Re: [asterisk-users] Circuit/channel Congestion
While this thread is from earlier in the week, I didn't see any responses asking if you were using a a crossover cable between your Asterisk box and the dialer. You'll have to research the cable specs if you aren't, but in the past I've gotten this working just fine with two asterisk boxes here. Paul Lincoln Zuljewic Silva wrote: Hello all. I have a Digium TE110P board and when I do a 'dial 4509' I got: Jul 24 11:44:33 NOTICE[8180]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) My zaptel.conf: span=1,0,0,cas,hdb3,crc4 bchan=1-15,17-31 dchan=16 My zapata.conf: [channels] context=demo priindication=outofband pridialplan=local prilocaldialplan=local overlapdial=yes immediate=no callprogress=yes busydetect=no switchtype=euroisdn signalling=pri_net group=1 callgroup=1 pickupgroup=1 channel = 1-15,17-31 My /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3//CRC4 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) My pri show span 1: Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: -1 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 My zap show channels: Chan Extension Context Language MusicOnHold pseudodemo 1demo 2demo 3demo 4demo 5demo 6demo 7demo 8demo 9demo 10demo 11demo 12demo 13demo 14demo 15demo 17demo 18demo 19demo 20demo 21demo 22demo 23demo 24demo 25demo 26demo 27demo 28demo 29demo 30demo 31demo Thanks a lot! Lincoln ___ --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] Sprint Nextel sueing over VoIP patents
Depending upon the patents in question, a few companies (Cisco comes to mind) may have prior art here. I know that a company cisco bought was doing VoIP in 1998, but no indications of which patents this is, or when they were filed. Paul trixter http://www.0xdecafbad.com wrote: Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge. Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. http://kansascity.bizjournals.com/kansascity/stories/2005/10/03/daily23.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A problem with queues
I'm assuming you are using Agent based queues... If so, in your dialplan what happens after your Dial with the timeout? Is this timeout greater than the queue timeout? If so I've seen this behavior on my system. My solution, always make the queue timeout less than the dial timeout. Paul Jorge Alayon wrote: Hello, I am implementing a small call center with 1 to 4 agents. For some reason I don't understand, if an agent is busy, and it is his/her turn in the queue round, asterisk gives an all destinations are busy message and hangs up the call. Agents are SIP lines registered with an audiocodes MP108FXS which registers each line independently. Ringing strategy is RoundRobin (most of this configured using AMPortal, but checked that is consistent with documentation on queues on the wiki). Supposedly, the roundrobin strategy is used on available agents, not busy ones, am I correct ? Where can the failure be ? The best way I was able to replicate the problem is using only two agents, one received a call that stayed on for several minutes, the second received another call that waqs a short one, and the third call did not reach the second agent as it wanted to reach the first agent (per roundrobin) and failed. I was expectin it to ring again on the second agent. Regards, Jorge A. ___ 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 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 Problems with 1.0.9
I'm experiencing rather severe problems with 1.0.9 (we've had to backrev to our last version we know works (1.0.5). We are running a single PRI line with a T100P card. After about 10 hours of asterisk running and the modules loaded we start hearing noise and stuttering on any call that passes over the PRI line. I've tried this with echo cancellation on and off with no difference. This is a new problem for us as 1.0.5 behaves perfectly in this regard (it has it own issues, but that's another story). We would like to move back to 1.0.9 however restarting out phone system (which is in production) every 10 hours isn't really an option. Is anyone experiencing any similar symptoms, and if not what information would the developers need to work on this. Please note that running unstable isn't an option as the only PRI line I have to play with at the moment is our main line. Paul ___ 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] Method not allowed error
One idea would be to convert this extension into a queue extension, and then set the strategy to be ringall. Paul Lenwood S. Sawyer, III wrote: I d/l CVS HEAD today and am getting the same error with a strange behavior. If I ring a number that calls two sip extensions and pick up the call on one of the extensions, then the other extension continues to ring indefinitely. Any ideas? Thanks, Lenny Sawyer Afzaal Mirza wrote: Hi, I am getting *“Got SIP response 405” Method not allowed”* error on CLI. I am also getting *Port restricted Cone NAT* error on my SJ phone. *Please help!* * * * * *Afzaal* ___ 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 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 noise
I'm wanting to know if anyone else is experiencing any noise with the 1.0.9 release. What we're hearing is what sounds like stutterering since we moved to 1.0.9. This issue wasn't present with 1.0.5 which we moved from. Any ideas? Paul ___ 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] missing cdr records
Rosario,
Unfortunately this problem doesn't just affect you, I'm also affected
and have been since 1.0.5. If you set the debugging high enough and use
mysql, you'll see the insert statements being generated by asterisk, but
they never make it to the DB.
I'm glad to know I'm not the only one affected. Any others experiencing
this problem or have a fix?
Paul
Rosario Pingaro wrote:
I am experiencing a very wired problem.
Some of my cdr are lost.
I use logging cdr to csv, mysql and odbc. But some of them are lost.
They miss in csv mysql and odbc, so i'm pretty sure it is related to
asterisk functioning.
I am running asterisk 1.0.7; this is simple configuration file:
extensions.conf
[general]
static=yes
writeprotect=no
[macro-gw-voipjet]
exten = s,1,SetCallerID(${CALLERIDNAME})
exten = s,2,Dial,IAX2/[EMAIL PROTECTED]/${ARG1}
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Noop
exten = _s-.,1,Congestion
[macro-gw-nufone]
exten = s,1,SetCallerID(${CALLERIDNAME})
exten = s,2,Dial,IAX2/[EMAIL PROTECTED]/${ARG1}
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Noop
exten = _s-.,1,Congestion
[ser]
; combinazione 81 - per provider americani - destinaione usa e canada
exten = _81.,1,Macro(gw-nufone,${EXTEN:1}) ; NuFone
exten = _81.,2,Macro(gw-voipjet,${EXTEN:1}) ; VoipJet.com
exten = _81.,3,Congestion
; combinazione 8011 - per provider americani - destinaione
rotteinternazionali
exten = _8011.,1,Macro(gw-voipjet,${EXTEN:1}) ; VoipJet.com
exten = _8011.,2,Macro(gw-nufone,${EXTEN:1}) ; NuFone
exten = _8011.,3,Congestion
the percentage of cdr lost is around 5% and they are pretty concentrate
in the meaning that if I loose 5 cdrs they are lost 3 in around 2
minutes interval and 2 in anothe short interval.
Any advice on how to debug ?
thanks
Rosario
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Re: [Asterisk-Users] MCI vs. XO/Allegiance
I'm using an XO pri, and as long as you never change anything on the pri XO's not bad. Our experience is that if you chance anything on the PRI configuration they'll screw it up somehow (YMMV). One thing we have learned is that XO doesn't monitor our voice circuits, so if one of our PRI's goes down, we have to notify them almost immediately or they decommission it so the alert goes away. Paul Wiley Siler wrote: Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___ 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 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] Toll Free DIDs
If you're having them forward to local numbers then it would suprise me if they would do that. In the past when I've seen this done, its always been when we had trunks directly from the Long Distance company and they would pass us DNIS numbers to make everything match. Paul Hugh L. Johnson wrote: I have several toll free numbers that get forwarded to a single local number assigned to a trunkgroup. I've asked the telco to not forward those toll free numbers but to assign them as DIDs to the trunkgroup, so that I can differentiate via DNID. They said that they can't do that. That toll free numbers must forward. I know that I could have them each forward to different local DIDs assigned to the trunkgroup, but that just doesn't seem necessary. Is the telco correct? ___ 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 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] Voicemailbox on Queue?
Jimmy, If you look at: http://voip-info.org/wiki-Asterisk+cmd+Queue If you specify a context in the queues.conf it you can define where the caller goes if they press a button. Paul ___ 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 MySQL Blobs
In ReiserFS3, the performance loss for a directory containing 10's of thousands of files is negligible. I've personally had directories with 70,000+ files in them, and the performance has been stellar. Most traditional unix file systems break down around 5k-10k files, but I'd trust ReiserFS with more than 100k in a single directory. Paul Adam Goryachev wrote: On Mon, 2005-03-07 at 13:30 -0800, beonice wrote: --- Colin Anderson [EMAIL PROTECTED] wrote: The problem I suspect will arise is the number of inodes allowed by the file system. I don't know the exact size of the typical inode-max, but this will also presumably become an issue when the user tries to scale to really large amounts of faxes or voicemail! Would it help to split the db off to a separate server (that should reduce the CPU load on the asterisk server)? Any other alternatives? Anyone verified whether the BLOB storage solution breaks down first or the number of inodes runs out first? :) reiserfs doesn't have inodes, it also uses 'compression' for files smaller than the 'block' size. ie, say you use 4k blocks, and you need to store 4 x 1k files, then it will store all 4 files into a single block, and record that the one block includes 4 files etc so it is space efficient for lots of small files. It is also very efficient in the case where you have a lot of files in a single directory, hence you don't need to worry about that (as much) either. Though if you are expecting 10's of thousands of files, I'd still do some sort of directory hashing (actually, I'd probably test it and see)... Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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 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] Monitor Application with Queued calls
Due to management concerns our asterisk system has been setup to record
all phone calls for some time now (before the 1.0 release). Everything
was working fine until we upgraded 1.0.5 where all calls are recorded
except those that pass through a queue (we are not using the queue
record functionality because there are some minor issues with using it
in our scenario). Specifically, the Monitor app is run, the recording
begins, the Dial command is called, and as soon as the agent picks up
the phone the recording ends. Console logging of it occurring:
Asterisk 1.0.5, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk 1.0.5 currently running on cm02 (pid = 24466)
Verbosity is at least 17
-- Accepting call from 'XX' to 'YY' on channel 0/2,
span 1
-- Executing Goto(Zap/2-1, neospire|s|1) in new stack
-- Goto (neospire,s,1)
-- Executing Wait(Zap/2-1, 1) in new stack
-- Executing Answer(Zap/2-1, ) in new stack
-- Executing Wait(Zap/2-1, 1) in new stack
-- Executing DigitTimeout(Zap/2-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/2-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing BackGround(Zap/2-1, neo-welcome-options) in new stack
-- Playing 'neo-welcome-options' (language 'en')
== CDR updated on Zap/2-1
-- Executing SetCIDName(Zap/2-1, NeoSpire Tech) in new stack
-- Executing Queue(Zap/2-1, tech-support|tT|||) in new stack
-- Started music on hold, class 'default', on Zap/2-1
-- Stopped music on hold on Zap/2-1
-- Playing 'queue-youarenext' (language 'en')
-- Told Zap/2-1 in tech-support their queue position (which was 1)
-- Playing 'neo-queue-thankyou' (language 'en')
-- Started music on hold, class 'default', on Zap/2-1
-- outgoing agentcall, to agent '8137', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/8137
-- Executing Macro(Local/[EMAIL PROTECTED],2,
stdexten|8137|SIP/8137) in new stack
-- Executing Monitor(Local/[EMAIL PROTECTED],2,
WAV|1109969569.15844|m) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/8137|18|tT|)
in new stack
-- Called 8137
-- SIP/8137-92b5 is ringing
-- Agent/8137 is ringing
-- SIP/8137-92b5 answered Local/[EMAIL PROTECTED],2
-- Agent/8137 answered Zap/2-1
-- Stopped music on hold on Zap/2-1
monitor executing ( nice -n 19 soxmix
/var/spool/asterisk/monitor/1109969569.15844-in.WAV
/var/spool/asterisk/monitor/1109969569.15844-out.WAV
/var/spool/asterisk/monitor/1109969569.15844.WAV rm -f
/var/spool/asterisk/monitor/1109969569.15844-* )
-- Channel 0/2, span 1 got hangup
-- Hungup 'Zap/2-1'
While it may not be completely obvious by the above, the soxmix is
getting run as soon as the agent answers the phone.
Dialing rules to get to this point:
[incoming]
;
; Context the zap channels are on
;
; Local Number
exten = YY,1,Goto(neospire,s,1)
[neospire]
;
; Main NeoSpire Greeting Menu
;
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,DigitTimeout(5)
exten = s,5,ResponseTimeout(10)
exten = s,6,Background(neo-welcome-options)
; Menu choices
; 1 - sales
; 2 - tech support
; 3 - billing
; 9 - company directory (have to write this)
; 0 - operator
; extension
; Replay menu if they don't respond in time
exten = t,1,Goto,s,6
; Menu prompts
; Tech support goes to the queue
exten = 2,1,SetCIDName(NeoSpire Tech);
exten = 2,2,Queue(tech-support|tT|||)
; Play invalid selection message
exten = i,1,Playback,invalid
exten = i,2,Goto,s,6
Whene Queue actually dials an agents it actually calls a macro (stdext):
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
; Ring the interface, 18 seconds maximum
: Note: This timeout must be greater than the queue timeout otherwise queue
;calls roll to voicemail
exten = s,1,Monitor(WAV,${UNIQUEID},m)
exten = s,2,Dial(${ARG2},18,tT,)
; Send to voicemail
exten = s,3,StopMonitor()
exten = s,4,Voicemail(u${ARG1})
exten = s,5,Hangup
; If unavailable, send to voicemail w/ unavail announce
exten = s,103,StopMonitor()
exten = s,104,Voicemail(b${ARG1})
exten = s,105,Hangup
; If they press *, send the user into VoicemailMain
exten = a,1,VoicemailMain(${ARG1})
It should also be noted, that this all worked correctly until we
upgraded to 1.0.5. Downgrading back down to 1.0.2 (the last release we
were on) isn't really an option because 1.0.2 crashed for us regularly,
1.0.5 while stable, doesn't record going to our agents. All other calls
are recorded just fine.
Anybody have any ideas, or is it time for me to post this on
bugs.digium.com?
Paul
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[Asterisk-Users] Asterisk 1.0.5 an MySQL CDR
Is anyone else seeing any problems with CDR when using MySQL, specifically dropped legs of the call? ie: +-+-++-+ | calldate| disposition | lastapp| channel | +-+-++-+ | 2005-02-17 12:44:03 | ANSWERED| Hangup | Zap/2-1 | | 2005-02-17 12:42:03 | ANSWERED| Hangup | Zap/1-1 | | 2005-02-17 12:40:03 | ANSWERED| Hangup | Zap/1-1 | | 2005-02-17 12:38:04 | ANSWERED| Hangup | Zap/2-1 | | 2005-02-17 12:38:03 | ANSWERED| BackGround | Zap/3-1 | | 2005-02-17 12:36:04 | ANSWERED| Hangup | Zap/2-1 | | 2005-02-17 12:36:02 | ANSWERED| BackGround | Zap/3-1 | | 2005-02-17 12:34:03 | ANSWERED| Hangup | Zap/2-1 | +-+-++-+ Each of these calls should contain both a BackGround and Hangup lastapp, yet the first 3 do. I'm seeing this during the day when we're taking more calls, and it seems to be progressive (ie. the longer asterisk is running the worse it gets). At night (even though these calls are continuing to happen, and they're supposed to) I'm not seeing these problems. Any ideas what this could be? Paul ___ 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] cdr_mysql losing logs
I noticed a problem this morning with our cdr logging. We have a cron
job that places a call file into the spool directory having asterisk
call itself to check to make sure its still handling incoming calls
correctly, then queries the CDR database in mysql and makes sure that
appropriate records exist.
I can confirm that the call is happening correctly, but I'm missing
records in the database:
select calldate, disposition, lastapp, channel from cdr where clid =
xx order by calldate desc limit 45;
| 2005-02-14 11:34:04 | ANSWERED| Hangup | Zap/2-1 |
| 2005-02-14 11:34:03 | ANSWERED| BackGround | Zap/3-1 |
| 2005-02-14 11:32:03 | ANSWERED| Hangup | Zap/1-1 |
| 2005-02-14 11:30:04 | ANSWERED| Hangup | Zap/3-1 |
| 2005-02-14 11:30:02 | ANSWERED| BackGround | Zap/4-1 |
Notice the missing BackGround entry from the 11:32 call.
The asterisk console logs for this same duration:
^M-- Attempting call on Zap/g1/2144680768 for [EMAIL PROTECTED]:1 (Retry 1)
Using channel 1
Urgent handler
Urgent handler
^M-- Remote UNIX connection disconnected
^M-- Accepting call from '' to '2144680768' on channel 0/3, span 1
Enabled echo cancellation on channel 3
Launching 'Goto'
^M-- Executing Goto(Zap/3-1, neospire|s|1) in new stack
^M-- Goto (neospire,s,1)
Launching 'Wait'
^M-- Executing Wait(Zap/3-1, 1) in new stack
Difference is 1120, ms is 160
Write returned -1 (Resource temporarily unavailable) on channel 2
Write returned -1 (Resource temporarily unavailable) on channel 2
Write returned -1 (Resource temporarily unavailable) on channel 2
Write returned -1 (Resource temporarily unavailable) on channel 2
Launching 'Answer'
^M-- Executing Answer(Zap/3-1, ) in new stack
Urgent handler
Launching 'Wait'
^M-- Executing Wait(Zap/3-1, 1) in new stack
Enabled echo cancellation on channel 1
Dropping duplicate answer!
^MChannel Zap/1-1 was answered.
Launching 'StopMonitor'
^M-- Executing StopMonitor(Zap/1-1, ) in new stack
Launching 'Answer'
^M-- Executing Answer(Zap/1-1, ) in new stack
Launching 'Playback'
^M-- Executing Playback(Zap/1-1, 30seconds) in new stack
Set channel Zap/1-1 to write format gsm
Scheduling timer at 160 sample intervals
^M-- Playing '30seconds' (language 'en')
Launching 'DigitTimeout'
^M-- Executing DigitTimeout(Zap/3-1, 5) in new stack
^M-- Set Digit Timeout to 5
Launching 'ResponseTimeout'
^M-- Executing ResponseTimeout(Zap/3-1, 10) in new stack
^M-- Set Response Timeout to 10
Launching 'BackGround'
^M-- Executing BackGround(Zap/3-1, neo-welcome-options) in new stack
Set channel Zap/3-1 to write format gsm
Scheduling timer at 160 sample intervals
^M-- Playing 'neo-welcome-options' (language 'en')
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/3-1 to write format ulaw
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
Launching 'Hangup'
^M-- Executing Hangup(Zap/1-1, ) in new stack
Spawn extension (neospire,6501,4) exited non-zero on 'Zap/1-1'
cdr_mysql: inserting a CDR record.
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcont
ext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,ac
countcode,uniqueid,userfield) VALUES ('2005-02-14
11:32:03','2147201442','214720
1442','6501','neospire',
'Zap/1-1','','Hangup','',16,16,'ANSWERED',3,'','1108402
321.10557','')
Hanging up channel 'Zap/1-1'
zt_hangup(Zap/1-1)
Set option AUDIO MODE, value: ON(1) on Zap/1-1
Hangup: channel: 1 index = 0, normal = 13, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
Urgent handler
disabled echo cancellation on channel 1
Set option TDD MODE, value: OFF(0) on Zap/1-1
Updated conferencing on 1, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/1-1
disabled echo cancellation on channel 1
^M-- Hungup 'Zap/1-1'
Urgent handler
Call completed to Zap/g1/2144680768
^M-- Channel 0/3, span 1 got hangup
cdr_mysql: inserting a CDR record.
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcont
ext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,ac
countcode,uniqueid,userfield) VALUES ('2005-02-14
11:32:01','','','s','neospire'
,
'Zap/3-1','','BackGround','neo-welcome-options',18,17,'ANSWERED',3,'','1108402
321.10558','')
Hanging up channel 'Zap/3-1'
zt_hangup(Zap/3-1)
Set option AUDIO MODE, value: ON(1) on Zap/3-1
Hangup: channel: 3 index = 0, normal = 15, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
Urgent handler
disabled echo cancellation on channel 3
Set option TDD MODE, value: OFF(0) on Zap/3-1
Updated conferencing on 3, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/3-1
disabled echo cancellation on channel 3
^M-- Hungup 'Zap/3-1'
Notice that
[Asterisk-Users] Phone Cutout Problems
I'm currently running an office (about 25 phones) of Cisco 7960G's running SIP back to my asterisk box running 1.0.1. The asterisk box is attached to the Telco via a PRI (via a T100P). I'm getting complaints that the phone calls are cutting out on people (both parties) at random intervals, for anywhere from a half second to 5 seconds. In review the recordings, there is no cutout occurring on there, both parties are being recorded by asterisk. Any ideas? Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents and Queues
I've just installed asterisk as a new phone system for our office but am having difficulty with the queues. Specifically I need a way to redirect our sales queue to voicemail when no one is logged in to the queue. I see I can use the joinonempty=no setting, however this setting doesn't work if you use the agent functionality (at least not with AgentCallbackLogin). I could, of course use the AddQueueMember/RemoveQueueMember, however my experience with our version (as well as several previous versions) is once an extension is ringing, it will continue ringing that same extension forever (tried for 5-10 minutes). Can anyone think of a way to accomplish what I want without using the Queue timeout parameter (when someone's logged in and taking phone calls, calls need to stay in the queue)? My config details are below: All phones are Cisco 7960g's running SIP (v. 6.3) I'm running version: Asterisk CVS-HEAD-09/11/04-15:55:23 queues.conf: [general] [sales] music=default strategy=leastrecent timeout=15 retry=5 maxlen=0 announce-frequency=60 announce-holdtime=yes queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-holdtime = queue-holdtime queue-minutes = queue-minutes queue-thankyou = queue-thankyou monitor-format = wav49 monitor-join = yes ; Members of this queue member = Agent/8108 member = Agent/8153 agents.conf: [agents] ackcall=no autologoff=15 ; Sales agent = 8108,8108,agent 1 agent = 8153,8153,agent 2 extensions.conf: exten = 1,1,SetCIDName(Sales Queue) exten = 1,2,Queue(sales-neospire|t|||) Anyone have any ideas? Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
