[asterisk-users] Removing Parsing /etc/asterisk/manager.conf from CLI
Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the == Parsing '/etc/asterisk/manager.conf': Found. (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) There is a very old feature request about this at http://bugs.digium.com/view.php?id=3085 but I cannot see the resolution. Mantis shows APPLICATION ERROR #801 at the end of the page... Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing Parsing /etc/asterisk/manager.conf from CLI
Adrian A wrote: Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager (...) Maybe a better solution is to rethink your architecture. The Manager API is well suited for long running connections, so there is no need to reconnect every few seconds. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] WWW: http://www.reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ -- 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] Festival in Italian does not work
I installed festival following this guide (method 1) http://www.voip-info.org/wiki/view/Asterisk+festival+installation When I use english voices Festival works, when I change in Italian voices, Festival return an error (generic). Someone has faced and solved the same problem? Thanks in advance for your help. Bye A.Santoro ___ -- 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] Time out to disconnec: IAX trunk, SIP Trunk, Zaptel Channel
Hi All; I would like to know if any one can helps or have idea on how to disconnect the call in case no media (all parties hanged up, but the channel still open) for the following cases: 1) If we have IAX trunk, then how to disconnect the call automatically after specific time (time out media) if all parties closed and no more media packets is moving between the endpoints? 2) If we have SIP trunk, and same above senaio? 3) In case the call came for the Zaptel channel (FXO) and the source disconnect the call and the disconnect signal was not detected as the telecom service provider does not send it and asterisk did not detect it, so: how to hangup zaptel automatically after a specific timeout as no more media is going through? Any help? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[asterisk-users] question about queue
HI all, I have set up a queue with 2 members (A B). 1st call is waiting in the queue and a queue member A is ringing but don't take the call. Member A keeps ringing. Then 2nd call is also get into the queue but I found that queue member B doesn't ring. That's mean member B is available to take call but somehow it can't get a call in the queue. After that, member A takes the call of the 1st caller and member B gets ring. Question: -Why the 1st call will be stick the queue even there are many call behind? -Is it a bug of the queue or just a setting of the queue to solve the problem? 5000 has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:1, A:1, SL:100.0% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Unknown) has taken 1 calls (last was 76 secs ago) Callers: 1. SIP/2003-02cf0940 (wait: 0:47, prio: 0) 2. SIP/10.100.0.109-e4096dc0 (wait: 0:15, prio: 0) ___ -- 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] Removing Parsing /etc/asterisk/manager.conf from CLI
On 10/04/2008, Stefan Reuter [EMAIL PROTECTED] wrote: Adrian A wrote: Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager (...) Maybe a better solution is to rethink your architecture. The Manager API is well suited for long running connections, so there is no need to reconnect every few seconds. Or use astmanproxy, which will open a persistent * Manager connection and allow you to make occasional polls to the proxy. Regards, Steve ___ -- 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] Grandstream BLF and Call-limit
On Fri, 28 Mar 2008 12:09:58 -0500, Peder @ NetworkOblivion wrote: Any idea? If I remove call-limit on the sip.conf entries, it all goes back to working fine. I tried 2, 9 and 99 on the call-limit and they all have the same issues. I can't imagine why call-limit causes hints to stop updating correctly. on the phone being monitored (sipA in this case), do an 'sip debug peer' and see the different notify messages sent to sipA. this would provide an indication, at the very least to developers if something in asterisk is broken. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- 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] Removing Parsing /etc/asterisk/manager.conf from CLI
On Wed, Apr 09, 2008 at 11:55:13PM -0700, Adrian A wrote: Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the == Parsing '/etc/asterisk/manager.conf': Found. (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) What verbosity level do you use? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
[asterisk-users] followme than voicemail
Hello, I've configured call Call forwarding using followme.conf. It works fine. If no number pick up a call, I want to send the call to the voicemail. How can I do this? followme.conf [150] music=default context=hledej number=12,10 number=13,10 number=14,10 extension.conf [from-internal] exten = 44,n,FollowMe(150|as) [hledej] exten = 12,1,Dial(SIP/12) exten = 13,1,Dial(SIP/13) exten = 14,1,Dial(SIP/14) ___ -- 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] A Simple Question
Hello, I want to get voice broadcasting system which have DTMF tone recognition. I have no time to learn and istall it. Is there anybody to sell me that system ? Thanks, Esref ALBAYRAK ___ -- 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] question about queue
Rilawich Ango Thursday, April 10, 2008 3:28 AM I have set up a queue with 2 members (A B). 1st call is waiting in the queue and a queue member A is ringing but don't take the call. Member A keeps ringing. Then 2nd call is also get into the queue but I found that queue member B doesn't ring. That's mean member B is available to take call but somehow it can't get a call in the queue. After that, member A takes the call of the 1st caller and member B gets ring. What version of Asterisk are you using? This is a know issue/feature with version 1.2.x In version 1.4.x, set autofill=yes in queues.conf and calls will fill in as expected. Don Pobanz ___ -- 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] [1.4.17] Disabled modules - Zaptel stops picking up calls
Hello I have a couple of questions about running 1.4.17 on FreeBSD 6.3: 1 .On a FreeBSD host, In modules.conf, I naively removed the following modules that I thought I didn't need, but after stopping/restarting Asterisk, Zaptel stops reporting calls: /usr/local/etc/asterisk/modules.conf noload = pbx_ael.so noload = res_smdi.so noload = chan_iax2.so = I don't use those features, so why would removing them affect Zaptel? 2. /usr/local/etc/rc.d/asterisk doesn't make use of the watchguard safe_asterisk, and just launches the asterisk binary directly. Why? Does someone have a modified copy of the script that uses safe_asterisk instead? Thank 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] [1.4.17] Disabled modules - Zaptel stops picking up calls
On Thu, Apr 10, 2008 at 03:09:18PM +0200, Vincent wrote: Hello I have a couple of questions about running 1.4.17 on FreeBSD 6.3: 1 .On a FreeBSD host, In modules.conf, I naively removed the following modules that I thought I didn't need, but after stopping/restarting Asterisk, Zaptel stops reporting calls: /usr/local/etc/asterisk/modules.conf noload = pbx_ael.so noload = res_smdi.so noload = chan_iax2.so = I don't use those features, so why would removing them affect Zaptel? 2. /usr/local/etc/rc.d/asterisk doesn't make use of the watchguard safe_asterisk, and just launches the asterisk binary directly. Why? Does someone have a modified copy of the script that uses safe_asterisk instead? chan_zap.so failed to load as it depends on res_smdi.so ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] [1.4.17] Disabled modules - Zaptel stops picking up calls
On Thu, 10 Apr 2008 16:14:01 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote: chan_zap.so failed to load as it depends on res_smdi.so ? I have no idea. Is there an up-to-date list somewhere, or some script that lists dependencies for each module, so that we have some way of knowing what can be safely disabled? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting dtmf mode for a particular peer
On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote: Have you tried the using the SIPDtmfMode function in your dial plan? Not sure how I would introduce that with my enum macro, but as a test I did try it for this particular peer: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/1011002206-b7232f10, 18668398145|1) in new stack -- Goto (internal-sip,18668398145,1) -- Executing [EMAIL PROTECTED]:1] SIPDtmfMode(SIP/1011002206-b7232f10, info) in new stack -- Executing [EMAIL PROTECTED]:2] Macro(SIP/1011002206-b7232f10, ringingdial|SIP/[EMAIL PROTECTED]) in new stack -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/1011002206-b7232f10, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/1011002206-b7232f10, SIP/[EMAIL PROTECTED]) in new stack But still, with sip set debug peer voipmich I see the initial SIP packets establishing the session but no SIP packets when I press buttons on my phone. It can be used to change the DTMF mode between two points in a call. Yeah. I had noticed it before but was not sure how I would introduce it given that selection of the given SIP server was more or less random given that it's an ENUM destination. The problem, I would think, would be if your phones are set up to ONLY send inband audio then you have to find someway to get audio to transcode the DTMF from inband to info. Oh, damnit. I thought for sure this phone I was using was configured for rfc2833 or at least info but it seems I have it set for inband. Is there any way to determine what methods a given SIP phone supports? I'm not familiar enough with the specifics of Asterisk's behavior to know whether that just works or if it needs some special setup. Try putting SipDtmfMode(info) just before the dial command and see what happens. Yeah, did that as above, but no joy. But that could be due to my sipphone-Asterisk connection. Does anyone know if Asterisk will convert an inband DTMF from one sip channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP channel? b. signature.asc Description: This is a digitally signed message part ___ -- 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] Phantom Rings
I'm having a major problem at one of my branch offices with Phantom Rings on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and have made several calls to Verizon with no solution to the problem. I know the telco switch that is feeding my analog lines is an old switch and can't even do CallerID with 2 lines in a rollover configuration. Audio quality on the line is perfect during voice calls. No static or other noise. I've asked for disconnect supervision to be added to the line, but It doesn't look like it's there. The line still seems to keep the channel open long after the far end hangs up. Has anyone else ever seen this problem or have any ideas how to eliminate it? Thanks, Brent Davidson ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went to background and connecting using -r, and colors disappeared for me as well. I'm using screen as well (ls -l --color=auto works fine in screen too). Is there a documented fix available, or is this more just an odd curiosity regarding termtypes? --J -Original Message- Sent: Wednesday, April 09, 2008 10:16 PM Tzafrir Cohen wrote: On Wed, Apr 09, 2008 at 08:00:38PM -0400, Mike wrote: Ah, not bad. When I start asterisk with /usr/sbin/asterisk -c I get the colors, but if I start it without -c and then connect to the console using /usr/sbin/asterisk -r I get no color. Since I want this to be running in the background, how do I fix this so I get to have my cake and eat it too? The patch is rather trivial. Just make Asterisk pretend that it is vt100 (or whatever) if it is running as a service. I cant get color using asterisk -r on 1.2.17 or 18 either. ___ -- 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] Phantom Rings
Quoting Brent Davidson [EMAIL PROTECTED]: I'm having a major problem at one of my branch offices with Phantom Rings on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and have made several calls to Verizon with no solution to the problem. I know the telco switch that is feeding my analog lines is an old switch and can't even do CallerID with 2 lines in a rollover configuration. Audio quality on the line is perfect during voice calls. No static or other noise. I've asked for disconnect supervision to be added to the line, but It doesn't look like it's there. The line still seems to keep the channel open long after the far end hangs up. Has anyone else ever seen this problem or have any ideas how to eliminate it? I had the phantom rings for years, once a day same time roughly every day, finally just got annoyed enough one day I trapped the telco on the phone with me till I finally got to talk to the right person. The right person knew instantly what I was talking about after months of previous denials. On DMS switches what you need to insist be added to your customer line profile is something called NLT or no line test. The wrong person can even look it up if you tell them the name of it - imagine that eh ? Thanks, Brent Davidson ___ -- 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.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] setting dtmf mode for a particular peer
Brian J. Murrell wrote: On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote: Does anyone know if Asterisk will convert an inband DTMF from one sip channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP channel? You might also try canreinvite=no for both your phone and the sip peer. I think it's normal procedure for Asterisk to drop out of the call path once the call is established between two peers. The canreinvite directive forces asterisk to remain as an intermediary, and it will probably do the transcoding that way. If I'm not mistaken this is also useful for making calls between two system that have no common codecs. -Brent ___ -- 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] multiple simultaneous access to single voice mail box
On Wednesday 09 April 2008 17:18:32 Bob Pierce wrote:
We are using Asterisk 1.2.18 at this site. One of the users brought this
to my attention today.
We have a problem when we take the message off the voice mail. If I am
taking off the messages it used to be [on the old phone system] that no
one else was able to go in take off the message. Now I can be taking
off the messages some one else can also be taking off the same
messages. We should not be able to do this!!
Has anyone else seen this? Is there a way to setup the voice mail so
that each box can only be accessed by one person at a time?
There's not a really good way of doing that, not in ways that can't be
subverted. I imagine that you have one extension for getting into voicemail,
no matter which mailbox you retrieve from.
If you instead use a separate extension, you can use groups to restrict the
number of people accessing a particular mailbox:
exten = 123,1,Set(GROUP()=123)
exten = 123,n,GotoIf($[${GROUP_COUNT()} 1]?busy)
exten = 123,n,VoicemailMain([EMAIL PROTECTED])
exten = 123,n,Hangup
exten = 123,n(busy),Busy
--
Tilghman
___
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Re: [asterisk-users] Phantom Rings
Jon Pounder wrote: I had the phantom rings for years, once a day same time roughly every day, finally just got annoyed enough one day I trapped the telco on the phone with me till I finally got to talk to the right person. The right person knew instantly what I was talking about after months of previous denials. On DMS switches what you need to insist be added to your customer line profile is something called NLT or no line test. The wrong person can even look it up if you tell them the name of it - imagine that eh ? Unfortunately the tech I spoke to said the switch we're connected to is so old it has no built-in test capabilities. To run any sort of line tests the line has to be disconnected from the switch and connected to an external test set. I guess that's one of the things you deal with when the boss decides to serve only rural markets. It would be hard to find locations any more rural than where our branches are. At present I have ztmonitor streaming all line activity to a file. I've got plenty of hard drive space so I can record all day if I need to. According to the manager of the branch in question, there were at least 50 phantom calls yesterday, but there has only been 1 this morning. The other curiosity here is that reviewing my asterisk logs all of the phantom calls are on 1 line and swapping ports, the calls follow the line. It's easy to spot the phantom calls in the logs because they always mention dropped frames (probably because of the dialtone coming from the Analog line card). Thanks, Brent ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
On Thu, Apr 10, 2008 at 11:43:18AM -0400, Joshua Kinard wrote: Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went to background and connecting using -r, and colors disappeared for me as well. I'm using screen as well (ls -l --color=auto works fine in screen too). Is there a documented fix available, or is this more just an odd curiosity regarding termtypes? Yes. THe patch is pretty simple: http://bugs.digium.com/9048 Basically Asterisk explicitly disables colors support when you try to open a remote terminal. In that patch I explicitly set the terminal type. This is a very silly hack - when you use a remote console you could not care less about the local console. Sadly Asterisk assumes that your remote console uses exactly the same settings (and escape sequences) as the local one. The fact that this has not generated many bug reports only serves to show how much compatible (to ansi) are different terminals these days. At least for the basics. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Joshua Kinard wrote: Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went to background and connecting using -r, and colors disappeared for me as well. I'm using screen as well (ls -l --color=auto works fine in screen too). The colors work if you use the supplied init scripts. cd /path/to/asterisk/source/contrib/init.d Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Phantom Rings
Quoting Brent Davidson [EMAIL PROTECTED]: Jon Pounder wrote: I had the phantom rings for years, once a day same time roughly every day, finally just got annoyed enough one day I trapped the telco on the phone with me till I finally got to talk to the right person. The right person knew instantly what I was talking about after months of previous denials. On DMS switches what you need to insist be added to your customer line profile is something called NLT or no line test. The wrong person can even look it up if you tell them the name of it - imagine that eh ? Unfortunately the tech I spoke to said the switch we're connected to is so old it has no built-in test capabilities. To run any sort of line tests the line has to be disconnected from the switch and connected to an external test set. I guess that's one of the things you deal with when the boss decides to serve only rural markets. It would be hard to find locations any more rural than where our branches are. At present I have ztmonitor streaming all line activity to a file. I've got plenty of hard drive space so I can record all day if I need to. According to the manager of the branch in question, there were at least 50 phantom calls yesterday, but there has only been 1 this morning. The other curiosity here is that reviewing my asterisk logs all of the phantom calls are on 1 line and swapping ports, the calls follow the line. It's easy to spot the phantom calls in the logs because they always mention dropped frames (probably because of the dialtone coming from the Analog line card). is that one line in the same cable bundle all the way back to the CO ? Could be picking up interference or something, or be half connected to some other line somewhere or some other weirdness. ask for a TDR reading of cable feet on a working and non-working line and they should be damn close to identical. Could also be a bridge tap picking up some stray signal somehow - have that checked for and disconnected. I have also had issues before where an analog modem just plain would not work as the management cct on a telco supplied T1, they changed every piece of wire all the way back to the CO and finally decided it was a bad switch port, swapped the line to a new port and voila worked - tech figured the port was just flaky from lightning damage or something, so some other poor schlub will get a line on there at some point since it tests out ok but doesn't work. I would like to see them go to this effort if it was my own line with my modem - never would have happened. initially all the fingers were pointed at bad building wiring, but that was ruled out in the first 5min, until eventually all that was left was the switch port. Thanks, Brent ___ -- 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Hmm, interesting, initially it wasn't working. Maybe I started it from outside of screen? Odd. BTW, is it possible for the SuSE script to support a variable to pass args to the daemon? Like perhaps modifying ASTARGS to be a changable param at the top of the script? I like the verbose output, and attempting to add it there myself (and change Line 71 to recognize its existence) didn't pan out right. Right now, I have to manually send core set verbose 999 when connecting in. Thanks!, --J -Original Message- Sent: Thursday, April 10, 2008 12:10 PM Joshua Kinard wrote: Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went to background and connecting using -r, and colors disappeared for me as well. I'm using screen as well (ls -l --color=auto works fine in screen too). The colors work if you use the supplied init scripts. cd /path/to/asterisk/source/contrib/init.d Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] best way for call detail logging
Hi, I would like to be able to log call details in Asterisk. The kind of logs that I like to generate is like this: From To Forward Time Incoming Call604-343-3334 503-233-4454 13:33:32 Extension Routing 503-233-4454 Extension 403 13:33:32 Forwarding 503-233-4454 454-444-2334 13:33:32 where 503-233-4454 is my DID number. Basically, I would like to log how calls are being handled in Asterisk. I understand I can use AGI to log the information in database, but I am wondering if this is scalable enough for large number of users. I am using realtime CDR but it does not record the kind of detail that I am looking for. If I don't use AGI, what would be the best way to do it? Can someone please give me some advice or inputs? Thank you very much in advance for your suggestion. Thanks, Pete ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Joshua Kinard wrote: send core set verbose 999 when connecting in. I'm running Mandriva and found the line that had -vvv. I like mine at 15, so just put 15 v's on that line. Worked great. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
On Thu, Apr 10, 2008 at 12:48:05PM -0400, Joshua Kinard wrote: Hmm, interesting, initially it wasn't working. Maybe I started it from outside of screen? Odd. Or maybe it's plain buggy. Bug reports are welcomed. BTW, is it possible for the SuSE script to support a variable to pass args to the daemon? Like perhaps modifying ASTARGS to be a changable param at the top of the script? I like the verbose output, and attempting to add it there myself (and change Line 71 to recognize its existence) didn't pan out right. Right now, I have to manually send core set verbose 999 when connecting in. init.d script source extra parameters from /etc/sysconfig/scriptname (/etc/default/scriptname on Debian). Set this variable in that /etc/sysconfig/asterisk (to -Fvv) BTW: why do you prefer it to start verbosely? This tends to clutter the logs with useless information. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
[asterisk-users] Voicemail: afternoon audio file is missing
Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with envelope=yes and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full: File digits/afternoon does not exist in any format [Apr 10 14:09:08] WARNING[12955]: file.c:866 ast_streamfile: Unable to open digits/afternoon (format 0x2 (gsm)): No such file or directory [Apr 10 14:09:08] WARNING[12955]: say.c:409 wait_file: Unable to play message digits/afternoon I use language=es and I use the AsteriskSounds_ES.tar.gz (unpacked in /var/lib/asterisk/sound/es) package in order to get Spanish audio files. What can I do to correct the afternoon file error ??? Special thanks Alejandro ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
On Thu, Apr 10, 2008 at 01:08:21PM -0400, Doug Lytle wrote: Joshua Kinard wrote: send core set verbose 999 when connecting in. I'm running Mandriva and found the line that had -vvv. I like mine at 15, so just put 15 v's on that line. Worked great. 15? What do you need that for? IIRC the highest verbosity level is 5. anything more than that doesn't change the clogging of your logs. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] setting dtmf mode for a particular peer
On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote: You might also try canreinvite=no for both your phone and the sip peer. Yeah, there is definitely no re-inviting going on. Both Asterisk and the local handset are in a local network behind NAT with reference to the SIP server that requires INFO. I think it's normal procedure for Asterisk to drop out of the call path once the call is established between two peers. The canreinvite directive forces asterisk to remain as an intermediary, and it will probably do the transcoding that way. Indeed, this is my understanding as well, but I am definitely not getting a bridging of the sipphone and sip provider through a re-invite. The NAT would not facilitate it. If I'm not mistaken this is also useful for making calls between two system that have no common codecs. Right. I need to use ekiga or one of the ATAs here that I know support rfc2833 so that I can eliminate this possible need to transcode inband to info/rfc2833 in order to narrow down the field. b. signature.asc Description: This is a digitally signed message part ___ -- 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] TE121, echo issues, NMIs
(The only issue we've had with the TE121 is echo on voice calls, even with the hardware echo cancelling module and lots of zapata.conf tuning... did You EVER get the echo resolved ? How ? We managed to get it tuned to the point where user complaints are minimal, but there is definitely still a problem. We've also had Asterisk randomly die a few times. Nothing is written to the logs in these cases. Digium investigated the echo issue (over SSH) and claimed that my system is handling several non-maskable interrupts and that I should pull out the TE121 and watch /proc/interrupts. I can't really pull a production system down for that kind of invasive testing, so we're just living with it at the moment. If anyone knows of a particular server in which the TE121 is known to work *reliably*, I'd be much obliged. We thought that a SuperMicro server would be a decent choice, but apparently that's not the case. -- Kevin DeGraaf ___ -- 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] best way for call detail logging
On 10:00, Thu 10 Apr 08, Pete Kay wrote: Hi, I would like to be able to log call details in Asterisk. The kind of logs that I like to generate is like this: From To Forward Time Incoming Call604-343-3334 503-233-4454 13:33:32 Extension Routing 503-233-4454 Extension 403 13:33:32 Forwarding 503-233-4454 454-444-2334 13:33:32 where 503-233-4454 is my DID number. Basically, I would like to log how calls are being handled in Asterisk. I understand I can use AGI to log the information in database, but I am wondering if this is scalable enough for large number of users. I am using realtime CDR but it does not record the kind of detail that I am looking for. If I don't use AGI, what would be the best way to do it? Can someone please give me some advice or inputs? Thank you very much in advance for your suggestion. Thanks, Pete Maybe write something that connects to the AMI and listens to what happens there. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Tzafrir Cohen wrote: On Thu, Apr 10, 2008 at 01:08:21PM -0400, Doug Lytle wrote: Joshua Kinard wrote: send core set verbose 999 when connecting in. I'm running Mandriva and found the line that had -vvv. I like mine at 15, so just put 15 v's on that line. Worked great. 15? What do you need that for? IIRC the highest verbosity level is 5. anything more than that doesn't change the clogging of your logs. This one goes to 11 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
-Original Message- Sent: Thursday, April 10, 2008 1:08 PM Or maybe it's plain buggy. Bug reports are welcomed. Nah, I think it was PEBKAC and PICNIC here :: sheepish grin :: I tried adding --style options to the DAEMON var assignment, and it looks like the -f check further down didn't like that. I'll look into the sysconfig setting...more used to Gentoo setups than SuSE. BTW: why do you prefer it to start verbosely? This tends to clutter the logs with useless information. Oh, this is just for a faxing system. We have an ancient Rolm in place for the actual phone calls. I want to monitor/log things when I roll it out for testing to catch any oddities that may occur with inbound and outbound faxes. Communication between it and the Rolm sometimes went to sleep (I sent a mail here on that, but got no responses), and so, I want to watch for it in case it happens again. Might've been a bug fixed in the newer releases of zaptel and asterisk (which I'm running now). --J ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
-Original Message- Sent: Thursday, April 10, 2008 1:22 PM 15? What do you need that for? IIRC the highest verbosity level is 5. anything more than that doesn't change the clogging of your logs. Ah, 5 is max? Kinda like gcc not supporting anything greater than -O3? Good to know that. I figured - was overkill, but I hadn't dived into the options parsing code to actually verify that. --J ___ -- 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] Voicemail: afternoon audio file is missing
Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with envelope=yes and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full: File digits/afternoon does not exist in any format [Apr 10 14:09:08] WARNING[12955]: file.c:866 ast_streamfile: Unable to open digits/afternoon (format 0x2 (gsm)): No such file or directory [Apr 10 14:09:08] WARNING[12955]: say.c:409 wait_file: Unable to play message digits/afternoon I use language=es and I use the AsteriskSounds_ES.tar.gz (unpacked in /var/lib/asterisk/sound/es) package in order to get Spanish audio files. What can I do to correct the afternoon file error ??? That's odd, my afternoon is not in the 'digits' folder. Maybe yours isn't either; try moving afternoon.* into the digits folder...? [EMAIL PROTECTED] ~]$ locate afternoon /var/lib/asterisk/sounds/afternoon.ulaw /var/lib/asterisk/sounds/afternoon.wav /var/lib/asterisk/sounds/afternoon.ul Moj -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
On Thu, Apr 10, 2008 at 01:37:34PM -0400, Joshua Kinard wrote: -Original Message- Sent: Thursday, April 10, 2008 1:22 PM 15? What do you need that for? IIRC the highest verbosity level is 5. anything more than that doesn't change the clogging of your logs. Ah, 5 is max? Kinda like gcc not supporting anything greater than -O3? Good to know that. I figured - was overkill, but I hadn't dived into the options parsing code to actually verify that. The option parsing code will happily add it. It is just that extra levels have no real effect. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
[asterisk-users] problems in REFER request to different machine
Hi everyone, I'm currently trying to enable call transfer to different domains in asterisk box (Asterisk 1.2.13 running on Debian etch). I have a configuration that requires me to transfer call to separate domains like [EMAIL PROTECTED]:5050. My calls come from a R2 channels in a board installed in the machine. When the call comes in I dial a sip address in another machine and I need to receive REFER from this other machine to transfer the call to a third sip URI, that may be or not in any of the two machines . My machines change all the time, so registering them in my asterisk box is not an option. The big picture here is this: I have a asterisk box to receive calls from PSTN and I send this calls to sip application that I made that will transfer the call from the user to different sip application depending on user input. And this other application also need the ability to transfer calls to different sip URI. The applications are conferences, voice mail and others, each running on a different sip uri ([EMAIL PROTECTED]:port) and the user needs to jump between them. So I need my asterisk box to accept arbitrary sip URI in a REFER (xfer) command. Right now it always tries to send the call to a local extension, for example, if I have a call from my asterisk to [EMAIL PROTECTED]:5060 and this application asks asterisk to transfer this call to [EMAIL PROTECTED]:5070 asterisk will try to send the to the local extension 666. Bellow I have a sip debug from the messages. My asterisk box is running in the IP 201.73.67.5, and my first application (the one that asterisk dials directly) is at the address 201.73.67.7:5080 and it transfers the calls to 201.73.67.7:5070, but it fails. All help is very much welcome. Thanks in advance, Thiago Sip debug: -- SIP read from 201.73.67.7:5080: REFER sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 201.73.67.7:5080;rport;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA To: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 Contact: sip:201.73.67.7:5080 Call-ID: [EMAIL PROTECTED] CSeq: 15651 REFER Event: refer Expires: 300 Accept: message/sipfrag;version=2.0 Allow-Events: presence, refer Refer-To: sip:[EMAIL PROTECTED]:5070 Referred-By: sip:[EMAIL PROTECTED] Content-Length: 0 --- (15 headers 0 lines) --- Transfer to 5070 in from-sip-external Transfer from 0778 in from-sip-external Transmitting (no NAT) to 201.73.67.7:5080: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 201.73.67.7:5080;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx;received=201.73.67.7;rport=5080 From: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA To: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 Call-ID: [EMAIL PROTECTED] CSeq: 15651 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing sip:201.73.67.7:5080 for address/port to send to set_destination: set destination to 201.73.67.7, port 5080 Reliably Transmitting (no NAT) to 201.73.67.7:5080: NOTIFY sip:201.73.67.7:5080 SIP/2.0 Via: SIP/2.0/UDP 201.73.67.5:5060;branch=z9hG4bK26db8c59;rport From: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 To: sip:[EMAIL PROTECTED]:5080;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=15651 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing sip:201.73.67.7:5080 for address/port to send to set_destination: set destination to 201.73.67.7, port 5080 Reliably Transmitting (no NAT) to 201.73.67.7:5080: BYE sip:201.73.67.7:5080 SIP/2.0 Via: SIP/2.0/UDP 201.73.67.5:5060;branch=z9hG4bK1e66e326;rport From: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 To: sip:[EMAIL PROTECTED]:5080;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA Call-ID: [EMAIL PROTECTED] CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- -- SIP read from 201.73.67.7:5080: SIP/2.0 200 OK Via: SIP/2.0/UDP 201.73.67.5:5060;rport=5060;received=201.73.67.5;branch=z9hG4bK26db8c59 Call-ID: [EMAIL PROTECTED] From: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 To: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA CSeq: 103 NOTIFY Contact: sip:201.73.67.7:5080 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Length: 0 --- (10 headers 0 lines) --- -- SIP read from 201.73.67.7:5080: SIP/2.0 200 OK Via: SIP/2.0/UDP 201.73.67.5:5060;rport=5060;received=201.73.67.5;branch=z9hG4bK1e66e326 Call-ID: [EMAIL PROTECTED] From: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 To: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA CSeq:
Re: [asterisk-users] Voicemail: afternoon audio file is missing
On Thursday 10 April 2008 12:14:49 Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with envelope=yes and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full: File digits/afternoon does not exist in any format [Apr 10 14:09:08] WARNING[12955]: file.c:866 ast_streamfile: Unable to open digits/afternoon (format 0x2 (gsm)): No such file or directory [Apr 10 14:09:08] WARNING[12955]: say.c:409 wait_file: Unable to play message digits/afternoon I use language=es and I use the AsteriskSounds_ES.tar.gz (unpacked in /var/lib/asterisk/sound/es) package in order to get Spanish audio files. What can I do to correct the afternoon file error ??? Sounds like you're using a custom set of recordings. I just checked the Asterisk standard distribution and digits/afternoon is exactly where it should be. Check with the person from whom you obtained the tarball. -- Tilghman ___ -- 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] multiple simultaneous access to single voice mail box
On Thu, 2008-04-10 at 11:25 -0500, Tilghman Lesher wrote: If you instead use a separate extension, you can use groups to restrict the number of people accessing a particular mailbox: Thanks Tilghman, I didn't think of that. I'm sure that will work just fine for what we need. Have a great day. Bob ___ -- 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] setting dtmf mode for a particular peer
Brian J. Murrell wrote: On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote: You might also try canreinvite=no for both your phone and the sip peer. Yeah, there is definitely no re-inviting going on. Both Asterisk and the local handset are in a local network behind NAT with reference to the SIP server that requires INFO. I think it's normal procedure for Asterisk to drop out of the call path once the call is established between two peers. The canreinvite directive forces asterisk to remain as an intermediary, and it will probably do the transcoding that way. Indeed, this is my understanding as well, but I am definitely not getting a bridging of the sipphone and sip provider through a re-invite. The NAT would not facilitate it. If I'm not mistaken this is also useful for making calls between two system that have no common codecs. Right. I need to use ekiga or one of the ATAs here that I know support rfc2833 so that I can eliminate this possible need to transcode inband to info/rfc2833 in order to narrow down the field. b. One more tidbit I just ran across in the upgrade.txt file, since you mention NAT: In 1.4, you need to set canreinvite=nonat to disable re-invites when NAT=yes. This is propably what you want. The settings are now: yes, no, nonat, update. Please consult sip.conf.sample for detailed information. Let us all know how the tests go with the other phone. -Brent ___ -- 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] Queue member state 'Not in use
I have an operator queue that is supposed to ring 2 phones, extension 10 and 11. Everything is working correctly but I keep seeing these messages in my log: The device state of this queue member, Sip/10, is still 'Not in Use'. Everything I've been able to find on this message so far points to the realtime system but I'm not using realtime. I've looked through the upgrade.txt file and the only thing I see that might be of interest is the call-limit option. I don't use that option since I am using Snom phones with MWI that requires at least call-limit = 2. Should I just not worry about these messages or is there an easy fix? Thanks, Brent ___ -- 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] Is Asterisk really good??
So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and making your own cool voice mail stuff. But before I delve into it, I thought a question to the community would be in order. 2 more questions. 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? 2 - What would it take to set one up? cards / computer power / pricing on software? What has your experience been? Thank you for taking time to look at this post! Eugen ___ -- 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] Is Asterisk really good??
Hi Eugene, Yes it's that good. All the functionality you posted is possible. Regarding your international calls, nothing more is required than two asterisk servers (one at each location) and a broadband connection - cards are only used to connect into pstn or isdn. Why don't you start with a preformatted iso like trixbox or druid and then jump into coding your own once you know what you are doing. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eugen Soare Sent: Thursday, 10 April 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is Asterisk really good?? So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and making your own cool voice mail stuff. But before I delve into it, I thought a question to the community would be in order. 2 more questions. 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? 2 - What would it take to set one up? cards / computer power / pricing on software? What has your experience been? Thank you for taking time to look at this post! Eugen ___ -- 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] Is Asterisk really good??
On Thu, 10 Apr 2008, Eugen Soare wrote: 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? No. There is no free lunch. It takes electricity, bandwidth, and depending on who you want to call in Germany, termination. Thanks in advance, Steve Edwards [EMAIL PROTECTED] 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
Re: [asterisk-users] setting dtmf mode for a particular peer
On Thu, 2008-04-10 at 13:36 -0500, Brent Davidson wrote: One more tidbit I just ran across in the upgrade.txt file, since you mention NAT: In 1.4, you need to set canreinvite=nonat to disable re-invites when NAT=yes. This is propably what you want. The settings are now: yes, no, nonat, update. Please consult sip.conf.sample for detailed information. REINVITEs are a total red herring to this problem. Please trust me on that. Let us all know how the tests go with the other phone. OK. Using an ATA and setting DTMF to INFO on it and the same for it's sip.conf entry has yielded some interesting results. I can confirm by using sip set debug peer ata that when I press digits, SIP INFO messages are being sent to Asterisk. Further, when on a call to voipmich when I press digits I can see on the Asterisk console: * DTMF-relay event received: digit when I press a digit. Good so far. However, even with both: exten = 18661234567,n,SIPDtmfMode(info) exten = 18661234567,n,Macro(ringingdial,SIP/[EMAIL PROTECTED]) In the dialplan for a given number and with the entry in sip.conf for voipmich: [voipmich] type=peer fromuser=nobody fromdomain=nodomain host=tf.voipmich.com dtmfmode=info And sip set debug peer voipmich enabled, I still don't see SIP INFO messages when I press digits on my phone. I do see the above DTMF-relay event received just not the corresponding SIP INFO messages outgoing to voipmich. So it seems that Asterisk is still not properly relaying the inbound DTMF into outbound DTMF. Ideas? b. signature.asc Description: This is a digitally signed message part ___ -- 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] Removing Parsing /etc/asterisk/manager.conf from CLI
Anything more than 'core set verbose 1' produces this message, however verbose 1 does not display much of anything. On Thu, Apr 10, 2008 at 1:53 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Apr 09, 2008 at 11:55:13PM -0700, Adrian A wrote: Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the == Parsing '/etc/asterisk/manager.conf': Found. (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) What verbosity level do you use? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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 ___ -- 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] Is Asterisk really good??
On Thursday 10 April 2008 02:14:17 pm Steve Edwards wrote: 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? No. There is no free lunch. It takes electricity, bandwidth, and depending on who you want to call in Germany, termination. though you may want to look into DUNDi and http://asterisk.li/peeringgraf.htm there seems to be a fairly strong DUNDi network in Europe -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Removing Parsing /etc/asterisk/manager.conf from CLI
There is an OpenSER proxy in front of Asterisk which handles the clients. The script is called by OpenSER whenever a client sends a SUBSCRIBE request for MWI. It uses php to connect to Asterisk like so: fsockopen($mhost,5038, $errno, $errstr, 5) and gets the user's voicemail counts. I'm not sure how I would maintain this as a persistent connection that would live if I restart Asterisk. I'd have to detect that somehow. Adrian On Thu, Apr 10, 2008 at 12:14 AM, Stefan Reuter [EMAIL PROTECTED] wrote: Adrian A wrote: Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager (...) Maybe a better solution is to rethink your architecture. The Manager API is well suited for long running connections, so there is no need to reconnect every few seconds. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] WWW: http://www.reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 ___ -- 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] Is Asterisk really good??
Steve Edwards wrote: On Thu, 10 Apr 2008, Eugen Soare wrote: 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? No. There is no free lunch. It takes electricity, bandwidth, and depending on who you want to call in Germany, termination. Yep. And if the transport is primarily the public Internet, be very aware of how call quality can be impacted over that kind of distance / that many routing hops, depending on what path you ride to Germany. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Is Asterisk really good??
BTW Eugene, was just reading Tom Keatings blog (he's a pretty well known reporter around here), he was just talking about a new commercial appliance called CogoBlue that you could send to the person at the other end if they don't know anything about voip or pabx's to make it even easier for them to setup calls at their end. http://blog.tmcnet.com/blog/tom-keating/asterisk/ispbx-launches-asterisk -appliances-with-cogoblue-asterisk-gui.asp Cheers, Dean -Original Message- From: Dean Collins Sent: Thursday, 10 April 2008 2:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Is Asterisk really good?? Hi Eugene, Yes it's that good. All the functionality you posted is possible. Regarding your international calls, nothing more is required than two asterisk servers (one at each location) and a broadband connection - cards are only used to connect into pstn or isdn. Why don't you start with a preformatted iso like trixbox or druid and then jump into coding your own once you know what you are doing. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eugen Soare Sent: Thursday, 10 April 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is Asterisk really good?? So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and making your own cool voice mail stuff. But before I delve into it, I thought a question to the community would be in order. 2 more questions. 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? 2 - What would it take to set one up? cards / computer power / pricing on software? What has your experience been? Thank you for taking time to look at this post! Eugen ___ -- 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] Is Asterisk really good??
Remember - True TELCO grade systems simply cannot be compared to anything else. You want the reliability , uptime, and all the bells and whistles of true Carrier Grade Hardware/Software then you pay for it. If you want something you can tinker under the hood with and do some cool stuff with, Asterisk is it. BUT - by it need NORTEL PBX reliability or the power and flexibility of a big Rockwell ACD, get out your check book. The two are not the same. Eugen Soare wrote: So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing ___ -- 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 member state 'Not in use
Brent Davidson wrote: I have an operator queue that is supposed to ring 2 phones, extension 10 and 11. Everything is working correctly but I keep seeing these messages in my log: The device state of this queue member, Sip/10, is still 'Not in Use'. Everything I've been able to find on this message so far points to the realtime system but I'm not using realtime. I've looked through the upgrade.txt file and the only thing I see that might be of interest is the call-limit option. I don't use that option since I am using Snom phones with MWI that requires at least call-limit = 2. Should I just not worry about these messages or is there an easy fix? Thanks, Brent The call-limit in sip.conf is the setting that needs to be set in order for the members to appear as in use when they answer calls. You need to be sure that you set the call-limit for each peer and not just set the call-limit in the general section. Mark Michelson ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Tzafrir Cohen wrote: On Thu, Apr 10, 2008 at 01:08:21PM -0400, Doug Lytle wrote: Joshua Kinard wrote: send core set verbose 999 when connecting in. I'm running Mandriva and found the line that had -vvv. I like mine at 15, so just put 15 v's on that line. Worked great. That I did not know. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Is Asterisk really good??
Asterisk as a PBX is fantastic. It offers the features found in the most sophisticated traditional Until now the third party PBX configuration software has not had the sophistication of Asterisk product itself. Check out cogoblue.com It is a visual drag and drop configuration tool that is powerful yest simple to use. Eugen Soare wrote: So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and making your own cool voice mail stuff. But before I delve into it, I thought a question to the community would be in order. 2 more questions. 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? 2 - What would it take to set one up? cards / computer power / pricing on software? What has your experience been? Thank you for taking time to look at this post! Eugen ___ -- 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] DTMF between Asterisk servers.
Just a thought. A while back there was discussion about the merits of having a product (in that case an O/S) with contracted vendor support or relying solely on list support. I note in the post below where one responder states It may also have been because less than 23 hours had elapsed Different strokes for different folks But 23 Hours is lng time in the production world with no help on a TELCO problem. Just an observation on how differently folks see things and what folk need to recognize before they dump their NORTEL etc and jump into open-source. Mark Hamilton wrote: No, I tried calling the inbound DID to see if DTMF passes through. And most times it does, however, it's not being relayed to the Asterisk server 2, and then to the direct external phoneline. I tried changing all dtmfmodes for the sip peer, for the inbound DID provider, and it didn't work, even tried playing with canreinvite, etc. Hence why my desperate plea for help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: April 8, 2008 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF between Asterisk servers. I believe that what you described should just work with the caveat that dtmf=inband is rarely the right thing to do over SIP, and is prone to all sorts of DTMF detection and debounce issues. I assume you've tried calling a POTS endpoint and listening to see if you get DTMF passed through? 1) You did not give a great deal of information about what the current situation was, or what investigations you've already tried, which is probably why no-one felt they could reply. 2) It may also have been because less than 23 hours had elapsed... Regards, Steve On 08/04/2008, Mark Hamilton [EMAIL PROTECTED] wrote: I find it hard to believe no one knows, so is it just plain no helping? J If someone would like to atleast point me in the right direction that will deal specifically with what I'm asking, that would be appreciated too. Much thanks. From: Mark Hamilton [mailto:[EMAIL PROTECTED] Sent: April 7, 2008 11:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: DTMF between Asterisk servers. Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/[EMAIL PROTECTED],,t T,) Where 12351 accepts the call on Asterisk 2, and in some cases, that call is transferred out to a PSTN number, or wherever, but not within Asterisk anymore via provider2, dtmf=rfc2833. When the call comes in, I'd like it to relay DTMF just dandy. How can I do so? There is no NAT between the Asterisk servers or in front of them. However, Asterisk2 has iptables which allows all UDP traffic to/fro Asterisk1. When Asterisk2 transfers the call to external endpoints, there might be a LAN, but relative ports are open on those LANs. Please help. Thanks in advance, Mark. ___ -- 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 ___ -- 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] Digium T1 Card Crashing Server (Dell 2950)
I installed the Digium T1 card on Dell Poweredge 2950 and the system crashed several times, we got a Kernel Panic and first though it was the OS so I switched from Fedora 7 to Centos 5.1. Our server was alarming in our monitoring system, when our Infrastructure department investigated the issue they found that the server was locked up at the console. They had to do a hard reboot of the server to bring it back. The following information was pulled from the logs: BUG: soft lockup detected on CPU#1! [c044b2a4] softlockup_tick+0x96/0xa4 [c042e214] update_process_times+0x39/0x5c [c04196ff] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 . Apr 10 11:07:39 Newark1 php: /var/lib/asterisk/agi-bin/libs_a2billing/Class.A2Billing.php[259]: Error parsing /etc/asterisk//a2billing.conf on line 48 Apr 10 11:07:39 Newark1 php: /var/lib/asterisk/agi-bin/libs_a2billing/Class.A2Billing.php[588]: Undefined index: asterisk_version Apr 10 11:07:39 Newark1 php: /var/lib/asterisk/agi-bin/a2billing.php[150]: Undefined index: intro_prompt Apr 10 11:39:59 Newark1 syslogd 1.4.1: restart. Apr 10 11:39:59 Newark1 kernel: klogd 1.4.1, log source = /proc/kmsg started. I believe the issue may be related to a driver for Our Digium Interface. Here is the versions we are running Asterisk 1.4.19 libpri-1.4.3 zaptel-1.4.10 Digium's TE220 PCI Express card with echo cancellation. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On Thursday 10 April 2008 15:43:33 John Signorello wrote: Asterisk as a PBX is fantastic. It offers the features found in the most sophisticated traditional Until now the third party PBX configuration software has not had the sophistication of Asterisk product itself. Check out cogoblue.com It is a visual drag and drop configuration tool that is powerful yest simple to use. Common ethics would require you to post a disclaimer that you are behind cogoblue, not a satisfied independent customer. And again, you are posting advertising on a non-commercial list. -- Tilghman ___ -- 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] Digium T1 Card Crashing Server (Dell 2950)
Any time you have this kind of hard lockup with a Digium card you should run, not walk to the nearest phone and call them. broadband Voice wrote: I installed the Digium T1 card on Dell Poweredge 2950 and the system crashed several times, we got a Kernel Panic and first though it was the OS so I switched from Fedora 7 to Centos 5.1. Our server was alarming in our monitoring system, when our Infrastructure department investigated the issue they found that the server was locked up at the console. They had to do a hard reboot of the server to bring it back. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Is Asterisk really good??
On Thu, Apr 10, 2008 at 04:05:47PM -0400, Al Baker wrote: Remember - True TELCO grade systems simply cannot be compared to anything else. You want the reliability , uptime, and all the bells and whistles of true Carrier Grade Hardware/Software then you pay for it. If you want something you can tinker under the hood with and do some cool stuff with, Asterisk is it. BUT - by it need NORTEL PBX reliability or the power and flexibility of a big Rockwell ACD, get out your check book. The two are not the same. Sure. But even commercial PBX/Hybrid stuff is often more robust than Asterisk-on-a-PC (I don't have enough anecdotes on appliances, though I expect they'd be better). Now, yes, carrier-grade equipment can be as complex and still 4-nines reliable (or 5-nines :-), but we're now talking NEBS, -48VDC, 23-inch racks, and 6-digit-plus price tags. And for what it's worth, at my new job I babysit about a 16-machine cluster running VICIdial for close to 200 agents, and by and large, it just runs. It's got about 20 T-1s feeding it. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Phantom Rings
Brent Davidson wrote: I'm having a major problem at one of my branch offices with Phantom Rings on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and have made several calls to Verizon with no solution to the problem. I know the telco switch that is feeding my analog lines is an old switch and can't even do CallerID with 2 lines in a rollover configuration. Audio quality on the line is perfect during voice calls. No static or other noise. I've asked for disconnect supervision to be added to the line, but It doesn't look like it's there. The line still seems to keep the channel open long after the far end hangs up. Has anyone else ever seen this problem or have any ideas how to eliminate it? Brent, I had a similar problem and I feel for you, its frustrating. Are you using polycom phones by chance? Here is the problem that I had, not sure if your problem is related. Specs: - 6 Polycom 301 phones. - CentOS 4 Server with Asterisk 1.2.x - Sangoma A200 card with 3 FXO ports. Pretty simple settings for a small office where a group ring between all 6 polycoms was initiated once the call was received. After that it would go to a auto attendant and give the caller option to continue to hold, leave a message, etc. At any rate, once in a while, Caller ID would fail, either on the Sangoma card or somewhere in Asterisk, not sure, but CLI clearly showed an error on caller ID every once in a while when a POTS call came in. All six polycoms would ring, but when you picked up the handset or hit the Answer soft button, nothing would happen, you couldn't answer the call. The phones would just ring, and ring and ring for the duration of the group ring (about 60) and the customer was really annoyed since it was a small office. Continuing, the problem finally turned out to be the polycoms! When no caller ID information was present, the polycoms wigged out and while they did ring, you could not get the phones to pick up. I could readily replicate the behavior by initiating a Call File without specifying the caller ID information using the local channel. It would happen every time. Specifying the CID would allow the polycoms to work correctly. On the customer side, I did a quick GoToIf in their dialplan to see if the caller id info was set and if it wasn't I would set it manually to something like: CALLERID(num)=555-555- CALLERID(name)=CID FAILURE That cleared up the problem. HIH -- Warm Regards, 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] Is Asterisk really good??
On Thu, 2008-04-10 at 16:05 -0400, Al Baker wrote: Remember - True TELCO grade systems simply cannot be compared to anything else. You want the reliability , uptime, and all the bells and whistles of true Carrier Grade Hardware/Software then you pay for it. If you want something you can tinker under the hood with and do some cool stuff with, Asterisk is it. BUT - by it need NORTEL PBX reliability or the power and flexibility of a big Rockwell ACD, get out your check book. The two are not the same. Eugen Soare wrote: So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing I agree, though be carefull not trying to compare apples and pears.. Think i can provide some extra perspective, as i worked over 18 years for a telco manufacturer... Reliability: With original telco equipment you have hardly any choice to make with respect to reliability i mean. The differ is size and hence the costs. You don't hace voip-pbx with a mtbf of A or B. With Asterisk its your choice. You not only may choose, not you have to choose! Is it around=the-corner COTS-quality? Commercial-grade? MIL-specs? fail-over? double? quad? A large farm? Reliability has it price... Functionality Hate to say it, but do you realy know what you/your company/your customer or clients want and-or-need? What is your budget? I've been in meetings with customers for more than a year to get on paper what they realy wanted. teams of engineers can code what ever you want. At a price. Do you want to choose from existing configurations? Then your choise will be limited to what previous customers or sleepless engineers may have imagined. Ofcourse, It may lack some functionaliteis, or more likely it has some functionalities you don't want. it can be modified surely You decide: Do you want to change existing equipment, do you want to change your requirements, or do you have no objections against new equipment having an impact your department.. With asterisk, its the same as with reliability. It's your call. You can make it exactly how you want it, nothing less, nothing more. But: A) know exactly what to want for now and the near future. B) don't under estimate the time and costs Customization Same as with functionality. With existing equipment, accepts what a supplier gives you or be prepared to digg deep in your wallet. Asterisk: 100%, full stop. management summary: 1) What ever you decide to, get your requirements on paper. 2) Determine if you can find people to do the job 3) Launch a pilot-project: get to know asterisk 4) re-examine your requirements 5) evaluate your pilot-project 6) check your budget 7) either decide or back to 3) or 4) It's doesn't matter if its for you own home, small company or a large firm ___ -- 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] Digium T1 Card Crashing Server (Dell 2950)
BUG: soft lockup detected on CPU#1! [c044b2a4] softlockup_tick+0x96/0xa4 [c042e214] update_process_times+0x39/0x5c [c04196ff] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 . You don't happen to be running a XEN Kernel are you? I saw this problem while running CentOS 5.1 XEN kernel and if you search their bug tracking system you will see some reports about this bug. A search on google revealed some possible solutions. This was the first thought that came to my mind when I saw this. Regards, Michael L. Young (elguero) ___ -- 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] Is Asterisk really good??
Quote And for what it's worth, at my new job I babysit about a 16-machine cluster running VICIdial for close to 200 agents, and by and large, it just runs. It's got about 20 T-1s feeding it Please share more about this. What/How are you clustering the boxes ? Is this all VOIP or TDMF front and VOIP for agents in back ? What kind of Boxes ? What O/S What tools are you using to monitor this big-azz mother ? Thx in in advance for sharing !! Jay R. Ashworth wrote: On Thu, Apr 10, 2008 at 04:05:47PM -0400, Al Baker wrote: Remember - True TELCO grade systems simply cannot be compared to anything else. You want the reliability , uptime, and all the bells and whistles of true Carrier Grade Hardware/Software then you pay for it. If you want something you can tinker under the hood with and do some cool stuff with, Asterisk is it. BUT - by it need NORTEL PBX reliability or the power and flexibility of a big Rockwell ACD, get out your check book. The two are not the same. Sure. But even commercial PBX/Hybrid stuff is often more robust than Asterisk-on-a-PC (I don't have enough anecdotes on appliances, though I expect they'd be better). Now, yes, carrier-grade equipment can be as complex and still 4-nines reliable (or 5-nines :-), but we're now talking NEBS, -48VDC, 23-inch racks, and 6-digit-plus price tags. And for what it's worth, at my new job I babysit about a 16-machine cluster running VICIdial for close to 200 agents, and by and large, it just runs. It's got about 20 T-1s feeding it. Cheers, -- jra ___ -- 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] Digium T1 Card Crashing Server (Dell 2950)
We're using PAE Kernel. On Thu, Apr 10, 2008 at 4:30 PM, Michael L. Young [EMAIL PROTECTED] wrote: BUG: soft lockup detected on CPU#1! [c044b2a4] softlockup_tick+0x96/0xa4 [c042e214] update_process_times+0x39/0x5c [c04196ff] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 . You don't happen to be running a XEN Kernel are you? I saw this problem while running CentOS 5.1 XEN kernel and if you search their bug tracking system you will see some reports about this bug. A search on google revealed some possible solutions. This was the first thought that came to my mind when I saw this. Regards, Michael L. Young (elguero) ___ -- 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] Excellent Paper on ECHO in VOIP Environment
http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.html I thought others would find this helpful. I have no tie to any vendor or company referenced in this paper and I believe the same concepts to be applicable to the Asterisk world. ___ -- 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] Phantom Rings
Lee Jenkins wrote: Brent, I had a similar problem and I feel for you, its frustrating. Are you using polycom phones by chance? Here is the problem that I had, not sure if your problem is related. Specs: - 6 Polycom 301 phones. - CentOS 4 Server with Asterisk 1.2.x - Sangoma A200 card with 3 FXO ports. Pretty simple settings for a small office where a group ring between all 6 polycoms was initiated once the call was received. After that it would go to a auto attendant and give the caller option to continue to hold, leave a message, etc. At any rate, once in a while, Caller ID would fail, either on the Sangoma card or somewhere in Asterisk, not sure, but CLI clearly showed an error on caller ID every once in a while when a POTS call came in. All six polycoms would ring, but when you picked up the handset or hit the Answer soft button, nothing would happen, you couldn't answer the call. The phones would just ring, and ring and ring for the duration of the group ring (about 60) and the customer was really annoyed since it was a small office. Continuing, the problem finally turned out to be the polycoms! When no caller ID information was present, the polycoms wigged out and while they did ring, you could not get the phones to pick up. I could readily replicate the behavior by initiating a Call File without specifying the caller ID information using the local channel. It would happen every time. Specifying the CID would allow the polycoms to work correctly. On the customer side, I did a quick GoToIf in their dialplan to see if the caller id info was set and if it wasn't I would set it manually to something like: CALLERID(num)=555-555- CALLERID(name)=CID FAILURE That cleared up the problem. HIH -- Warm Regards, Lee That really doesn't surprise me with Polycoms. They have excellent voice quality but their interface is dismal. I'm using Snom phones and I'm fairly certain the problem is with the phone line. I have all callerID settings disabled as the Telco is unable to provide it along with our rollover line setup due to limitations in their antiquated switch. The CLI and Logs all plainly show the calls as if they were normal calls with the exception of a message about Failed to write frame and no DTMF attempts, then the call is routed into the operator queue. The calls always came in on Zap1-1 so I tried swapping the 2 lines to see if it stayed on port 1 or if the phantom followed the line. As expected, the phantom rings followed the line and began showing up on Zap2-1. So it pretty has to be something in the telco, but I'm not sure what. Putting WaitForRing(3) before the Answer command in my IVR menu eliminates most of them, but sometimes more of them slip through. -Brent ___ -- 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] Removing Parsing /etc/asterisk/manager.conf from CLI
On Thu, Apr 10, 2008 at 12:21:46PM -0700, Adrian A wrote: Anything more than 'core set verbose 1' produces this message, however verbose 1 does not display much of anything. So obviously there are two ways to resolve thins (apart from leaving things as they are now) - demoting the priority of this message, or promoting the priority of those important messages you miss. That message is often useful for me, if I have #include-s . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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
[asterisk-users] Any body tried MysqlPool-1.4
I saw an add-on for Asterisk at http://www.yosd.at/index.php?option=com_contenttask=viewid=23Itemid=38 Anyone tried any of these ? How well did they work ?? Asterisk Applications Print http://www.yosd.at/index2.php?option=com_contenttask=viewid=23pop=1page=0Itemid=38 The Asterisk PBX core system offers you great flexibility because of its internal design. It does come with support for a varity of codecs, channels and applications. But sometimes there is the need for a special application which is not included in the basic system... Here you can find some very usefull asterisk addons, which we have developed in the past time. Download them - use them. If you need support setting up one of the applications, or if you need a special version (with custom modifications) then contact us under This email address is being protected from spam bots, you need Javascript enabled to view it . We do also offer creating complete new applications for the asterisk pbx system. * MysqlPool MysqlPool is not an asterisk application. MysqlPool is an asterisk ressource which is needed for the following applications to work correctly. The MysqlPool ressource will handle multiple connections to one or more Mysql servers using sockets or tcp/ip. It does register some CLI commands which makes it possible to get the state of the connections, and to change to number of connections for one host. The ressource does offer public functions for the other applications to get a valid Mysql connection handle without locking problems. * LCDial LCDial is a failsafe least cost routing engine. * DBQuery With DBQuery you can access you Mysql Database and query it for the needed data, or update data in your Mysql Database. It is also possible to register DBQuery as cdr backend - so you can define your own sql statement for writing cdr data. * DBRewrite DBRewrite offers you regular expression matching and substitution. It does support multiple keys, and for each key you can define as many regular expessions and substitution rules as you want. ___ -- 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] Digium T1 Card Crashing Server (Dell 2950)
Hello, It might not be Digium's fault, I ran into similar problems with Dell 2950 servers and other PCIexpress cards. I even went so far as to have several components replaced by Dell on one of the affected servers to no avail. After many months of banging my head against a wall I stumbled across the following posts on the Trixbox forums: http://www.trixbox.org/forums/trixbox-forums/open-discussion/acpi-default-install-2-4-0 http://www.trixbox.org/forums/trixbox-forums/open-discussion/tb-2-4-crashing-asus-amd-and-new-dell-server-spec After talking to some computer engineers at a few companies I learned that It seems Dell does not have very good quality control on the power control chipsets that they use and so on some machines you have to disable acpi(or enable it) at the kernel level. If you do not set it correctly, when the power saving functions trigger there is a higher likelyhood that an error will occur leading to a kernel panic. This is most likely the same problem so take a look at the forum postings and try disabling/enabling acpi in your grub startup. Of course it could be something else entirely, but this problem does seem to be common with Dell 2950, and this did fix the problem for me on more than one Dell 2950. MATT--- On 4/10/08, broadband Voice [EMAIL PROTECTED] wrote: We're using PAE Kernel. On Thu, Apr 10, 2008 at 4:30 PM, Michael L. Young [EMAIL PROTECTED] wrote: BUG: soft lockup detected on CPU#1! [c044b2a4] softlockup_tick+0x96/0xa4 [c042e214] update_process_times+0x39/0x5c [c04196ff] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 . You don't happen to be running a XEN Kernel are you? I saw this problem while running CentOS 5.1 XEN kernel and if you search their bug tracking system you will see some reports about this bug. A search on google revealed some possible solutions. This was the first thought that came to my mind when I saw this. Regards, Michael L. Young (elguero) ___ -- 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 ___ -- 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] question about queue
Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes On Thu, Apr 10, 2008 at 8:57 PM, Don Pobanz [EMAIL PROTECTED] wrote: Rilawich Ango Thursday, April 10, 2008 3:28 AM I have set up a queue with 2 members (A B). 1st call is waiting in the queue and a queue member A is ringing but don't take the call. Member A keeps ringing. Then 2nd call is also get into the queue but I found that queue member B doesn't ring. That's mean member B is available to take call but somehow it can't get a call in the queue. After that, member A takes the call of the 1st caller and member B gets ring. What version of Asterisk are you using? This is a know issue/feature with version 1.2.x In version 1.4.x, set autofill=yes in queues.conf and calls will fill in as expected. Don Pobanz ___ -- 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] odd error compiling zaptel-1.4.10
CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o LD [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/wcte12xp.o /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/zconfig.h:91:41: error: missing binary operator before token ( CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/card_fxo.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/card_fxs.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/card_pri.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/xbus-core.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/xpp/xbus-sysfs.o Havent seen that before... Any ideas. I am running centos 5.1 amd x86_64. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On Thu, Apr 10, 2008 at 05:49:26PM -0400, Al Baker wrote: Please share more about this. What/How are you clustering the boxes ? Is this all VOIP or TDMF front and VOIP for agents in back ? What kind of Boxes ? What O/S What tools are you using to monitor this big-azz mother ? What, Matt? You haven't already talked about this here? :-) My new job is Matt Florell's old job, where VICIdial got started. I haven't counted the boxes lately, but I think there are 14 with quad-T cards in them, separate boxes for MySQL and Apache. Our architecture is FXS T-1 channel banks for the agent phones, usually 1 + 3 IXC spans per box, though we turned up a box a couple weeks ago with 3 channel banks, and no spans. All TDM; the only VoIP is the IAX trunks hauling load-balance calls around. And just the usual VICIdial tools, mostly, though I'm fixin to deploy either Big Sister or Nagios. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] question about queue
Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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
[asterisk-users] Newbie Polycom: Will a big 0000-directory.xml crash the phone?
I am exploring the contacts directory in Polycom and I am wondering if a big -directory.xml on the boot server will eat up the memory and crash the Polycom phone once downloaded onto the phone. The asterisk directory extension is good but because users cannot see the names I thought to find another alternative. Does anyone have any experience? ___ -- 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] Running asterisk + T1 + ztdummy on Debian vserver
Hi I am wondering if anyone has experince running multiple instances of Asterisk using Debian vserver. The senario I want to implement is to have a couples of DIDs. Some DIDs are handled by Asterisk instance #1 and some DIDs are handled by Asterisk instance #2. The two Asterisk are within different Debian vservers. The calls are coming from a T1 line. Has anyone tried this out and maybe give me some guidelines on how that can be done? Specific questions I have are: 1. How do the two Asterisk instances connect to the T1 line via (1 or more?)ztdummy? 2. How to distribute incoming DID calls from T1 based on rules? Any suggestion will be greatly appreciated. Regards, Mark ___ -- 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] TDM400P Dialtone problem
Hi Guys, I have a TDM400P (2FXS and 2FXO) installed on my asterisk (1.4.18.1) Zaptel (1.4.9.2) running on Fedora Core 7, below are my configurations and some diagnosis I did. The problem is when I connect an analogue phone on either of the FXS channels I don't get a dialtone, I can't call any of the sip clients or even call my echo test, number, which is an context included in that of the FXS port. How can I solve this problem? Please assist. After doing a /sbin/ztcfg -vv, after listing the zaptel channels configuration it says 4 channels to configure instead of 4 channels configured. is there any additional configuration or dependency I need to load (I've done modprobe wctdm and modprobe zaptel before doing /sbin/ztcfg -vv). zaptel.conf # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) fxoks=1 fxoks=2 fxsks=3 fxsks=4 # Global data loadzone = us defaultzone = us zapata.conf [trunkgroups] ; define any trunk groups [channels] ; hardware channels ;language=en ;context=from-zaptel ;signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; usedistinctiveringdetection=yes ; default usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 echotraining=yes rxgain=0.0 txgain=0.0 ;group=0 callgroup=1 pickupgroup=1 immediate=no context=phone signalling=fxo_ks channel = 1 ;callerid= ;mailbox= ;group= context=phone ;;; line=2 WCTDM/0/1 FXOKS (In use) signalling=fxo_ks ;context=phone channel = 2 ;callerid= ;mailbox= ;group= context=incoming ;;; line=3 WCTDM/0/2 FXSKS (In use) signalling=fxs_ks ;callerid=asreceived ;group=0 channel = 3 context=incoming ;;; line=4 WCTDM/0/3 FXSKS (In use) signalling=fxs_ks ;callerid=asreceived ;group=0 ;context=incoming channel = 4 ;context=default [EMAIL PROTECTED] /]# /sbin/ztcfg -vv Zaptel Version: 1.4.9.2 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. Asterisk CLI core score show channeltypes Type Description Devicestate Indications Transfer -- --- --- --- Feature Feature Proxy Channel Driver no yes no IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes Local Local Proxy Channel Driver yes yes no SIP Session Initiation Protocol (SIP) yes yes yes Phone Standard Linux Telephony API Driver no yes no MGCP Media Gateway Control Protocol (MGCP) yes yes no Agent Call Agent Proxy Channel yes yes no -- 7 channel drivers registered. *CLI module reload chan_zap.so No such module 'chan_zap.so' *CLI -- Murithi Martin NOTICE: The contents of this e-mail and any accompanying documentation is confidential and any use thereof, in whatever form, by anyone other than the addressee for whom it is intended is strictly prohibited. ___ -- 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 logging
Hi, I'm not looking for a programma that show the queue logging. But is there a way to check during a call, which member is connected to the caller. Kind Regard, Arjan Kroon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Wolfe Sent: woensdag 9 april 2008 17:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] queue logging You could ASTassistant to see this. Its Freeware. www.astassistant.com - Original Message - From: Arjan Kroon | Mobillion mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com Sent: Wednesday, April 09, 2008 1:01 AM Subject: [asterisk-users] queue logging Hi, I' using with asterisk a queue with tree members and round robin. When a caller enters this queue and it is connecting to one of the members, is there a possibility to see which member the caller is connected to? And is there a way to see in de application to see if the connection from the caller to one of the members was successful of not successful? I know you can see it in de queue. log. But I want to know if I can see it also in the hangup (h) in de application? Kind Regards ___ -- 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] tdm410p w/ echo - no full duplex
hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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 and MACRO
Has anyone gotten Realtime and Macros to work with Asterisk? Or is it better to run the macros directly from the flat conf file? I have the macro working without using ARA but as soon as I use ARA it fails. Any assistance or pointing me in the right direction is appreciated. ___ -- 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] Loosing SIP registration.
Hi All, I am having problems with some SIP peers. I seem to loose registration. If I reload SIP the registration comes back. They usually stay registered for about 2 days before they drop. The problem is not all of them drop usually just the list 2 in the list. The other strange thing is that the 2 the do drop their registration do not occur at the exact same time. It could be many hours between them. I am using Asterisk 1.4.18.1 Any help would be greatly appreciated. My parent's server is having the problems. My server does not exhibit this problem. I just took my router/firewall down to them as I have just purchased a new one and they are still experiencing the problem. sip show registry Host Username Refresh StateReg.Time 202.168.56.133:506061990xx 105 Registered Fri, 11 Apr 2008 15:15:58 sip.pennytel.com:5060 61289xx 105 Request Sent Thu, 10 Apr 2008 21:38:54 sip2.bbpglobal.com:5060 617000xxx 105 Request Sent Thu, 10 Apr 2008 20:43:20 sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip-register.conf': Found == Parsing '/etc/asterisk/sip-klavo.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found sip show registry Host Username Refresh StateReg.Time 202.168.56.133:506061990xx 105 Registered Fri, 11 Apr 2008 15:16:15 sip.pennytel.com:5060 61289xx 105 Registered Fri, 11 Apr 2008 15:16:16 sip2.bbpglobal.com:5060 617000xxx 105 Registered Fri, 11 Apr 2008 15:16:16 ___ -- 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
