[Asterisk-Users] Polycom ip500 dial prob
I have a Polycom IP500 that will not dial out if it is off-hook. From speakerphone it will work, I presume that it has to do with caching of digits, but if the handset is picked up it will drop the dial attempt after the 7th or 8th digit. Has anyone else seen this and if so how do I correct it? Lane Hoskins, MCP Network Engineer (540) 767-7600 main (540) 767-7626 direct image001.gif___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM 200 question
Question for other snom 200 users: 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the handset earpiece which seems to quiet after there is audio on the other end. Ideas? 4. Sound quality to called parties outside our system is intermittently horrible: static filled and raspy where we have to ask people to repeat themselves many times. Could this be related to powerline noise or something like that? We have 8 lines coming into our building. Two are the main lines which we have ringing to the receptionist first and then to selected other extens. This part works great. We need to map the keys on the SNOM 200 such that when there is a call on line 1 the top key flashes/lights steady depending on call state and any extension can pick it up even if it doesn't ring there by pressing the button. This needs to hold true for the 1st two lines, and one of the remaining 6 lines at each extension as we have direct dials. All calls come to * via a T1 Digium card and an Adtran TSU 600. There are 8 separate POTS lines to our building for voice. So in example - call comes in on pstn line 1 , button one flashes at all phones, someone answeres it, button one solid on all phones, call comes in on line 2, button 2 flashes on all phones,can be answered from anywhere by simply hitting that button, gets answered and button changes to solid on all phones, call comes to me from line 8 (my direct dial line) and button 3 flashes on my phone only (my phone will also ring b/c it's set up that way in the dialplan) I can put other caller on hold and answer line 8 simply by pressing the button. Is this an easy thing to do that I'm simply not seeing? Thanks, Lane Hoskins, MCP Network Engineer Automated Horizons Inc. Direct - 540.767.7626 Main - 540.767.7600 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call quality questions
Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200's with various firmware revisions from 2.2t to 2.3o. Any help is appreciated. 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the handset earpiece which seems to quiet after there is audio on the other end. Ideas? 4. Sound quality to called parties outside our system is intermittently horrible: static filled and raspy where we have to ask people to repeat themselves many times. Could this be related to powerline noise or something like that? Lane Hoskins, MCP Network Engineer Automated Horizons Inc. Direct - 540.767.7626 Main - 540.767.7600 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call quality questions
Thanks... -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Fri 1/30/2004 5:16 PM To: '[EMAIL PROTECTED]' Cc: Subject: RE: [Asterisk-Users] Call quality questions Hello, Did you set the flag in the makefile for zaptel for SMP kernels? Yes, we are using the SMP settings 1. I have a couple Snom200 phones on my system running redhat with a P4 HT and haven't had any issues with horrible sound quality using 711ulaw. Have tried everything - alaw, ulaw, gsm - currently using gsm 2. As for the speakerphone cutout, that's to be expected, The snom200s are just half-duplex speakerphones. If you want a good speakerphone get a Polycom. wish I'd known that before I recommended we buy 12 of them :-( :-) 3. If you have silence suppression turned on anywhere I would turn it off. We had the same problem with our Sipura adapters until we turned off the silence supression. You could also mess with the level of echo supression and see if that makes a difference. this I need to look into 4. do you have your T1 set in zaptel.conf to be the primary timing source for your card? yes Hope that helps, MATT--- -Original Message- From: Lane Hoskins [mailto:[EMAIL PROTECTED] Sent: Friday, January 30, 2004 4:22 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call quality questions Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200's with various firmware revisions from 2.2t to 2.3o. Any help is appreciated. 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the handset earpiece which seems to quiet after there is audio on the other end. Ideas? 4. Sound quality to called parties outside our system is intermittently horrible: static filled and raspy where we have to ask people to repeat themselves many times. Could this be related to powerline noise or something like that? Lane Hoskins, MCP Network Engineer Automated Horizons Inc. Direct - 540.767.7626 Main - 540.767.7600 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
[Asterisk-Users] 7960 Problems
This is not specifically related to * but * is the software Im using so here goes Does anyone have the correct file set for a 7960?? Ive been trying to get the release 6 SIP load on one I have without any luck. The phone keeps getting the same 2 files from the tftp server and starting over. If you have the files other than the POS30600.bin which I know is licensed could you please send them to me so I can figure out if its my files or my phone?? I really would appreciate any possible help with this. Thanks, Lane Hoskins, MCP Network Engineer 540.767.7626
RE: [Asterisk-Users] Problem at compiling zaptel
Hello,
I'm also new to * but I think that this is what we had to do:
You need to make sure that the following packages are installed on your
system:
-OpenSSL-Devel
-Ncurses
-Ncurses-Devel (C++)
-sox
-kernel sourses
-kernel development
-bison
-newt
-newt-devel
-readline
-readline-devel
If you already have all of these I know that there were some bugs in the
latest CVS but I'm not sure if that's the problem. I'm also curious
which distribution you're running. If you do have these please do the
following and then try to compile again:
In /usr/src/linux
#make mrproper
#make menuconfig (exit without changing anything)
#make dep
then in /usr/src/zaptel
make clean install
Hope this helps
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: Franz Edler [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 9:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem at compiling zaptel
Hi all!
Can anybody please give me some advice, what is wrong at my first try to
compile Asterisk. I have successfully downloaded the sources from CVS,
but
now the next step at zaptel clean; make install fails.
Please have a look at the error-log below.
There must be a fundamental mis-configuration I suppose, but I am
unfortunately not an expert in this area.
Franz
-- error log -
lpc:/usr/src # cd zaptel
lpc:/usr/src/zaptel # make clean; make install rm -f torisatool makefw
tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f
zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h
rm
-f libtonezone* rm -f tor2ee rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__
-DEXPORT_SYMTAB
-I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes
-fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I
/usr/src/linux/include -I/usr/src/linux/include/net
-DSTANDALONE_ZAPATA -c
zaptel.c
In file included from /usr/include/linux/module.h:20,
from zaptel.c:44:
/usr/include/asm/module.h:54:2: #error unknown processor family In file
included from /usr/include/linux/mm.h:205,
from /usr/include/asm/pci.h:7,
from /usr/include/linux/pci.h:677,
from zaptel.c:46:
/usr/include/linux/page-flags.h:119: error: `CONFIG_X86_L1_CACHE_SHIFT'
undeclared here (not in a function)
/usr/include/linux/page-flags.h:119: error: requested alignment is not a
constant In file included from zaptel.c:48:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the
compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header
files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
recommended by Linus Torvalds)
/usr/include/linux/version.h:8:2: #error These files are glibc internal
and
may not match the
/usr/include/linux/version.h:9:2: #error currently running kernel. They
should only be
/usr/include/linux/version.h:10:2: #error included via other system
header
files - user space
/usr/include/linux/version.h:11:2: #error programs should not directly
include linux/*.h or
/usr/include/linux/version.h:12:2: #error asm/*.h as well.
/usr/include/linux/version.h:13:2: #error
/usr/include/linux/version.h:14:2: #error To build kernel modules
please do
the following:
/usr/include/linux/version.h:15:2: #error
/usr/include/linux/version.h:16:2: #error o Have the kernel sources
installed
/usr/include/linux/version.h:17:2: #error
/usr/include/linux/version.h:18:2: #error o Make sure that the
symbolic
link
/usr/include/linux/version.h:19:2: #error/lib/modules/`uname
-r`/build
exists and points to
/usr/include/linux/version.h:20:2: #errorthe matching kernel source
directory
/usr/include/linux/version.h:21:2: #error
/usr/include/linux/version.h:22:2: #error o Configure kernel sources:
/usr/include/linux/version.h:23:2: #error- cd /usr/src/linux
/usr/include/linux/version.h:24:2: #error- make mrproper
/usr/include/linux/version.h:25:2: #error- make cloneconfig
/usr/include/linux/version.h:26:2: #error- make dep
/usr/include/linux/version.h:27:2: #error
/usr/include/linux/version.h:28:2: #error o When compiling, make sure
to
use the following
/usr/include/linux/version.h:29:2: #errorcompiler option to use the
correct include files:
/usr/include/linux/version.h:30:2: #error
/usr/include/linux/version.h:31:2: #error-I/lib/modules/`uname
-r`/build/include
/usr/include/linux/version.h:32:2: #error
/usr/include/linux/version.h:33:2: #errorinstead of
/usr/include/linux
RE: [Asterisk-Users] Thank You All
I'd be happy to give my docs to the project. I just noticed that it was in progress after I posted but I'd be happy to help. Lane Hoskins, MCP Network Engineer 540.767.7626 -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Monday, January 12, 2004 1:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Thank You All On Mon, 2004-01-12 at 09:31, Lane Hoskins wrote: [snip] The only snags we ran into were during basic configuration due to some things that were written about contexts but not clearly explained. As such we are working on a basic guide/manual similar to the 'Getting Started' pages on the wiki for those who want another perspective on installing and configuring this great system. [snip] This will not be a huge project but should be around 20-30 printed pages letting the noob (like us) get up and running smoothly and pointing to the correct places for help. Again, Thanks to the entire community and I hope that our documentation will be of help. Would you mind contributing your writing to the Asterisk Documentation Project at http://www.asteriskdocs.org/? We could certainly use your help in writing good solid Asterisk documentation. If you use IRC, we're usually hanging out in #asterisk-doc on freenode.net. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inbound call routing problem
I have come to a stumbling block. We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. Lane Hoskins, MCP Network Engineer 540.767.7626 image001.gif
RE: [Asterisk-Users] inbound call routing problem
Thanks David, That is exactly what we had to do. We got some help from Digium as well and have it taken care of. Lane Hoskins, MCP Network Engineer 540.767.7626 -Original Message- From: David Gomillion [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] inbound call routing problem Lane Hoskins wrote: I have come to a stumbling block. We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 - 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. Lane Hoskins, MCP Network Engineer 540.767.7626 I have not done it yet, but it would seem to me that the key to this exercise would be having 7 contexts: 1 for lines 1+2 (which rings all lines or a queue or IVR/ACD) and then one for each line 3-8. This means that each of your incoming lines can have their very own s extension. You can define each line's context in the .conf in Asterisk's etc directory. Hope this helps, David Gomillion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Linux Distribution
Just to add my .02 don't use SuSE. We tried with 8.0, 8.1, 8.2, 9.0 and had a horrible time - zaptel never did compile correctly no matter what we tried. We wound up going to Red Hat 9.0 but from what I've heard most distros will work: Mandrake, Debian, etc. The best advice I have seen is use whatever you're comfortable with. Lane Hoskins, MCP Network Engineer 540.767.7626 -Original Message- From: Jean-Christophe Heger [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:58 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best Linux Distribution Il personally use Mandrake 9.2 and it works perfectly. On Debian, we've never got the FritzCard USB2 ISDN card working, but nothing to do directly with Asterisk. The only performance issue I've got was while running X (many comments around this issue). JC [EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? thanks mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inbound call routing problem - RESOLVED
Thanks we just figure it out a bit ago. It's amazing how simple some things are when you just ask - and then realized that you were making it too hard to begin with!! :-) Lane Hoskins, MCP Network Engineer 540.767.7626 -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] inbound call routing problem On Tue, 2004-01-13 at 07:52, Lane Hoskins wrote: We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 - 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. You need to understand more about contexts. If you put lines 1 and 2 in a context (let's call it [everyone]) and each of the other lines in it's own context (let's say [line3], [line4], etc.), then you can control what happens in each context. If you haven't figured out where to assign a context to each line, it's in your /etc/asterisk/zapata.conf file. After setting those in zapata.conf, your (very simplified) extensions.conf file will look something like this: [everyone] ; ring everyone exten=s,1,Answer() exten=s,2,Dial(SIP/JohnSIP/MarySIP/FredSIP/Bob) [line3] exten=s,1,Answer() exten=s,2,Dial(SIP/John,20,r) exten=s,3,Dial(John's cellphone goes here,10,r) exten=s,4,VoiceMailMain(John's mailbox) exten=s,5,Hangup() exten=s,103,Dial(John's cellphone goes here,10,r) exten=s,104,VoiceMailMain(John's mailbox) exten=s,105,Hangup() exten=s,204,VoiceMailMain(John's mailbox) exten=s,205,Hangup() [line4] exten=s,1,Answer() exten=s,2,Dial(SIP/Mary,20,r) exten=s,3,Dial(Mary's cellphone goes here,10,r) exten=s,4,VoiceMailMain(Mary's mailbox) exten=s,5,Hangup() exten=s,103,Dial(Mary's cellphone goes here,10,r) exten=s,104,VoiceMailMain(Mary's mailbox) exten=s,105,Hangup() exten=s,204,VoiceMailMain(Mary's mailbox) exten=s,205,Hangup() ... etc., etc. ... Hope that gets you started... While this should work, I take no responsibility for typos and or stupid mistakes I may have made while writing this in a hurry... Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thank You All
We are very near to going live with our * system here at my office and I am very excited. I want to thank everyone here on the list who so willingly shares their knowledge and the entire * community in general for working together to put out s great product that is imho far better than most/all commercial offerings. The only snags we ran into were during basic configuration due to some things that were written about contexts but not clearly explained. As such we are working on a basic guide/manual similar to the Getting Started pages on the wiki for those who want another perspective on installing and configuring this great system. **Andy Powell, if you dont mind wed like to copy and paste some of your document into ours as well as provide a direct link to it for clarification of several points we cover. If anyone has found something simple that they didnt quite understand please e-mail me with the question and solution and Ill include it. Again, this is going to be a VERY basic how-to just to get up and running with a configuration similar to ours which is a T10P/channelbank/and * server with snom sip phones. We are not going to try to go any further into specific configuration mostly we will show working examples and try to link to better resources for more detailed explanations; essentially a compilation of other resources with our own experiences commented in. This will not be a huge project but should be around 20-30 printed pages letting the noob (like us) get up and running smoothly and pointing to the correct places for help. Again, Thanks to the entire community and I hope that our documentation will be of help. Lane Hoskins, MCP Network Engineer 540.767.7626 image001.gif
