Re: [asterisk-users] Zaptel Not Releasing Channel (PRI)
I had this problem with a box that I was using Festival tts on. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Thursday, April 23, 2009 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zaptel Not Releasing Channel (PRI) I have an issue with one of my installations running Asterisk 1.4.20 that I need some help with. Not sure if this is new or not but Zaptel or Libpri is not releasing channels properly. I have had issues with calls that stay up for eight hours, long distance on the telco side, so it is more than just a nuisance. Does anyone know if there was a related bug in the 1.4 branch of Asterisk, Zaptel, Libpri? I searched and did not find anything except references to an issue a few years ago. Any easy pointers before I start to dig more? The box is running with a Digium Quad T1 card. One port is set to normal PRI settings for US to the telco. Another span is connected to an Adit Channel Bank and breakout box to twenty or so phones. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- 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] interesting problem update
The problem I was experiencing is still occurring, and it is getting
worse. There are several names that Festival gets stuck on. I don't
know if it is a Festival problem or an Asterisk problem. The scenario, a
call comes in goes through the dialplan (shown below in original message),
and either reads the value ${FULLNAME}, but then hangs. It doesn't go to
the next command (VoiceMail), and the channel is never released. Other
times, it doesn't even read the name, it just hangs.
exten = _5[14-9]XXX,n,Festival(${FULLNAME})
somehow it gets stuck between the above line and the below line
exten = _5[14-9]XXX,n,VoiceMail(${ext...@students)
Log from a successful call:
[Dec 22 14:24:50] VERBOSE[2715] logger.c: == Parsing
'/etc/asterisk/festival.conf': [Dec 22 14:24:50] VERBOSE[2715] logger.c:
Found
[Dec 22 14:24:50] DEBUG[2715] app_festival.c: Text passed to festival
server : Jxx Lxxx Cxxx (name blocked for confidentially)
[Dec 22 14:24:50] DEBUG[2715] app_festival.c: Passing text to festival...
[Dec 22 14:24:50] DEBUG[2715] app_festival.c: Passing data to channel...
[Dec 22 14:24:50] DEBUG[2715] app_festival.c: Festival WV command
[Dec 22 14:24:52] DEBUG[2715] app_festival.c: Last frame
Log from an unsuccessful call (never get Last frame message):
[Dec 22 13:03:22] VERBOSE[1912] logger.c: == Parsing
'/etc/asterisk/festival.conf': [Dec 22 13:03:22] VERBOSE[1912] logger.c:
Found
[Dec 22 13:03:22] DEBUG[1912] app_festival.c: Text passed to festival
server : Jxxx Wx Kx (name blocked for confidentially)
[Dec 22 13:03:22] DEBUG[1912] app_festival.c: Passing text to festival...
[Dec 22 13:03:22] DEBUG[1912] app_festival.c: Passing data to channel...
[Dec 22 13:03:23] DEBUG[1912] app_festival.c: Festival WV command
Any ideas for troubleshooting?
-Original Message-
From: Eve Ellen Cole [mailto:ec...@mail.plymouth.edu]
Sent: Tuesday, December 16, 2008 6:35 PM
To: asterisk-users@lists.digium.com
Subject: interesting problem
I've got an interesting problem and am wondering if anyone can shed light
.
I am running Asterisk on RHEL Server release 5.2 connecting to an Avaya
Definity G3R via a Digium TE220.
Asterisk 1.4.20
Zaptel 1.4.4
Libpri 1.4.4
MySQL 5.0.45
Festival Speech Synthesis System: 1.95
We have about 4200 accounts in a MySQL db. Asterisk retrieves the user
information from the database, festival tts says the name, then the caller
can leave a voicemail, which is then emailed to the user. At this time,
the system only takes calls, no calls go out.
The problem is that at times Asterisk doesn't release the channel.
Messages in the log file show indicate the channel is busy. The only way
I can find to get the channel to release is to restart Asterisk.
[Dec 16 14:39:06] DEBUG[11141] chan_zap.c: Ring requested on channel 0/1
already in use or previously requested on span 1. Attempting to
renegotiating channel.
Since this is happening on a regular basis, I've been doing some
troubleshooting and can now predictably cause this problem. It mainly
seems to happen with one particular mailbox, and festival seems to be a
factor. When this particular mailbox is dialed, Asterisk goes through the
dialplan up to and including the Festival(${FULLNAME}) step, but not
beyond. Just for yucks, I changed the fullname of the person with that
mailbox by taking out the middle name. All seems to work fine without the
middle name. If I put a middle initial or middle name, the channel locks
up again. I've wondered if Festival has a problem with the length of the
name, but there are other students with longer names and this problem
doesn't occur with their extensions. Any thoughts?
Dialplan
exten = _5[14-9]XXX,1,Answer()
exten = _5[14-9]XXX,n,Playtones(ring)
exten = _5[14-9]XXX,n,MYSQL(Connect CONNID localhost asterisk
HG06e6kghpUjtGvnX asterisk)
exten = _5[14-9]XXX,n,MYSQL(Query RESULTID ${CONNID} Select 'fullname'
from voicemail_users Where mailbox=${EXTEN})
exten = _5[14-9]XXX,n,MYSQL(Fetch FETCHID ${RESULTID} FULLNAME)
exten = _5[14-9]XXX,n,MYSQL(Disconnect ${CONNID})
exten = _5[14-9]XXX,n,GotoIf($[${FETCHID} = 1]?connect:disconn)
exten = _5[14-9]XXX,n(connect),StopPlaytones()
exten = _5[14-9]XXX,n,Wait(2)
exten = _5[14-9]XXX,n,Playback(you-have-dialed)
exten = _5[14-9]XXX,n,Playback(the-mailbox)
exten = _5[14-9]XXX,n,Playback(for)
exten = _5[14-9]XXX,n,Festival(${FULLNAME})
exten = _5[14-9]XXX,n,VoiceMail(${ext...@students)
exten = _5[14-9]XXX,n,Playback(goodbye)
exten = _5[14-9]XXX,n,Hangup() exten =
_5[14-9]XXX,n(disconn),Zapateller()
exten = _5[14-9]XXX,n,Playback(you-dialed-wrong-number)
exten = _5[14-9]XXX,n,Playback(check-number-dial-again)
exten = _5[14-9]XXX,n,Playtones(congestion)
exten = _5[14-9]XXX,n,Wait(3)
exten = _5[14-9]XXX,n,Hangup()
___
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[asterisk-users] interesting problem
I’ve got an interesting problem and am wondering if anyone can shed light …
I am running Asterisk on RHEL Server release 5.2 connecting to an Avaya
Definity G3R via a Digium TE220.
Asterisk 1.4.20
Zaptel 1.4.4
Libpri 1.4.4
MySQL 5.0.45
Festival Speech Synthesis System: 1.95
We have about 4200 accounts in a MySQL db. Asterisk retrieves the user
information from the database, festival tts says the name, then the caller can
leave a voicemail, which is then emailed to the user. At this time, the system
only takes calls, no calls go out.
The problem is that at times Asterisk doesn’t release the channel. Messages in
the log file show indicate the channel is busy. The only way I can find to get
the channel to release is to restart Asterisk.
[Dec 16 14:39:06] DEBUG[11141] chan_zap.c: Ring requested on channel 0/1
already in use or previously requested on span 1. Attempting to renegotiating
channel.
Since this is happening on a regular basis, I’ve been doing some
troubleshooting and can now predictably cause this problem. It mainly seems to
happen with one particular mailbox, and festival seems to be a factor. When
this particular mailbox is dialed, Asterisk goes through the dialplan up to and
including the Festival(${FULLNAME}) step, but not beyond. Just for yucks, I
changed the fullname of the person with that mailbox by taking out the middle
name. All seems to work fine without the middle name. If I put a middle
initial or middle name, the channel locks up again. I’ve wondered if Festival
has a problem with the length of the name, but there are other students with
longer names and this problem doesn’t occur with their extensions. Any
thoughts?
Dialplan
exten = _5[14-9]XXX,1,Answer()
exten = _5[14-9]XXX,n,Playtones(ring)
exten = _5[14-9]XXX,n,MYSQL(Connect CONNID localhost asterisk
HG06e6kghpUjtGvnX asterisk)
exten = _5[14-9]XXX,n,MYSQL(Query RESULTID ${CONNID} Select 'fullname' from
voicemail_users Where mailbox=${EXTEN})
exten = _5[14-9]XXX,n,MYSQL(Fetch FETCHID ${RESULTID} FULLNAME)
exten = _5[14-9]XXX,n,MYSQL(Disconnect ${CONNID})
exten = _5[14-9]XXX,n,GotoIf($[${FETCHID} = 1]?connect:disconn)
exten = _5[14-9]XXX,n(connect),StopPlaytones()
exten = _5[14-9]XXX,n,Wait(2)
exten = _5[14-9]XXX,n,Playback(you-have-dialed)
exten = _5[14-9]XXX,n,Playback(the-mailbox)
exten = _5[14-9]XXX,n,Playback(for)
exten = _5[14-9]XXX,n,Festival(${FULLNAME})
exten = _5[14-9]XXX,n,VoiceMail(${ext...@students)
exten = _5[14-9]XXX,n,Playback(goodbye)
exten = _5[14-9]XXX,n,Hangup() exten = _5[14-9]XXX,n(disconn),Zapateller()
exten = _5[14-9]XXX,n,Playback(you-dialed-wrong-number)
exten = _5[14-9]XXX,n,Playback(check-number-dial-again)
exten = _5[14-9]XXX,n,Playtones(congestion)
exten = _5[14-9]XXX,n,Wait(3)
exten = _5[14-9]XXX,n,Hangup()
___
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Re: [asterisk-users] Trouble with PRI config
Steve, I didn't do a very good copy and paste, that is why Primary d-channel set to 01C14 in the message, although it was actually set to 01C1424. I went back to square one, and stepped through the setup quite methodically. I now have a working PRI between my Avaya Definity G3R v11 and my Asterisk 1.4.20 system. - Eve Ellen The working configuration: Asterisk [Digium TE220B] Zaptel.conf # G3R TN464GP 01C14 span=1,1,0,esf,b8zs # get clock from interface, primary bchan=1-23 dchan=24 loadzone = us defaultzone=us Zapata.conf ; Span 1 G3R TN464GP 01C14 switchtype=national signalling=pri_net ;G3R Connect: pbx, Interface: user context=from_pbx group=1 channel = 1-23 Avaya [TN464GP] ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: pbx Interface: user Protocol version: a Near-end CSU type: other (for the T1 crossover) signaling group 6 Primary d-channel set to 01C1424 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 8:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Primary d-channel set to 01C14. Why doesn't it say 01C1424 then? On Thu, Jun 19, 2008 at 7:48 PM, Eve-Ellen [EMAIL PROTECTED] wrote: The d-channel on the Avaya would be 01C1424. The rest of 01C14 would be the b-channels. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble with PRI config
I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config pri_net usually when connecting to a legacy system. Thanks, Steve T On Thu, Jun 19, 2008 at 1:38 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
The d-channel on the Avaya would be 01C1424. The rest of 01C14 would be the b-channels. - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 19, 2008 6:41:06 PM GMT -05:00 US/Canada Eastern Subject: Re: [asterisk-users] Trouble with PRI config On Thu, Jun 19, 2008 at 4:11 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Thu, 19 Jun 2008, Eve-Ellen Cole wrote: Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. It's not my area of expertise, but I have issues with T1 numbering between vendors -- what they said was 1, 2, 3, 4 turned out to be 4, 3, 2, 1. You might try swapping the T1s and see what happens. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 I have actually done this with a Definity G3 but the cards were not PRI, I had to use EM_W. This doesn't make sense to me though. Primary d-channel set to 01C14 I memory servers me correctly (and it has been a couple years) the 01 means cabinet 1, C is the slot, and 14 is the port number. I would expect it to say 01C24 for the D chan. I could be completely wrong but it is something to try and would explain why your PRI chans are not coming up. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel issue
Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further ___ -- 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] zaptel issue
Interesting, the bottom of my previous email disappeared ... so here is again. Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further. I get past make clean, ./configure, then when I get to make install I get this error. #$ sudo make install make[1]: Entering directory `/var/ports/zaptel-1.4.11' make -C /lib/modules/2.6.18-53.1.19.el5/build ARCH=x86_64 SUBDIRS=/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=pciradio.o tor2.o torisa.o wcfxo.o wct1xxp.o wctdm.o wcte11xp.o wcusb.o zaptel.o ztd-eth.o ztd-loc.o ztdummy.o ztdynamic.o zttranscode.o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/ wcte12xp/ modules make[2]: Entering directory `/usr/src/kernels/2.6.18-53.1.19.el5-x86_64' scripts/Makefile.build:17: /kernel/Makefile: No such file or directory make[3]: *** No rule to make target `/kernel/Makefile'. Stop. make[2]: *** [_module_/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-53.1.19.el5-x86_64' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/var/ports/zaptel-1.4.11' make: *** [all] Error 2 -Original Message- From: Eve-Ellen Cole [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 10, 2008 4:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: zaptel issue Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further. ___ -- 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] Dell 1950
I am thinking of going with a Dell PowerEdge 1950 ||| for a new CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB memory, two 500GB hard drives (presumably mirrored). I also plan to get a Digium TE220B to go with it. (a non-dell server is not an option, but I am wondering if there is a better one to consider) The system will be a voice mail repository for 4-6,000 students. Each time a voice mail is left, an email notification will be sent to the student. The email notification will provide a web link to direct the student to the voice mail itself. Anything I need to consider changing? I'm interested in any feedback you are willing to provide. Many 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] Dell 1950
Steve, You are correct. I've asked this question before. Now that my sysadmin group has determined the server they would prefer to support, I wanted to double check. We won't be relying heavily on the T1 card. I will also be setting up H.323 trunking to our Avaya Definity G3R. Although there will be many accounts on the system, we do not forsee many simultaneous calls, or much use at all to tell the truth. We'll be lucky if just 1% of the users utilize the system. In fact, there is a good chance the system will become a test environment for other unified communications solutions ... that's why I've upped the RAM. From a budget standpoint, its easier to get it now, rather than fight for it later. I'm still toying with whether or not to go RAID 1 or RAID 5, but otherwise I think I'm fairly confident with what we've come up with. Thank you all for the input. Much appreciated. - Eve Ellen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, April 28, 2008 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dell 1950 On Mon, Apr 28, 2008 at 10:16 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Mon, Apr 28, 2008 at 9:21 AM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I am thinking of going with a Dell PowerEdge 1950 ||| for a new CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB memory, two 500GB hard drives (presumably mirrored). I also plan to get a Digium TE220B to go with it. (a non-dell server is not an option, but I am wondering if there is a better one to consider) The system will be a voice mail repository for 4-6,000 students. Each time a voice mail is left, an email notification will be sent to the student. The email notification will provide a web link to direct the student to the voice mail itself. Anything I need to consider changing? I'm interested in any feedback you are willing to provide. Many thanks! I don't use Dells except in pure VoIP setups. I found this link http://www.gomiem.org/conferences/TC08_9_ip_telephony.pdf I think the newer Digium cards play MUCH better with Dell now. Take a look at CSID configuration 1. It seems they use it as their platform so I would take that as a good sign. Thanks, Steve Totaro You asked this identical question a week or two ago. I have an Avaya Definity G3R. Calls to students will be routed through the G3R, to the Asterisk system so the caller can leave a message. I'm not sure how many channels I'll really need, but I expect no more than 23 simultaneous calls. In fact, maybe no more than 10 simultaneously. I guess you bumped up the concurrent call number a bit or are just planning for the future. As far as the RAM disk or OrecX, you should not need it. From personal experience the I/O barrier is ~60-70 calls. Thanks, Steve Totaro ___ -- 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] Dell 1950
Ah, managing user accounts. That is going to be very challenging. I haven't looked at this too thoroughly yet, but I will need to very soon. We have limited resources, so will be looking at a way to automate anything and everything possible. I'm open to suggestions. In my initial test instance, I have installed ARI, FOP and FreePBX. I have run into permission issues that I need to troubleshoot though. I've also experienced problems getting anywhere with http://www.littlejohnconsulting.com/. It seems to be unavailable more than available to date. It also doesn't resolve the mass import issue I will need to get figured out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Monday, April 28, 2008 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dell 1950 The 2 Port card may not provide the number of channels you may need to do this. I would bump it up to a four port. I would also look at more HD space. You are fine on RAM memory, if you need to for budget constraints I would be OK with dropping the RAM and upping the Hard Drive Space. 2-4 GB of ram is PLENTY. What are you interfacing with and how (RBS T1, PRI). How are you planning on managing the user accounts? Are you building a web interface. One simple way although it would break the Directory application is to create All the possible mailbox numbers and then create aliases in your email program to route them. IE VMBox emailreal email 12345 [EMAIL PROTECTED] [EMAIL PROTECTED] Of you could build a Dialplan application that uses LDAP and query the database. I am not sure on your network setup so I can't really give you specifics only hints. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eve-Ellen Cole Sent: Monday, April 28, 2008 9:21 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Dell 1950 I am thinking of going with a Dell PowerEdge 1950 ||| for a new CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB memory, two 500GB hard drives (presumably mirrored). I also plan to get a Digium TE220B to go with it. (a non-dell server is not an option, but I am wondering if there is a better one to consider) The system will be a voice mail repository for 4-6,000 students. Each time a voice mail is left, an email notification will be sent to the student. The email notification will provide a web link to direct the student to the voice mail itself. Anything I need to consider changing? I'm interested in any feedback you are willing to provide. Many 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] capacity
Thank you all for the great advice. Although fairly new to Asterisk, and relearning systems administration, it has helped put some perspective on the matter for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, March 19, 2008 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] capacity I use standard wav (most compatible with players) so about a meg a minute. In my experience, most people (users) use their voicemail similar to email, they keep everything. Especially love struck college kids. I think Asterisk has a soft limit of 1,000 (maybe it is 999) messages as the max per inbox that can be changed in source. I suppose if you limit the max time allowed and the max inbox limit it might help but I think your 60GB estimate would be quite low in the real world. BUT, that is based on when I was in college and I was one of the very few to have my own cell phone (dating myself a bit). So in the real world, I am not sure how much use the system would actually see. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 12:33 PM, Drew Gibson [EMAIL PROTECTED] wrote: Our office averages around 1.5MB / mailbox, call it 10MB for rounding. 6,000 x 10MB = 60GB (n'est pas?) 2 x 250GB drives, mirrored, should cover that and the system quite nicely. regards, Drew Disclaimer: Most of our employees are programmers so probably don't have any friends to call and leave messages! :-) Steve Totaro wrote: RAID arguments (preference really) aside, 4k - 6k worth of student voicemails is going to require quite a bit of storage space. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 12:01 PM, Drew Gibson [EMAIL PROTECTED] wrote: Having ventured high enough and far enough to view the curvature of the Earth and having stayed up late enough long enough (why do disks only fail at the weekend?) to rebuild and restore RAID 5 sets, I proffer the following (not so) Humble Opinion . Dual power supplies, two thumbs up but RAID 5 is only good for reducing storage costs on large volumes of data. It reduces performance and reliability over RAID 1. Don't put the OS on RAID 5 unless you like rebuilding servers from bare metal. It's much easier to rebuild and restore the data on RAID 5 sets if the OS is already up and running. Your OS and other system critical files (Asterisk) should be on RAID 1 for performance, redundancy and cost reasons. More disks = higher cost and higher chance of failure. Asterisk in general does not need much disk storage. The minimum drive size available in a new server tends to be overkill. Two drives as RAID 1 gives you redundancy and performance. Adding a third drive for RAID 5 adds cost, increases complexity and reduces reliability just to add storage capacity that you don't really need. (but the reseller WILL make more money and impress you with their command of the big words and acronyms on the spec sheet.) If and only if you need to store many hundreds of gigs of data (eg. recording a very large volume of calls) then RAID 5 becomes useful (or RAID 10 or RAID n). You should add this bulk storage IN ADDITION TO the mirrored pair holding the OS. regards, Drew Steve Totaro wrote: And I can post a link that shows a bunch of guys think the earth is flat with a 5/10 google ranking also (like the barf guys). http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm I usually just call my guy at CDW and give him my needs, he is a former techie gone sales. He puts together a quote and emails it to me for approval. I find HP server are very robust and rock solid at a decent price point (IBM as well). I like the 380 because you get six hot swap scsi bays and redundant power supplies in a 2u profile, also, Digium and Sangoma T1 cards have never given me an issue. Many on this list love Supermicro, I have yet to try them but I will in the near future. I have not heard a single complaint, only rave reviews. I guess my original point was going for redundancy as far as storage and power supplies with your dollar, not the fastest proc or maxed out RAM that will not be needed. Regardless of the actual hardware or RAID setup, that is the angle I suggest you take. 4k - 6k students will require quite a bit of storage. Thanks, Steve Totaro On Wed, Mar 19, 2008 at 9:38 AM, Ron Joffe [EMAIL PROTECTED] wrote: On Tuesday 18 March 2008 22:12, Steve Totaro wrote: For your use, I would go for a RAID 5 I would highly recommend against a raid 5 set. I can give you more details if you are interested, but these guys have most if it down : www.baarf.com see
[asterisk-users] capacity
Hi, I am planning to deploy an Asterisk system to supply 4-6,000 students with voicemail capabilities. The system will be set up with non-DIDs, route incoming calls to voicemail, then send an email notification. Anyone with some ideas on how I should go about spec'ing the server this use? - Eve Ellen ___ -- 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] capacity
I have an Avaya Definity G3R. Calls to students will be routed through the G3R, to the Asterisk system so the caller can leave a message. I'm not sure how many channels I'll really need, but I expect no more than 23 simultaneous calls. In fact, maybe no more than 10 simultaneously. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, March 18, 2008 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] capacity On Tue, Mar 18, 2008 at 1:55 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: Hi, I am planning to deploy an Asterisk system to supply 4-6,000 students with voicemail capabilities. The system will be set up with non-DIDs, route incoming calls to voicemail, then send an email notification. Anyone with some ideas on how I should go about spec'ing the server this use? - Eve Ellen Strictly VM? How are the calls going to arrive? How many simultaneous accesses, both leaving messages and retrieving (highest peak). I believe Vonage uses Asterisk for their VM (not sure where I heard that). Thanks, Steve Totaro ___ -- 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] user web interface
Hi, I am interested in offering our users a website interface for retrieving their voice mail on asterisk. It seems that there are several possibilities. I am interested in hearing from others regarding the pros and cons of various options. I would like something secure, yet fairly simple and direct for the user. - Eve Ellen ___ -- 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
