Re: [Asterisk-Users] Transient SIP Registration Issues
Richard J. Sears wrote:
Hey Everyone -
I am having a problem that is keeping me awake at night.ok, so maybe
not keeping me awake, but it is frustrating. :-)
I am running Asterisk 1.0.7 on Gentoo (2.6.10-gentoo-r6) on an Intel
700Mhz box with 512MB of RAM.
The system is very light, with maybe 35 SIP and IAX connections. I am
using NuFone and Konfer for dialtone with no traditional TDM cards
installed at all. Overall system load is around .4 or less most of the
time.
Overall - a very simple configuration.
I am using (mostly) the Linksys PAP2-NA units for deployment. I
preconfigure the units, then ship them out to the people that need them.
I also have several of the Digium IAXy units in use.
The problem I am starting to see is that a person's extension will work
great, and then I will start to see failed registrations for their unit
over and over again. When this happens, the units fall offline. Then the
unit will magically reregister and start to work again.
I had assumed (initially) that it was a bad unit, so I replaced it, but
then it started to happen to other units as well.
When registered, the units in question have ping time under 50 to 60 ms,
and no latency associated with them. Packet loss is extremely minimal or
none at all.
Here is one example - I have included the relevant portions of my
sip.conf and extensions.conf:
sip.conf
[1028]
type=friend
username=GSynn
secret=
qualify=500
host=dynamic
fromuser=GSynn
dtmfmode=rfc2833
nat=yes
canreinvite=no
disallow=all
allow=g729
callerid=Gary Synn 3050
context=secure
extension.conf
[gary_synn]
exten = 3050,1,Macro(stdexten,SIP/1028)
[macro-stdexten]
;; ARG1 = Phone ID to dial
exten = s,1,NoOp(${CALLERID}) ; Grab Caller ID Info
exten = s,2,Playback(${VMDIR}/${MACRO_EXTEN}/greet); Grab the
dialed extension and play a greeting
exten = s,3,Dial(${ARG1},15,rtm)
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten = s-CHANUNAVAIL,1,Voicemail(u${MACRO_EXTEN}) ;if chan
unavail (sip phone not regisitered?)
exten = s-CONGESTION,1,Voicemail(u${MACRO_EXTEN}) ;if chan
congested
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
[secure]
include = pstn_outbound
include = system_extensions
Here are the errors that I see on the colsole -
Apr 4 17:35:38 NOTICE[27276]: chan_sip.c:7681 handle_request:
Registration from 'GSynn sip:[EMAIL PROTECTED]' failed for
'5.63.198.220'
Apr 4 17:35:42 NOTICE[27276]: chan_sip.c:7681 handle_request:
Registration from 'GSynn sip:[EMAIL PROTECTED]' failed for
'5.63.198.220'
Apr 4 17:35:46 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:35:50 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'DMadore sip:[EMAIL PROTECTED]' failed for '69.17.136.238'
Apr 4 17:35:50 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:35:54 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:35:58 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:02 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:33 NOTICE[27276]: chan_sip.c:7681 handle_request:
Registration from 'GSynn sip:[EMAIL PROTECTED]' failed for
'5.63.198.220'
Apr 4 17:36:34 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:37 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:40 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:44 NOTICE[27276]: chan_sip.c:7681 handle_request:
Registration from 'GSynn sip:[EMAIL PROTECTED]' failed for
'5.63.198.220'
Apr 4 17:36:48 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Apr 4 17:36:52 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration from
'GSynn sip:[EMAIL PROTECTED]' failed for '5.63.198.220'
Any ideas would be greatly appreciated.
Thanks !!
**
Richard J. Sears
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There is another thread that is dealing with the same symptom
as you are experiencing. Subj: Previous sip reload not yet done
It does
Re: [Asterisk-Users] realtime management for sip with mysql
I believe if you read the wiki, assuming you are successful in locating the correct entry, there is a statement indicating sip show peers will not reveal sip users. g William M. Sandiford wrote: Hi Bioz: I'm having the same problem with Realtime and CVS-HEAD from 4/4/2005. I haven't found a solution yet, but I have also posted this to the list in the last 24 hours and hopefully someone will help out soon. There was a similar thread from about a month ago that seemed to solve the problems of someone else with a similar problem, but it didn't help me, so you may want to look for it. I believe it was called Realtime is not working Regards, Bill PS, if you get it working, be sure to let me (and the list know) -- Hi I use realtime management for sip with mysql and it works correctly. After 1 or 2 hours having started, I look at sip show peers and I have no more sip customers registered. I must do a restart for sip working. Someone have an idea of what goes wrong ? Thanks by advance for your answer. Bioz ___ 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] Previous sip reload not yet done
administrator tootai wrote: Olle E. Johansson a écrit : administrator tootai wrote: Nabeel Jafferali a écrit : Does anyone else have this problem? Is there a workaround? Yeah, I had this problem when I added a lot of SIP register statements and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the problem. It seem * was getting stuck waiting for DNS lookups. Thanks. We put everywhere it was accepted the IP address and will see. FYI, sipgate.de doesn't accept to register with IP address. CLI SIP reload command is now applied much faster as with FQDNs in sip.conf Changing register= statements to IP addresses is a bad idea. SIP is domain name based and (as proved by sipgate) an IP address points to *one* host, whereas a SIP domain by using SRV records can point to many IP addresses and servers. There's a huge difference between sending a REGISTER to [EMAIL PROTECTED] and [EMAIL PROTECTED] See this as a short time fix. We need to make a better solution on the REGISTER parsing to prevent this from happening, it's clearly a bug. Well noticed. Should I concider bugs #3850 and #3933 including this matter or should I open a new one? We had the same problem, on two different hardware platforms. 2 flavors of pentium 4/board combos grandstream and sipura (handset/ata) devices the only thing that has worked for us was to eliminate the registration process all together. This has been going on since last October that I am aware of which means it has been in every cvs since then. sip.conf host=device ip (not dynamic) qualify=yes ___ 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] blind transfer question
Hello, When performing a blind transfer to another extension i.e. originating extension = 103 transfer extension = 101 # 101 as soon as the extension rings, the handset initiating (103) the transfer gives a busy tone (or congestion) once the transfer extension rings asterisk returns: SIP/101-71ec is ringing Got SIP response 486 Busy back from 192.168.1.2 SIP/103-7394 is busy question - Is there some way to force the originating handset to go silent then hang up? wiki has not yielded anything for me neither has google Regards g ___ 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] ${DIALSTATUS}
Manuel Schroeder wrote:
Hi list,
I try to explore making use of the variable ${DIALSTATUS} for
auto-answering purposes an similar.
On my asterisk box this does not work because ${DIALSTATUS} always
returns empty :(
Didn't find much in the web on this issue.
Can someone help?
regards Manuel
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It is my understanding, ${DIALSTATUS} is only filled when a
Dial command is initiated. or maybe I am misunderstanding your question
Regards
g
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Re: [Asterisk-Users] Is this possible?
Paul wrote: I'd like to setup my Asterisk box to receive a call on the incoming POTS line and immediately redirect back out to connect to another phone number. Im thinking I could use either the threeway feature of that POTS line, or a second POTS connected to a different FXO card. Does ANYONE know if this is possible and if so, how it's accomplished? 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 Assumption being you want the original line to become available for another call... Don't think this is possible in asterisk as you will still be tying up the original incoming pots line to make the call to another line. you will have to go further up the provider chain to where your pots originates. This would be a great feature g ___ 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] Vonage a provider?
Frank Abernathy wrote: I am new to the mailing list, but I am very interested in running my small home business office phone system using Asterisk. However, Broadvoice, a VoIP provider of choice based on my research, is not available in my area. I currently use Vonage VoIP. Their website mentions nothing about being able to link to Asterisk. I was wondering if any US subscribers have been able to configure Vonage with Asterisk. Or if anyone has found Vonage to be a non-compatible provider. When I spoke to them, the only help I got was to plug their device into an fxo port on the asterisk server and use it just as you would a pots line. wasn't what I was looking for. g ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ...
Ronald Wiplinger wrote: I try to get Realtime to work, ... the debug looks like below. Mar 12 00:56:56 DEBUG[25640]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '621' Mar 12 00:56:56 DEBUG[25640]: MySQL RealTime: Everything is fine. Mar 12 00:56:56 DEBUG[25640]: Unable to find key '621' in family 'SIP/Registry' Mar 12 00:56:56 DEBUG[25640]: Setting NAT on RTP to 524288 Mar 12 00:56:56 DEBUG[25640]: Exiting with DIALSTATUS=CONGESTION. Mar 12 00:56:56 DEBUG[25640]: /var/spool/asterisk/voicemail/other/621/unavail doesn't exist, doing what we can There are two things: 1. Unable to find key '621' in family 'SIP/Registry' where have I forgotten to set that? 2. /var/spool/asterisk/voicemail/other/621/unavail doesn't exist, doing what we can it is not there, because it is in /var/spool/asterisk/vm/621/ Where to correct that? bye Ronald We bailed on it for now, as it does not appear to be 100%. Phones would not re-register, calls would fail.(just a lot of headaches) We went back to using the config files and all settled down. It looks as tho the db addon is just that, patches to make it work instead of rewriting the thing from scratch with all storage in a database and not in conf files. You might try storing all of your config settings to a database and writing them out to the config dir when there are changes. I think someone wrote an app to do that. g ___ 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] Tired of trying to fix this echo problem
Martin Roy wrote: snip I'm tired of beeing unable to get rid correctly of the echo problem. I have 3 TDM04B installed in one server. We had to adjust the [rx | tx}gain settings in zapata.conf for a couple of phones to get rid of the echo. Most is gone. you might try setting the tx to a less than zero db value while keeping rx at zero for starters. g ___ 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] NuFone + VoIPJet = busy busy busy
Jean-Michel Hiver wrote: Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! How are you determining a fallback condition from one voip to another? greg ___ 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] NuFone + VoIPJet = busy busy busy
Andrew Kohlsmith wrote:
On March 9, 2005 10:43 am, Cirelle Internet Products wrote:
How are you determining a fallback condition from one voip to another?
Mine's rather simple but it works well:
[macro-nufone-dial]
exten = s,1,GotoIf($[$ACCOUNTCODE != ],s,gotac)
exten = s,n,SetVar(ACCOUNTCODE=${ARG2})
exten = s,n,GotoIf($[{$ARG2} != ],s,gotac)
exten = s,n,SetVar(ACCOUNTCODE=benshaw)
exten = s,n(gotac),SetAccount(${ACCOUNTCODE})
exten = s,n,GotoIf($[${LEN(${ARG1})} = 10]?s,add1)
exten = s,n,Dial(IAX2/[EMAIL PROTECTED]/${ARG1},,g)
exten = s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = s,n(add1),Dial(IAX2/[EMAIL PROTECTED]/1${ARG1},,g)
exten = s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = dial-CANCEL,1,Hangup
exten = dial-ANSWER,1,Hangup
exten = dial-NOANSWER,1,Hangup
exten = dial-BUSY,1,Busy
exten = dial-CONGESTION,1,Macro(pri-dial,${ARG1},${ARG2})
exten = dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})
; handle NXX-NXX-, 1-NXX-NXX- and 011...
[nufone]
exten = _NXXNXX,1,Macro(nufone-dial,${EXTEN})
exten = _1NXXNXX,1,Macro(nufone-dial,${EXTEN:1})
exten = _011.,1,Macro(nufone-dial,${EXTEN})
You can ignore the accountcode stuff, we handle calls for several businesses
so I sort the accounting out that way.
For contexts that I want to have calls go out to Nufone I include the 'nufone'
context. As you can see, it handles 10-digit, 11-digit and international
(variable-digit) extensions.
Basically if it's a 10-digit #, add a '1' to it. Then attempt to Dial()
through my Nufone account. You'll notice the 'g' flag to the Dial()
application which tells it to go on in context after a hangup. I then check
the status of DIALSTATUS and if the result was CONGESTION or CHANUNAVAIL I
fall back and dial out my PRI.
Personally I think that CONGESTION should never be returned unless the other
side SAYS piss off, I'm too busy to handle your call but IAX will throw
back a CONGESTION status if it can't reach the other side, which is why I
have to check for both CONGESTION and CHANUNAVAIL.
-A.
Thanx Andrew
greg
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[Asterisk-Users] determining an available channel question
Hello,
I have an outgoing dial plan which utilizes FWD for any of our
outgoing 800,877,866, etc while, toll and local calls get routed
to one of our pots lines.
when we use the pots lines, we use the chanisavail function
to choose an available pots line.
for example
exten = _1800NXX,1,ChanIsAvail(Zap/26Zap/25)
exten = _1800NXX,2,Cut(theChannel=AVAILCHAN,,1)
exten = _1800NXX,3,Dial(${theChannel}/w${EXTEN})
works like a champ
Recently, we have had to stop using FWD because at times the call
cannot be processed (due to load?? probably). But there is a
voice response indicating the call cannot go through.
Currently, all tollfree calls chew up a pots line.
When this happens, I would like the call to failover to one of
the available pots lines.
ChanIsAvail is supposed to work with SIP from what I read, but
since FWD answers, the call is complete.
Has anybody confronted this type of configuration with any
success? Is it possible? If so, a pointer would be appreciated.
Best Regards
Greg
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Re: [Asterisk-Users] LiveVoIP Problems?
Asterisk guy wrote: www.mutualphone.com This company only accepts CC via PayPal doesn't sound good to me, right up there with shopping on ebay. No published address, service by calling card. Not sure about this one. Lots of red flags. I guess if it sounds too good to be true, it is. My 2cents Greg ___ 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] LiveVoIP Problems?
Jay Milk wrote: snip and you need rock-solid performance, there are a couple of contenders out there. My most problem-free provider so far has been Vonage -- they're not very flexible, and not very open to work with their customers, but that's probably why their service has the best uptime of all the ones I used so far. Broadvoice -- read thread. Iax.cc started off promising, but it's getting spotty in places. Myphonecompany.com so far (going on three weeks) has a solid track record. Only one issue so far, and that was on my end. Aren't these just Retailers? Greg ___ 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] ANNOUNCEMENT : Asterisk-Stat V2.0 - CDR Analyser
Areski wrote: Hi Greg, How many calls do you have by hours ? BR, Areski Hi Areski, Some hours 0 most hours 1-3 with bursts of 7 - 14 Not a lot of traffic on the dates I had checked. Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Mark Eissler wrote: Hasn't anyone noticed that LiveVoip seems to happily blame just about everything on Asterisk? FWIW, I have experienced the same type of problem on a Sprint cell phone and also using a residential VOIP account with Broadvox. Both were able to correct the problem at THEIR end. Since no one else on this list seems to be complaining about the problem using provider's other than LV, I would suggest sacking them and getting DIDs from some other place. Seems like that is always the first thing they suggest too so they must not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2 Feb 2005 13:25:29 -0500, Brian Dingman [EMAIL PROTECTED] wrote: Finally got a reply from LV support. Not what I was hoping for. Hopefully they will file a bug with Digium since they investigated the issue not holding my breath. Since this is such basic * functionality that they can't seem to accomplish I would think twice before aquiring DID's from them. LiveVoip Support Our people have looked into this matter over the past few days. They tell me that it is a problem with Asterisk. We are not going to be able to help you with this. If you would like a refund so that you can migrate to another service provider we will be happy to do so. With each rev. of Asterisk more and more improvements are made. At some point these issues may resolve but, for the time being it is not a problem we can help you with. On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier [EMAIL PROTECTED] wrote: I just got a couple of numbers (activated Friday) from livevoip, I am having similar issues. When you call the number, I get ring back, but as soon as IVR picks up, I should here extensioni I don't hear that but then I dial an extension number and there is no ring back. I don't have this issue from other voip providers. Steve ___ 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 -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.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 is this using their asterisk city, or just a straight sip account?? Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip
Ed Greenberg wrote: --On Friday, March 04, 2005 11:58 AM -0600 James Taylor [EMAIL PROTECTED] wrote: It would be nice if they told us what the problem with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The LiveVoip people claim that they have been ignored on the Asterisk List and they indeed blame Asterisk for everything from lost dtmf to other failures. That said, they are the only company I've found that offers inbound DIDs with multiple simultaneous calls, suitable for a call center or calling card application. Most others limit you to one, or a small few, inbound paths. They (Level 3, actually) also have the widest coverage for DIDs in the US. At the current level of service, LiveVoip is not going to get my business. If I can find anybody else to provide my inbound service, I'm very interested in talking to them. /edg ___ 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 I believe LiveVOIP is a reseller of Level 3. From what I understand, you need to buy millions of minutes to get decent pricing at Level 3 as they are a mega wholesaler... I may be wrong, but that's what I got out of it. Greg ___ 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] ANNOUNCEMENT : Asterisk-Stat V2.0 - CDR Analyser
Areski wrote: Dear ALL, As everybody seems to like very much Asterisk-Stat, I decided to make couples of improvements... so here we go with a new version :D FEATURES : - CDR report (monthly or daily) - monthly traffic reports (pie graph) - DAILY LOAD !!! - compare call load with previous days - many criterias to define the report - export CDR report to PDF - export CDR report to CSV - support MYSQL POSTGRESQL - etc... Better to check out the screenshot: http://areski.net/asterisk-stat-v2/about.php Waiting for your feedbacks! Enjoy and have a good weekend, Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belad Arezqui Web: http://areski.net/ Email: areski ($alt) gmail ($dot) 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 it appears the image created by graph_hourlydetail.php cannot be displayed because it has errors it works fine if there are no calls for the hour chosen. also, you might consider modifying the querys to include multiple categories, for example a good query would be one that displays the calls made to a particular destination and calls received from a particular source, not necessarily the same number. (example, I call an 800 number to report a problem and open a tickey, all calls returned to me are from a totally different number). The graph thingy, I have no idea why it contains errors regards greg ___ 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] wctdm and two tdm cards
Matt Gibson wrote:
Still not working -
I did notice something kinda weird tho, After adding
{ 0xe159, 0x0001, 0xa900, PCI_ANY_ID, 0, 0, (unsigned long) wctdmh },
to wctdm.c, and rebooting
when I issue lspci -v, the PCI id on the card has changed (?). Is this
a normal thing to happen?
Instead of being 0xa900 it's now seeing as a9fd:0003
I havent changed anything cept rebuilt the zaptel source.
Matt
ps: none of the methods mentioned have worked so far.
You might want to send this off to Digium
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Re: [Asterisk-Users] wctdm and two tdm cards
Matt Gibson wrote: I tried this, but I think this message is slightly outdated, as In my wctdm.c (not wcfxs.c) I have the following, which leads me to believe that it should be already incorporated. Yeah the file name has changed, but the concept is still valid. Is there some way to send a command to the card on reboot to signify that it has lost power and should come back up or something? I'm not familiar at all with low level driver programming or anything, but just a thought. Not that I am aware regards ___ 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] Warning Message with voicemail CVS 3-3-05
I just updated our asterisk zaptel libpri to the cvs 3-3-05 8:07am and now after leaving a voicemail we are getting the following in our logs : Mar 3 10:50:25 WARNING[4408]: Can't change device '**Unknown**' with no technology! Mar 3 10:50:25 WARNING[4409]: Can't change device '**Unknown**' with no technology! Mar 3 10:50:25 WARNING[4410]: Can't change device '**Unknown**' with no technology! No other changes to configuration were made for the update any ideas? Regards Greg ___ 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] kernel error with Zaptel cards
Christopher wrote: It's the server's status light that's flashing, and the lcd display also reads an error as well (PCI parity error). Plus, I prefer fixing a hole in the floor rather than covering it with newspaper :). Just give it a decorative edge and label it waste disposal. ;) ___ 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] wctdm and two tdm cards
Matt Gibson wrote: Greetings, I have a server I'm working on here with two tdm cards in it. 4 FXS and 4FX0. Both cards work fine on their own. The problem lies with using both in the system at once. I have verified the IRQ's are fine. I have tried switching the slots the cards reside in, no luck though. I am using ACPI but not APM. I am using gentoo latest, with vanilla 2.6(.10) kernel and udev. CVS as of CVS-HEAD-03/02/05-03:42:41. The problem is as follows: If I power up the system from system off, the cards both get detected If I reboot the system with reset button, ctrl alt del, or 'reboot' the TDM04P does not get detected. If I then reboot, then hit the power button, and let it turn off, then turn it back on again and boot, it detects both cards fine. I have tried searchign the list archives, but I have not had much luck. One person on IRC mentioned he's seen this before, but didn't have any solutions. Does anyone here know what might be the problem? or have a fix/work around? I know I shouldnt be rebooting servers, but I have to make sure it works upon reboot as it is going to be installed in a power-outtage happy part of the world :) TIA, Matts Sounds like you would be fine if the power completely drops out. but there are 2 things I can think of. In zconfig.h (in the zaptel directory) there is a line at the bottom (at least in my cvs version) /* * Uncomment if you happen have an early TDM400P Rev H which * sometimes forgets its PCI ID to have wcfxs match essentially all * subvendor ID's */ #define TDM_REVH_MATCHALL The other is a bit more involved and can be found here: http://lists.digium.com/pipermail/asterisk-users/2004-October/07.html This might be a bit different with the newer software, but the basic idea is the same Regards Greg ___ 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
