX.,1,Answer
exten = _XXX.,2,SetAccount(${DNID})
exten = _XXX.,3,AGI(callhandle.php)
exten = _XXX.,4,GotoIf($[${application} = none]?6:5)
exten = _XXX.,5,Goto(${application},s,1)
exten = _XXX.,6,Hangup
The callhandle.php script looks in a database for the number that's being
called and sets variables like application, email address (and speed --
that's still buggy at the moment)
I set up further applications or contexts to handle the calls
[unknownnumber]
exten = s,1,Playback(unknownnumber)
exten = s,2,Hangup
[fax2email]
exten = s,1,SetAccount(${DNID})
exten = s,n,Wait(2)
exten = s,n,Wait(2)
exten = s,n,Wait(2)
exten = s,n,Wait(2)
exten = s,n,Wait(2)
exten = s,n,Hangup
exten = fax,1,Set(FAXFILE=/var/lib/asterisk/fax/${UNIQUEID}.tif|${SPEED})
exten = fax,2,rxfax(${FAXFILE})
exten = fax,3,Hangup
exten = h,1,System(/var/lib/asterisk/agi-bin/email.php ${EMAIL} ${FAXFILE}
${RESCALE})
Now, I have only been use asterisk for about 6 months and this may not be
the best solution, but when I started I tried to get *everything* into the
PHP scripts and just use the dialplan for as little as I could. It just
didn`t work, I wasted hours on it. I guess some mistakes you just have to
learn yourself.
Some simple advice (that I have learnt) :
use the dialplan as much as possible.
develop and test your scripts outside of asterisk before using them.
bookmark http://www.voip-info.org/tiki-index.php?page=Asterisk , I refer to
it everyday.
Mark
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RE: [Asterisk-Users] Please help
Shouldn't they be in /var/lib/asterisk/sounds? By default? Where is that path coming from? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: 13 January 2006 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Please help look in /etc/asterisk-1.2.0/sounds/ and see if you have sounds in that directory. ___ --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] Fax baud rate
Does anyone know if it's possible to set the incoming and or outgoing fax baud rate in asterisk ? 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
[Asterisk-Users] Asterisk not picking up calls.
Hiya, anyone have an idea what I need to do to fix this, I have a TDM400P and asterisk 1.2, when I make a call to the system asterisk see the phone ringing and looks like it picks it up from the console, but the phone actually just continues to ring. I am thinking I have something silly in the config or the cable from the TDM400P to the phone socket is dodgy. Anyone got any thoughts? Mark -- Starting simple switch on 'Zap/4-1' Nov 21 15:15:27 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:28 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:31 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:33 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... -- Detected ring pattern: 386,366,266 -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Set(Zap/4-1, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing Set(Zap/4-1, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing BackGround(Zap/4-1, demo-congrats) in new stack -- Playing 'demo-congrats' (language 'en') Nov 21 15:15:36 WARNING[13716]: chan_zap.c:3904 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 localhost*CLI /etc/zaptel.conf loadzone=uk defaultzone=uk fxsks=4 /etc/asterisk/zapata.conf [channels] language=en callwaiting=no callprogress=no usecallerid=yes echocancel=yes echocancelwhenbridged=no usedistinctiveringdetection=yes busydetect=no rxgain=0.0 txgain=0.0 group=1 immediate=no context=inbound-from-pstn signalling=fxs_ks channel = 4 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk not picking up calls.
All, I thought I'd post the answer to this, After I found what the problem was. It was the cable from the TDM card to the phone socket. I used one that came with an old modem and it worked a charm :-) Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Ackroyd Sent: 21 November 2005 15:19 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk not picking up calls. Hiya, anyone have an idea what I need to do to fix this, I have a TDM400P and asterisk 1.2, when I make a call to the system asterisk see the phone ringing and looks like it picks it up from the console, but the phone actually just continues to ring. I am thinking I have something silly in the config or the cable from the TDM400P to the phone socket is dodgy. Anyone got any thoughts? Mark -- Starting simple switch on 'Zap/4-1' Nov 21 15:15:27 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:28 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:31 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:33 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... -- Detected ring pattern: 386,366,266 -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Set(Zap/4-1, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing Set(Zap/4-1, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing BackGround(Zap/4-1, demo-congrats) in new stack -- Playing 'demo-congrats' (language 'en') Nov 21 15:15:36 WARNING[13716]: chan_zap.c:3904 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 localhost*CLI /etc/zaptel.conf loadzone=uk defaultzone=uk fxsks=4 /etc/asterisk/zapata.conf [channels] language=en callwaiting=no callprogress=no usecallerid=yes echocancel=yes echocancelwhenbridged=no usedistinctiveringdetection=yes busydetect=no rxgain=0.0 txgain=0.0 group=1 immediate=no context=inbound-from-pstn signalling=fxs_ks channel = 4 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1 PRI slips on TE410P
Any idea how to disable it completely? You can disable it in the BIOS as well. Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel: chan_zap.c:6514 mkintf: Unable to open channel 1 : Operation not supported by device
I get this when I boot asterisk. I have a Wildcard TDM400P REV H with 1 FXO board on it. [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/usr/local/etc/asterisk/zapata.conf': Found Nov 8 22:11:51 WARNING[24698]: chan_zap.c:935 zt_open: Unable to specify channel 1: Operation not supported by device Nov 8 22:11:51 ERROR[24698]: chan_zap.c:6514 mkintf: Unable to open channel 1: Operation not supported by device here = 0, tmp-channel = 1, channel = 1 Nov 8 22:11:51 ERROR[24698]: chan_zap.c:10344 setup_zap: Unable to register channel '1' Nov 8 22:11:51 WARNING[24698]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 my ztcfg all looks good though... Keyword: [loadzone], Value: [uk] Keyword: [defaultzone], Value: [uk] Keyword: [fxoks], Value: [1] Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK BT IDSN30e 'pass through' with TE205P/AvayaArgentOffice?
You should also ensure the PRI is really configured for EuroISDN, many BT PRI's are actually UK ISDN which Asterisk doesn't support (it's an older version). I had a problem along these lines, when I first started with asterisk, the PRI was originally DASS2, but needed to be Q931 Full ETSI for it to work. In between we had it configured for Q931 1/2 ETSI and the outbound didn't work. What the actually difference is I don't know. Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI why oh why?
I have been learning AGI, and have got to grips with most of it, one thing just confuses me. I have written a PHP class that does input and outs stuff to the AGI. It all seems to work. However on a stream file command. It's a bit different. If I have a php file that just plays a file and hangs up, the dial and hang up are in the dialplan, so the php proggie looks alittle like this. $ast-out(STREAM FILE home \\); $ast-in(); If I run this, with AGI DEBUGGING on I get something like this. AGI Tx 200 result=1 // last line of the variables passed in. AGI Rx STREAM FILE home -- AGI Script callhandle.php completed, returning 0 -- Executing Hangup(SIP/blah, ) in new stack The program does not play the file, it simple hangs up. If I place a delay in PHP program $ast-out(STREAM FILE home \\); sleep(20); $ast-in(); I get.. AGI Tx 200 result=1 AGI Rx STREAM FILE home AGI Tx 200 result=0 endpos=62407 -- AGI Script callhandle.php completed, returning 0 -- Executing Hangup(SIP/blah, ) in new stack Surely the stream file function, should play the file and wait until a key press or finish before the script continues. ??? Is this what it's supposed to do or I have done something wrong? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FreeBSD 5.4 (Asterisk 1.0.9) compile error
Darren Wiebe wrote Did you do a make clean? I just, as in 1 hour ago, successfully installed 1.0.9 using the port on FreeBSD. Yeah, even deleted all the files in the asterisk ports , and refreshed it ports collection. Always fails to compile at this point. Am I missing a package dependency somewhere? Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.0.9/channels' gmake: *** [subdirs] Error 1 *** Error code 2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreeBSD 5.4 (Asterisk 1.0.9) compile error
Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.0.9/channels' gmake: *** [subdirs] Error 1 *** Error code 2 Anyone got any ideas? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] file.c:1073 ast_waitstream_full: Wait failed (Interrupted system call)
Hi I have a PHP agi-bin scripted called callhander.php and its setup to answer anything that comes into the PBX, In the script I am trying to the get the system to play a file called home which I know works, as I can get the Play function to work from the extensions.conf file. However within the script the STREAM FILE function give this message in a debug Jun 6 15:37:07 WARNING[776]: file.c:1073 ast_waitstream_full: Wait failed (Interrupted system call) The say numbers commands work no problem. The OS is FreeBSD 5.4 anyone got any idea how to fix this? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hi I have a PHP agi-bin scripted called callhander.php and its setup to answer anything that comes into the PBX, In the script I am trying to the get the system to play a file called home which I know works, as I can get the Play function to work from the extensions.conf file. However within the script the STREAM FILE function give this message in a debug Jun 6 15:37:07 WARNING[776]: file.c:1073 ast_waitstream_full: Wait failed (Interrupted system call) The say numbers commands work no problem. The OS is FreeBSD 5.4 anyone got any idea how to fix this? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
