[asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- 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] Sip Trunk takes incomming calls for 2 minutes and then stops
do you use the qualify=yes option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- 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] Sip Trunk takes incomming calls for 2 minutes and then stops
2010/1/26 Yves Arikoglu yves...@gmx.de: do you use the qualify=yes No, If I do it does not work at all. I've found if I set defaultexpiry to 30 it works fine. and was infact working for 30 seconds every two minutes before, It looks like sipgate.co.uk are expiring there registry attempts very quickly. Peter option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- 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] Error and call drops
Hi, does anyone have an info into what could cause [Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:26:23] ERROR[13098] utils.c: write() returned error: Broken pipe Is it a write process or a problem with one of the scripts I am running? I am seeing this over and over again and experience call drops on a percentage of calls. Thanks Lee -- _ -- 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] Sip Trunk takes incomming calls for 2 minutes and then stops
2010/1/26 Peter Childs pchi...@bcs.org: 2010/1/26 Yves Arikoglu yves...@gmx.de: do you use the qualify=yes No, If I do it does not work at all. I've found if I set defaultexpiry to 30 it works fine. and was infact working for 30 seconds every two minutes before, It looks like sipgate.co.uk are expiring there registry attempts very quickly. However I'm not totally sure this fixes the whole problem, as it still only works sometimes. Its just its works more often now than it did before. Peter. Peter option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- 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] settings for soft phones
I have always experienced very good call quality via my asterisk server snom phones. With soft phones especially on mobile (sipdroid or asip) quality is poor. With commercial sip services on my mobile - that do not of course use my asterisk server but use the same good wifi connection - quality is very good and much better than I achieve with a sip client pointing to my own server. Are there some things that I could be missing on the asterisk configuration that could improve the call quality for sip clients on mobile device. -- - Eric Smith -- _ -- 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] Detected digit 'f'
Kingsley Tart wrote: Thanks for the link. I looked at that page but couldn't see how it helped with my specific issue, unfortunately, though I admit I'm fairly new to asterisk so I don't fully understand what's going on. The page gave an indication, as Lee stated, that you'd see this error with fax detection turned on and there was no fax context created. Turning off the fax detection on incoming calls would mean that you'd have to use a dedicated number to route incoming calls to your fax receive code. Doug -- _ -- 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] How to make SpeechBackgroundkeepplayingifutterance doesn't match our grammar
AIR, the $GARBAGE sort of forces a match. The way I test a
grammar/background is something like this:
(this is longer because it uses Waitexten when no lumenvox license is
available)
[main-menu-select]
exten = s,1(start_menu),noop(select 0-9 except 8)
exten = s,n,Gotoif($[${usedtmf} = 1]?no-voice-menu)
exten = s,n,SpeechLoadGrammar(menu|/etc/asterisk/grammars/menu.gram)
exten = s,n,SpeechActivateGrammar(menu)
exten = s,n,Set(SPEECH_DTMF_MAXLEN=1)
exten =
s,n(say_menu),SpeechBackground(${Playmenu}${Playxfer}${Pressxfer}${Playns
f}${Pressnsf}${Playimg}${Pressimg}${Playpin}${Presspin}${Playmsg}${Pr
essmsg}${Opermsg}${Selmsg}|10)
exten = s,n,noop(Verbose(speech results ${SPEECH(results)}))
exten = s,n,noop(Verbose(heard ${SPEECH_TEXT(0)} ))
exten = s,n,GotoIf($[${SPEECH(results)} = 0]?:repeat_menu)
exten = s,n,Playback(telbank/pleasesayagain)
exten = s,n,Goto(say_menu)
exten = s,n(repeat_menu),Set(TEXT=${SPEECH_TEXT(0)})
exten = s,n,noop(score is ${SPEECH_SCORE(0)})
exten = s,n,GotoIf($[${SPEECH_SCORE(0)} = ${THRESHOLD}]?:spoke_menu)
exten = s,n,Gosub(bank-speech-confirm,s,1,(${SPEECH_TEXT(0)}))
exten = s,n,GotoIf($[${TEXT} = 99]?end-call|s|1)
exten = s,n,GotoIf($[${TEXT} = 1]?expect1)
exten = s,n,GotoIf($[${TEXT} = 2]?expect1)
exten = s,n,GotoIf($[${TEXT} = 3]?expect1)
exten = s,n,GotoIf($[${TEXT} = 4]?expect1)
exten = s,n,GotoIf($[${TEXT} = 5]?expect1)
exten = s,n,GotoIf($[${TEXT} = 6]?expect1)
exten = s,n,GotoIf($[${TEXT} = 7]?expect1)
exten = s,n,GotoIf($[${TEXT} = 9]?expect1)
exten = s,n,GotoIf($[${TEXT} = 0]?expect1)
exten = s,n,GotoIf($[${CONFIRM} = no]?:spoke_menu)
exten = s,n,Goto(say_menu)
exten = s,n(spoke_menu),Set(SEL=${TEXT})
;test one
exten = s,n,GotoIf($[${TEXT} = 99]?end-call|s|1)
exten = s,n,GotoIf($[${TEXT} = 1]?expect1)
exten = s,n,GotoIf($[${TEXT} = 2]?expect1)
exten = s,n,GotoIf($[${TEXT} = 3]?expect1)
exten = s,n,GotoIf($[${TEXT} = 4]?expect1)
exten = s,n,GotoIf($[${TEXT} = 5]?expect1)
exten = s,n,GotoIf($[${TEXT} = 4]?expect1)
exten = s,n,GotoIf($[${TEXT} = 5]?expect1)
exten = s,n,GotoIf($[${TEXT} = 6]?expect1)
exten = s,n,GotoIf($[${TEXT} = 7]?expect1)
exten = s,n,GotoIf($[${TEXT} = 9]?expect1)
exten = s,n,GotoIf($[${TEXT} = 0]?expect1)
exten = s,n,Playback(pbx-invalid)
;exten = s,n,SayNumber(${TEXT})
exten = s,n,Goto(main-menu-select|s|1)
exten = s,n(expect1),noop(${TEXT})
exten = s,n,SpeechDeactivateGrammar(menu)
exten = s,n,SpeechUnloadGrammar(menu)
exten = s,n,Set(MENUPRESS=${TEXT})
exten = s,n,Goto(ffb|s|select-return)
exten =
s,n(no-voice-menu),Background(${Playmenu}${Playxfer}${Pressxfer}${Playnsf
}${Pressnsf}${Playimg}${Pressimg}${Playpin}${Presspin}${Playmsg}${Pre
ssmsg}${Opermsg}${Selmsg}|3)
exten = s,n,WaitExten(3,m)
exten = s,n,Goto(main-menu-select|s|1)
exten = 0,1,Set(MENUPRESS=0)
exten = 0,n,Goto(ffb|s|select-return)
exten = 1,1,Set(MENUPRESS=1)
exten = 1,n,Goto(ffb|s|select-return)
exten = 2,1,Set(MENUPRESS=2)
exten = 2,n,Goto(ffb|s|select-return)
exten = 3,1,Set(MENUPRESS=3)
exten = 3,n,Goto(ffb|s|select-return)
exten = 4,1,Set(MENUPRESS=4)
exten = 4,n,Goto(ffb|s|select-return)
exten = 5,1,Set(MENUPRESS=5)
exten = 5,n,Goto(ffb|s|select-return)
exten = 6,1,Set(MENUPRESS=6)
exten = 6,n,Goto(ffb|s|select-return)
exten = 7,1,Set(MENUPRESS=7)
exten = 7,n,Goto(ffb|s|select-return)
exten = 9,1,Set(MENUPRESS=9)
exten = 9,n,Goto(ffb|s|select-return)
exten = i,1,Goto(main-menu-select|s|1)
exten = *,1,Goto(main-menu-select|s|1)
exten = t,1,Goto(main-menu-select|s|1)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Quinn Weaver
Sent: Monday, January 25, 2010 5:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make
SpeechBackgroundkeepplayingifutterance doesn't match our grammar
On Mon, Jan 25, 2010 at 2:56 PM, Danny Nicholas da...@debsinc.com wrote:
Since you're Perling it, why not just put the $sb_retval in a while loop
like this:
- my $response_good=0;
- my $sb_retval=undef;
- while (! $response_good) {
- my $tmp_retval = $c-agi-exec('SpeechBackground', $path);
- if ($tmp_retval eq 'play_next') {
$sb_retval=$tmp_retval;
$response_good=1;
}
...
}
If we did that, we'd be replaying $path from the beginning every time
the user said something that didn't match the grammar. For a podcast
episode like a radio show, that's badyou don't want to be 30 seconds
or two minutes into the content and have to start over.
Also, as I said, it's always matching one of the rules in our
grammar--even if I literally say goobledegook. So it's unclear how
we'd implement $response_good.
--
Quinn Weaver Consulting, LLC
Full-stack web design and development
http://quinnweaver.com/
510-520-5217
--
_
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Re: [asterisk-users] Disa not fully bridging outbound call
John Millican wrote: Hello, I have a situation where a remote worker dials in to the asterisk server, enters the secret code, then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] -- Moving call from channel 21 to channel 2 [Jan 25 17:51:40] -- Zap/0:2-1 answered Zap/1-1 [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] -- Native bridging Zap/1-1 and Zap/0:2-1 [Jan 25 17:51:49] -- Channel 0/1, span 1 got hangup request, cause 16 [Jan 25 17:51:49] -- Hungup 'Zap/0:2-1' [Jan 25 17:51:49] == Spawn extension (from-inside-redir, 16037649936, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Executing [...@from-inside-redir:1] Hangup(Zap/1-1, ) in new stack [Jan 25 17:51:49] == Spawn extension (from-inside-redir, h, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Hungup 'Zap/1-1' [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:8615 pri_fixup_principle: Call specified, but not found? [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:9759 pri_dchannel: Hangup on bad channel 0/2 on span 1 This says it is using DAHDI but it is actually still Zaptel as I have not had much success getting DAHDI to work on OpenSuSE, but that is another post for a later date. Any help is greatly appreciated. Thank You As an FYI reply to my own post I was able to clear up the issue by rmmod and restart of zaptel. Not what I would call a good solution but it worked. Does not tell me what caused the problem but at least the customer is happy for now. JohnM -- _ -- 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] [inter-pbx commnication] trying to make PBX1 talk to PBX2
Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: register = pbx1:p...@172.16.200.175 pbx1%3ap...@172.16.200.175 [pbx2] type=friend host=dynamic trunk=yes sercret=pass context=[default] ; i used the biggest context to avoid confusion as i don't master contexts deny=0.0.0.0/0.0.0.0 permit=172.16.200.175/255.255.255.128 when i type iax2 show peers i notice that pbx's are registred. of course still didn't attend my goal, do anybody have an idea how to make this happend?! -- Abdullah -- _ -- 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] queue
The idea is that the Queue() application uses different strategies to ring agents, so it decouples you from having to worry about that. You could have that by setting the queue to rinagll strategy. l. 2010/1/25 bhrugu mehta mehtabhr...@gmail.com Hi, all Is ther any way to pass channel queue such a way Queue(SIP/1001SIP/1002SIP/1003) thanks, Bhrugu Mehta -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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] [inter-pbx commnication] trying to make PBX1 talk to PBX2
This is how I did it.. I have to Servers, SRV1 and SRV2 In SRV1 iax.conf [SRV1-SRV2] type=peer username=SRV1-SRV2 secret=Password1 host=IP OF SRV2 qualify=yes [SRV2-SRV1] type=user username=SRV2-SRV1 secret=Password2 context=from-iax host=IP OF SRV2 quailfy=yes If I need to make calls on other box, I do Dial(IAX2/SRV1-SRV2/XX) where X is in destination from-iax context On SRV2 iax.conf [SRV1-SRV2] type=user username= SRV1-SRV2 secret=Password1 host=IP of SRV1 context=from-iax qualify=yes [SRV2-SRV1] type=peer username= SRV2-SRV1 secret=Password2 host=IP of SRV1 qualify=yes And calls from Here to There are Dial(IAX2/SRV2-SRV1/X) where is in destination from-iax context From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: Tuesday, January 26, 2010 10:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] [inter-pbx commnication] trying to make PBX1 talk to PBX2 Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: ... ... ... when i type iax2 show peers i notice that pbx's are registred. of course still didn't attend my goal, do anybody have an idea how to make this happend?! -- Abdullah -- _ -- 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] Anyone going to HD Communications Summit - Europe Feb 12th?
I realize that many of you are too far away to consider it, but I know of a couple of people who are considering going. Is anyone tempted? I am planning on going and have a promo code for you if you'd like one. r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended Transfer with REFER
Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a timeout in the Dial command is not suitable unless I am able to tell somehow that the call in question is being forwarded (which is of course not the case, as the Dial command is called befer the REFER is sent). Can anyone think of a way to get the call back to the transferrer after this timeout? Best regards, Örn Arnarson -- _ -- 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] SIP Hard Phone with SMS
Johann: Do you know how is the SMS sent over the IP, does it use SIP INFO message or somthing like that? Regards, Juan Johann Steinwendtner wrote: randulo schrieb: 2009/10/9 "Juan E. Rodrguez" jerdg...@gmail.com: Does any one know about a SIP hard phone capable of sending SMS messages (Or a SIP MESSAGE) that could be read from Asterisk dial plan?? The Gigaset S675IP series of DECT/SIP phone has SMS capability, but not sure it can work with Asteris. Yes, they do. (app_sms) Make sure you have installed the latest FW. Before, they sent the SMS out on the analog port only. Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- _ -- 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] Problem with Digium card, not transfering outgoing calls [Solved]
Hello, I have found a number of postings about XXX Message longer than it should be?? XXX but I guess these problems have been fixed in the current versions. obviously this was not true. In the file q931.c of libpri-1.4.20.2 I had to remove line 3458: return -1; After recompiling libpri and dahdi everything seems to be okay. 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
Re: [asterisk-users] ASTSBINDIR not being picked up by safe_asterisk
When I run make install I don't see this file getting overwritten. Do I have to delete it to get this to happen? On 1/25/2010 7:06 PM, Tilghman Lesher wrote: On Monday 25 January 2010 08:52:45 Mark Hulber wrote: Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory Automatically restarting Asterisk. But I think this is just a side effect of not finding asterisk in the /usr/sbin directory in the first place. Anyone run across this or have an idea what might have happened? I don't know if it was a Redhat update issue or some change in my configuration or what. When I make the following change in safe_asterisk it works ok: ASTSBINDIR=__ASTERISK_SBIN_DIR__ ASTSBINDIR=/usr/sbin Sounds like you manually copied the safe_asterisk script to /usr/sbin, instead of relying on 'make install' to do it for you. The install target does some extra processing of the script for you. -- _ -- 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] ASTSBINDIR not being picked up by safe_asterisk
On Tuesday 26 January 2010 10:08:39 Mark Hulber wrote: On 1/25/2010 7:06 PM, Tilghman Lesher wrote: On Monday 25 January 2010 08:52:45 Mark Hulber wrote: Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory Automatically restarting Asterisk. But I think this is just a side effect of not finding asterisk in the /usr/sbin directory in the first place. Anyone run across this or have an idea what might have happened? I don't know if it was a Redhat update issue or some change in my configuration or what. When I make the following change in safe_asterisk it works ok: ASTSBINDIR=__ASTERISK_SBIN_DIR__ ASTSBINDIR=/usr/sbin Sounds like you manually copied the safe_asterisk script to /usr/sbin, instead of relying on 'make install' to do it for you. The install target does some extra processing of the script for you. When I run make install I don't see this file getting overwritten. Do I have to delete it to get this to happen? Correct. It's only created if it doesn't already exist. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error and call drops
I've found that I get this error when I don't properly listen for asterisk responses to my commands in my agi scripts. Anytime you send a command to asterisk from an agi script, asterisk sends a response to the script with the result of the command (i.e a 200 ok response if asterisk was able to properly execute the command). If your script doesn't properly handle these responses, you get the error mentioned below. It's never caused any of my calls to drop, though. Try turning on AGI debug to see if this is the case for you. Thanks, --Warren Selby On Jan 26, 2010, at 5:11 AM, Lee Archer lee.arc...@thebigword.com wrote: Hi, does anyone have an info into what could cause [Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:26:23] ERROR[13098] utils.c: write() returned error: Broken pipe Is it a write process or a problem with one of the scripts I am running? I am seeing this over and over again and experience call drops on a percentage of calls. Thanks Lee -- _ -- 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] Detected digit 'f'
On Tue, 2010-01-26 at 07:46 -0500, Doug Lytle wrote: Kingsley Tart wrote: Thanks for the link. I looked at that page but couldn't see how it helped with my specific issue, unfortunately, though I admit I'm fairly new to asterisk so I don't fully understand what's going on. The page gave an indication, as Lee stated, that you'd see this error with fax detection turned on and there was no fax context created. Turning off the fax detection on incoming calls would mean that you'd have to use a dedicated number to route incoming calls to your fax receive code. Ah OK I see what you mean now. In our application we do, as you suggest, have dedicated fax numbers that route to the fax receive code. We already have fax detection turned off in the dahdi config. I'm now even more baffled by the whole thing, seeing as the Detected digit 'f' message appears in the log after iaxmodem has already answered and it's presumably now talking to hylafax. I'm sorry to keep asking questions but I'm at my wit's end with this and have no idea what to look at next. I do really appreciate your comments. Any idea what I should look at next? I've double checked the configs and they're the same as a different server that is working fine with this. -- Cheers, Kingsley. -- _ -- 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] Detected digit 'f'
Kingsley- On Tue, 2010-01-26 at 07:46 -0500, Doug Lytle wrote: Kingsley Tart wrote: Thanks for the link. I looked at that page but couldn't see how it helped with my specific issue, unfortunately, though I admit I'm fairly new to asterisk so I don't fully understand what's going on. The page gave an indication, as Lee stated, that you'd see this error with fax detection turned on and there was no fax context created. Turning off the fax detection on incoming calls would mean that you'd have to use a dedicated number to route incoming calls to your fax receive code. Ah OK I see what you mean now. In our application we do, as you suggest, have dedicated fax numbers that route to the fax receive code. We already have fax detection turned off in the dahdi config. I'm now even more baffled by the whole thing, seeing as the Detected digit 'f' message appears in the log after iaxmodem has already answered and it's presumably now talking to hylafax. I'm sorry to keep asking questions but I'm at my wit's end with this and have no idea what to look at next. I do really appreciate your comments. Any idea what I should look at next? I've double checked the configs and they're the same as a different server that is working fine with this. How do you know for sure fax detection is turned off? It sounds to me like your changes to the dahdi config file are being ignored. Maybe put something in there that should cause an error or something clearly observable, then see whether that actually occurs. -Jeff -- _ -- 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] Attended Transfer with REFER
checkout ${BLINDTRANSFER}
On Tue, 26 Jan 2010 15:48:51 +, Örn Arnarson wrote: Hi guys,
I am wondering (and have been unable to find out thus far) whether
Asterisk sets some special channel variables or something when a call is
transfered with the REFER method. Basically, I'm trying to figure out if it
is possible to somehow get a transferred call back to the transferrer (as
it is done with the built-in atxfer) after X seconds (or an unsuccessful
attempt).
Using a timeout in the Dial command is not suitable unless I am able to
tell somehow that the call in question is being forwarded (which is of
course not the case, as the Dial command is called befer the REFER is
sent).
Can anyone think of a way to get the call back to the transferrer after
this timeout?
Best regards, Örn Arnarson
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Re: [asterisk-users] Detected digit 'f'
Jeff Brower wrote: How do you know for sure fax detection is turned off? It sounds to me like your changes to the dahdi config file are being ignored. Maybe put something in there that should cause an error or something clearly observable, then see whether that actually occurs. Or even easier, use 'dahdi show channel X' and see if faxdetect is indeed disabled. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attended Transfer with REFER
26 jan 2010 kl. 16.48 skrev Örn Arnarson: Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a timeout in the Dial command is not suitable unless I am able to tell somehow that the call in question is being forwarded (which is of course not the case, as the Dial command is called befer the REFER is sent). Can anyone think of a way to get the call back to the transferrer after this timeout? THe transferred call is sent to a context set with the channel variable TRANSFER_CONTEXT before you call DIAL(). In there, run DUMPCHAN to see which variables you have and then dial with a timeout. After the timeout, dial back. /O -- _ -- 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] ASTSBINDIR not being picked up by safe_asterisk
Great, do you know of any other files outside of /usr/lib/asterisk/modules that get recreated? I also place rc.redhat.asterisk as asterisk in /etc/rc.d/init.d I don't see that safe_asterisk_restart gets placed anywhere. It looks like astgenkey and autosupport both get written over. On 1/26/2010 11:15 AM, Tilghman Lesher wrote: On Tuesday 26 January 2010 10:08:39 Mark Hulber wrote: On 1/25/2010 7:06 PM, Tilghman Lesher wrote: On Monday 25 January 2010 08:52:45 Mark Hulber wrote: Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory Automatically restarting Asterisk. But I think this is just a side effect of not finding asterisk in the /usr/sbin directory in the first place. Anyone run across this or have an idea what might have happened? I don't know if it was a Redhat update issue or some change in my configuration or what. When I make the following change in safe_asterisk it works ok: ASTSBINDIR=__ASTERISK_SBIN_DIR__ ASTSBINDIR=/usr/sbin Sounds like you manually copied the safe_asterisk script to /usr/sbin, instead of relying on 'make install' to do it for you. The install target does some extra processing of the script for you. When I run make install I don't see this file getting overwritten. Do I have to delete it to get this to happen? Correct. It's only created if it doesn't already exist. -- _ -- 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 phone DND state
On Fri, Jan 22, 2010 at 7:50 AM, Mike l...@virtutel.ca wrote: I know having Asterisk aware of Polycom Do No Disturb state wasn't working before (1.4), but is this working in any recent version? Is there any custom way of doing this? Our Asterisk servers (1.2 and 1.4) get SIP response 603 Decline when our Polycoms are on DND. CP -- _ -- 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] Realtime Queue not work in 1.6.2.1
hi,all i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except realtime queue. it seems queue_table works fine, but queue_member_queue not work, the two tables works fine when in 1.4.28. is that something changed related to realtime queue configuration? more detail about two table definition and data stored in , please see: http://pastebin.com/m33f9539e the extconfig.conf file, please see: http://pastebin.com/m2008ced1 and the res_mysql.conf file: http://pastebin.com/m27d3fdc5 Could you tell me what's wrong with me ? Thanks! -- Best regards, Sucan -- _ -- 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 phone DND state
I am using 1.4.21.2 and DND is definitely working. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Saturday, 23 January 2010 2:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom phone DND state Hi, I know having Asterisk aware of Polycom Do No Disturb state wasn't working before (1.4), but is this working in any recent version? Is there any custom way of doing this? Regards, Mike -- _ -- 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] Snom vs Polycom
On Mon, Jan 25, 2010 at 12:07:55PM -0600, Karl Fife wrote: From: cb c...@mythtech.net Sent: Sunday, January 24, 2010 12:42 I use the Snom 370 all day long at work. I have never had a problem adjusting the volume. I change it multiple times a day as I keep my handset on one volume and my headset on another, so I'm always going up and down and I've never accidentally pressed any other key. I will however agree with you on the Mute button, any time I want to mute a call, I have to stop and look at the buttons and figure out which one of the tiny ones is mute. You are right, especially with practice. I still think it's a little easier if the buttons are distinct from one another by appearance, size and/or placement.. I think Polycom made a good design decision by not making you 'reach over' any buttons to press these common buttons. I also like the fact that the mute button turns bright red when activated. Come to think of it, I wish the DND button turned red when activated :-). Those type of arguments are basically why I stick with a soft-phone. I have a giga-bit and wifi phone. And the firmware is not an issue of a hit-or-miss. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error and call drops
Hi, does anyone have an info into what could cause [Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe I have had the same issue with a PHP script that logs into the manager interface. If you don't wait for the AMI response, then log off before closing the connection you will get those errors. I also haven't seen any call drops. I would urge you to check your scripts, and put some 2 second waits before a logoff and closing the socket and see if that helps. -- _ -- 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] MYSQL grammar diff in 1.6.2.1?
2010/1/26 Tilghman Lesher tles...@digium.com:
On Monday 25 January 2010 03:12:08 Zhang Shukun wrote:
hi, dear all
MYSQL commands work well in 1.4.28 edition, but not in 1.6.21
is that the grammar is different between them?
extensions.conf
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ companycode = ${ARG2} and callerid=${ARG1} and
blockenabled = 1)
cli:
-- Executing [...@macro-checkblacklist:2] MYSQL(SIP/1003-0006,
Query resultid 1 SELECT\ callerid\ from\ blacklist\ where\
companycode = 95040654321 and callerid=1003 and blockenabled = 1) in
new stack
[Jan 25 17:05:34] WARNING[2583]: app_addon_sql_mysql.c:374
aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an
error in your SQL syntax; check the manual that corresponds to your
MySQL server version for the right syntax to use near '\ callerid\
from\ blacklist\ where\ companycode = 95040654321 and callerid=1003 '
at line 1
I have no idea why you backslashed your spaces in 1.4, as that has never been
a requirement (as is evident later in the line, where you neglected them).
wow, i learn this method from voip-info and do the same as the example showed.
so i add backslash.
This is the problem, and if you remove the backslashes, you should be fine.
you are right! it really works now by remove backslash.
maybe in asterisk 1.4 add backslash or not is both working.
but in asterisk 1.6 , must not contain backslash. or it will not work any more.
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Re: [asterisk-users] MYSQL problem
2010/1/23 Steve Edwards asterisk@sedwards.com: On Fri, 22 Jan 2010, Zhang Shukun wrote: as you know, we can use MYSQL command to visit mysql database but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ? ODBC will do what you want. Thanks, while i think because oracle has no offical ODBC for linux system. is that better use mysql than oracle. considering the perfoermace and speed. many people around think mysql is not a good option for database, they think mysql is only suit for small business. but i want to have a try. i need to convince them to use this. Personally, I'd vote for an AGI using whatever C API your DB provides -- like Pro*C to access Oracle. You will have access to all of the features of your DB and your dialplan will be a lot cleaner and easier to maintain. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- 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] MySQL RealTime Error
2010/1/26 Carlos Chavez cur...@telecomabmex.com: You must read the upgrade instructions. The database definitions in res_mysql.conf have changed. The way you reference the database in extconfig.conf is also different. solved... it is my configuration error of res_mysql.conf and extconfig.conf file the database name not matched. On Mon, 2010-01-25 at 09:33 +, Ishfaq Malik wrote: What happens when you try the command mysql -uroot -proot asterisk Ish Zhang Shukun wrote: hi,all when i upgrade from 1.4.28 to 1.6.2.1, i can't connect to mysql database anymore, error as follow: [Jan 25 15:38:25] WARNING[3003]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk (check res_mysql.conf) the content of res_mysql.conf is: http://www.pastebin.org/81966 i've try command mysql -uroot -proot ,i can connect to mysql successfully. Could you tell me what's wrong with me ? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- Best regards, Sucan -- _ -- 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] GoToIfTime issue
2010/1/22 Tilghman Lesher tles...@digium.com: On Friday 22 January 2010 04:06:29 Zhang Shukun wrote: 2010/1/22 Randy R randulo2...@gmail.com: On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun bit...@gmail.com wrote: exten = 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1) but what should i do. if i want to set seperate weekdays,like mon,wed. not continuous weekday like mon-fri. I couldn't find any reference to multiple, non-contiguous days on a quick Google, but this would work at the cost of an extra line: exten = 222,1,GoToIfTime(11:00-14:00|mon|*|*?1:3,1) exten = 222,2,GoToIfTime(11:00-14:00|wed|*|*?1:3,1) Thank you, but why don't it to be comma seperate to represent seperate weekdays? as | mon,wed,fri | it's also very understandable. Starting in 1.6.2, you can use the ampersand to join days. when i update from 1.4.28 to 1.6.2.1, the join days problem solved. but not the realtime queue not work. should i change the queue_table and queue_member_table definition along with the upgrading? because i don't know how the two table match each other. mysql select * from queue_table; +--+---+-+ | name | beginworktime | endworktime | +--+---+-+ | 950401234561 | 09:30:00 | 17:30:00| | 950401234562 | 11:30:00 | 17:30:00| | 950401234563 | 16:30:00 | 17:30:00| +--+---+-+ 3 rows in set (0.00 sec) mysql select * from queue_member_table; +--++--+---+-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++--+---+-++ | 18 | Zhang Shukun | 950401234561 | SIP/1001 | 0 | 1 | | 19 | Li Aiwei | 950401234561 | SIP/1002 | 0 | 1 | | 20 | Zhang Jianming | 950401234561 | SIP/1003 | 0 | 1 | +--++--+---+-++ 3 rows in set (0.00 sec) how colume name queue_name in queue_member_table match name in queue_table? how does the system recognize them. i mean queue_name is not an configure option in agent.conf -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- 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] MYSQL problem
Un-mid-posting... On Fri, 22 Jan 2010, Zhang Shukun wrote: as you know, we can use MYSQL command to visit mysql database but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ? 2010/1/23 Steve Edwards asterisk@sedwards.com: ODBC will do what you want. Personally, I'd vote for an AGI using whatever C API your DB provides -- like Pro*C to access Oracle. You will have access to all of the features of your DB and your dialplan will be a lot cleaner and easier to maintain. On Wed, 27 Jan 2010, Zhang Shukun wrote: Thanks, while i think because oracle has no offical ODBC for linux system. is that better use mysql than oracle. considering the perfoermace and speed. many people around think mysql is not a good option for database, they think mysql is only suit for small business. but i want to have a try. i need to convince them to use this. So don't use ODBC, use Pro*C... Back when Yahoo was relevant, they ran on MySQL. Can you quantify your requirements (number of rows, queries per second, simultaneous connections) and test it on hardware similar to your production environment? While I'm sure Yahoo spent a lot of time and money designing and tuning their system, sometimes explain plan can point you to small changes that yield significant results. If your shop is committed to Oracle, can finance the licenses, and has the in-house talent -- use it. Nobody ever lost their job by buying IBM... -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
