[asterisk-users] Cisco 7970G line buttons
I just upgraded my 7970G to the SIP firmware. What I'd like to do is have the 8 line buttons be able to make outbound calls using the same account (for practical purposes, same caller-ID). Since the phone is going to have a single public DID, when a call comes in, it should ring on the first available line. So, if I'm on line 1 and a call comes in, it should ring on line 2. How can this be done? Can I configure all lines to the same SIP user and will that do what I want? Do I have to configure each line to a separate user and then setup some roll-over configuration in Asterisk? 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] Kirk IP600 V3 DECT Wireless server
Hi the list, 600v3 with last firmware works fine with Asterisk and SIP. I use it every days with success, no issue. I recommend it and think it's more reliable than WiFi for a great number of handsets or industrial deployment with multicells. Best Regards, Francois BERGERET, France -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Garth van Sittert Envoye : vendredi 29 juin 2007 11:20 A : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Kirk IP600 V3 DECT Wireless server Hi Remco I have used the IP600 v3 with SIP support on Asterisk... apparently I was the 1st person globally to run it at a site. The 1st firmware was a bit buggy at times, but seems to be much better on the later versions. Kind Regards Garth Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO MSN:[EMAIL PROTECTED] Web:www.bitco.co.za Remco Barendse wrote: Hi list! Does anyone have experiences with the updated model of the Kirk IP600? The V3 model is supposed to support SIP instead of only SCCP or H323 which would make the use with Asterisk a lot easier. I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is still giving me severe headaches : - the standard Skinny driver in * doesn't work, only the version of Sergio Chersovani is working but with some stability problems - audible echo - i have 2 kirk handsets and 2 Siemens handsets, the first 2 Kirk handsets always ring on an incoming call, the Siemens handsets only sometimes ring (don't know if this is caused by a problem in the SCCP driver or poor support of the IP600 for non-Kirk phones??) Because of my earlier experiences I am hesitating to try the V3 version. Experiences anyone? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by 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] Query
For this you have to make entry in sip.conf. it will be better if you use host=dynamic in both the phones in sip.conf and what is the IP you are putting in phones which are on your PC. Also check that your both sip phones which are on PC, are sending requestr to asterisk server or not. Kesh. [EMAIL PROTECTED] wrote: Hi, I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running 1st PC is having IP Adress : 192.168.1.149 2nd PC is having IP Adress : 192.168.1.53 I am trying to dial from 1st PC to 2nd PC through asterisk server The problem is 1st PC is calling directly to 2nd PC not through asterisk server I am doing the following additions in configuration files 1) sip.conf [general] context=sip bindport=5060 bindaddr=0.0.0.0 [phone1] type=friend host=192.168.1.149 port=5060 nat=yes dtmfmode=rfc2833 context=sip [phone2] type=friend host=192.168.1.53 port=5060 nat=yes dtmfmode=rfc2833 context=sip 2) extensions.conf exten = 11,1,Dial(SIP/phone2,20,tr) Now, I am calling from sip phone1 by name [EMAIL PROTECTED] It is not being called through asterisk server running on linux m/c. as no packet dumping us taking palce. As, I am running sip debub no messages are seen on screen. What additional routing informations are to be added to sip.conf, inorder to make it work . Thanx and regards sanchal ___ --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 - Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search.___ --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] Cisco 7970G line buttons
On 6/30/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I just upgraded my 7970G to the SIP firmware. What I'd like to do is have the 8 line buttons be able to make outbound calls using the same account (for practical purposes, same caller-ID). Since the phone is going to have a single public DID, when a call comes in, it should ring on the first available line. So, if I'm on line 1 and a call comes in, it should ring on line 2. How can this be done? Can I configure all lines to the same SIP user and will that do what I want? Do I have to configure each line to a separate user and then setup some roll-over configuration in Asterisk? Welcome to our Planet. Here on Planet Earth we have a search engine named Google. Specifically, www.google.com, please reference this amazing resource sometime while you are visiting our planet. We may seem to be simpletons to your advanced civilization, but most of us reference this resource before asking questions that have been answered thousands of times in the past. Thank you, and please enjoy your stay! Remember, www.google.com, it's amazing! ___ --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] awful list delays: 4 days!
It's messages from the list that get delivered after a few days. l. In data Fri, 29 Jun 2007 23:31:44 +0200, Mojo with Horan Company, LLC [EMAIL PROTECTED] ha scritto: Is it taking a while for _your_ messages to post to the list, or do you mean messages from the mailing list software take days to get to you? Moj -- Home of QueueMetrics - http://queuemetrics.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] AEL + Realitme?
Hello
I've spent the past hour searching and reading and searching some
more. I've jet to come to a solution.
I want to have some of my extensions in extensions.ael, while some
should reside in Realtime.
Realtime works just fine, so does my extensions in AEL. But not on the
same time.
This is how my extensions.ael look like now:
context internal
{
100 = Playback(tt-monkeys);
101 = Dial(SIP/cgm);
102 = Dial(SIP/bluecommand);
_9XX = ??
500 = Agi(agi://localhost/internal.agi);
_8XXX =
{
NoOp(Calling ${EXTEN});
Dial(SIP/${EXTEN});
}
};
context database
{
eswitches
{
Realtime/[EMAIL PROTECTED];
};
}
I want it to jump from _9XX to the context database, passing along
what context it came from as a parameter.
Is this possible at all?
Greetings,
Christian
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[asterisk-users] Asterisk 1.4.6 Fedora 7 configure error
Hi guys I'm at a loss in getting ./configure to complete successfully with asterisk 1.4.6 on Fedora 7 x86_64, as it complains about no termcap support, even though it is installed (see below). Any ideas where to go next? checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no configure: error: *** termcap support not found [EMAIL PROTECTED] asterisk-1.4.6]# rpm -qv termcap termcap-5.5-1.20060701.1 [EMAIL PROTECTED] asterisk-1.4.6]# rpm -qv libtermcap libtermcap-2.0.8-46.1 libtermcap-2.0.8-46.1 [EMAIL PROTECTED] asterisk-1.4.6]# rpm -qv libtermcap-devel libtermcap-devel-2.0.8-46.1 libtermcap-devel-2.0.8-46.1 [EMAIL PROTECTED] asterisk-1.4.6]# I also noticed a few WARNING messages about things that were installed but not compilable (see below).What's the problem here? checking sys/select.h usability... no checking sys/select.h presence... yes configure: WARNING: sys/select.h: present but cannot be compiled configure: WARNING: sys/select.h: check for missing prerequisite headers? configure: WARNING: sys/select.h: see the Autoconf documentation configure: WARNING: sys/select.h: section Present But Cannot Be Compiled configure: WARNING: sys/select.h: proceeding with the preprocessor's result configure: WARNING: sys/select.h: in the future, the compiler will take precedence checking for sys/select.h... yes checking for pow... no checking for pow in -lm... no configure: WARNING: cannot find library containing definition of pow checking linux/soundcard.h usability... no checking linux/soundcard.h presence... yes configure: WARNING: linux/soundcard.h: present but cannot be compiled configure: WARNING: linux/soundcard.h: check for missing prerequisite headers? configure: WARNING: linux/soundcard.h: see the Autoconf documentation configure: WARNING: linux/soundcard.h: section Present But Cannot Be Compiled configure: WARNING: linux/soundcard.h: proceeding with the preprocessor's result configure: WARNING: linux/soundcard.h: in the future, the compiler will take precedence checking for linux/soundcard.h... yes Thanks for great software, this is the first time I have had problems with the installation. Cheers Bill -- Bill Maidment Maidment Enterprises Pty Ltd www.maidment.vu ___ --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 Compilation
bilal ghayyad wrote: Hi List; I am facing a problem relaed to menuselect when I am trying to compile zaptel -1.4.2.1, the error as following: [EMAIL PROTECTED] zaptel-1.4.2.1]# make linux26 m You no longer need to do a make linux26. Just delete the zaptel directory and extract a fresh copy from the .gz file. Do the following: ./configure make make install You were carbon copied on this email, since it's taking days for it to show up on the list. 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] Rining 180 and 183
you should turn on sip debug on asterisk and median and see, if sip/180 ringing messagess are propagated through mediant to avaya, avaya should react to sip/180 ringing with generating ringback to calling phone... sip/183 is progress message, in this case is audio path open to playback progress messages to calling phone like destination is not available etc... PJ satish patel wrote: Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-[mediant 2000][Avaya] when i call from avaya side to --- asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of this type of problem regards Satish patel - Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. ___ --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] Fwd: Problems with zap following 1.4.6 install
Thanks for responding Russell! What output do you get if you run module unload chan_zap.so == Unregistered application 'ZapSendKeypadFacility' and then module load chan_zap.so ? == Registered application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Jun 30 07:46:51] WARNING[12132]: chan_zap.c:903 zt_open: Unable to specify channel 1: No such device or address [Jun 30 07:46:51] ERROR[12132]: chan_zap.c:7157 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 [Jun 30 07:46:51] ERROR[12132]: chan_zap.c:10463 build_channels: Unable to register channel '1' Hopefully you'll be able to tell me what the above means! I'm using the same zapata.comf that I used with 1.2... Thanks, H ___ --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] Fwd: Problems with zap following 1.4.6 install
Figured it out. I forgot that /etc/zaptel.conf got clobbered during the install. Once I fixed that and ran: /sbin/ztcfg - It worked. Zaptel.conf looks like this: loadzone = us defaultzone=us fxoks = 1-2 fxsks = 4 I have 1 FXO module 2 FXS modules. Thanks for putting me on the right track!! H On 6/30/07, hugolivude [EMAIL PROTECTED] wrote: Thanks for responding Russell! What output do you get if you run module unload chan_zap.so == Unregistered application 'ZapSendKeypadFacility' and then module load chan_zap.so ? == Registered application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Jun 30 07:46:51] WARNING[12132]: chan_zap.c:903 zt_open: Unable to specify channel 1: No such device or address [Jun 30 07:46:51] ERROR[12132]: chan_zap.c:7157 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 [Jun 30 07:46:51] ERROR[12132]: chan_zap.c:10463 build_channels: Unable to register channel '1' Hopefully you'll be able to tell me what the above means! I'm using the same zapata.comf that I used with 1.2... Thanks, H ___ --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] Asterisk 1.4.5 : Use I386 ou X86_64
Hi Everyone, Im testing a ML100 G4 (Pentium D) Server from HP with a TDM400P from Digium. I just installed, with success, the following O.S. with Asterisk 1.4.5 1) Centos 4.4 2) Centos 4.5 3) Centos 5.0 Id like to receive a recommendation about whats S.O do you recommend install for Asterisk 1.4.5 a) Centos 4.x ou Centos 5.x b) I386 ou X86_64 Thanks in Advanced, André Lomonaco ___ --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 Compilation
In zaptel compilation, in the newer version in both 1.2 and 1.4, make linux will not work. To install the zaptel-1.4 you can use these lines:-- cd zaptel1.4.XX ./configure make menuselect make make install Regards, Keshav Doug Lytle [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi List; I am facing a problem relaed to menuselect when I am trying to compile zaptel -1.4.2.1, the error as following: [EMAIL PROTECTED] zaptel-1.4.2.1]# make linux26 m You no longer need to do a make linux26. Just delete the zaptel directory and extract a fresh copy from the .gz file. Do the following: ./configure make make install You were carbon copied on this email, since it's taking days for it to show up on the list. 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 - Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. ___ --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] 1.4.5
I have installed asterisk 1.4.5 on Centos 4.4 with the kernel, 2.6.9-55.ELsmp In starting it showed some errors related to gtk2, while running make. After updating gtk It has been installed and working , I have test few basic thing , which are fine , but it has echo very much.. i'll test it further and will update if i will face any issue. --Keshav - Shape Yahoo! in your own image. Join our Network Research Panel today!___ --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] Missing 'init keys' command
Jonathan Unai Marquez wrote: Thanks for your answer Jared, but I also tried that with no luck: Connected to Asterisk 1.4.5 currently running on moe (pid = 22879) -- Remote UNIX connection Verbosity is at least 6 moe*CLI keys show No such command 'keys show' (type 'help' for help) Any clue of what can be wrong with my intallation? You probably don't have the res_crypto module compiled and installed. The most likely reason is that you didn't have openssl and the related development package installed. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] mISDN Fritz Passive Card
Hi all, I am using mISDN with a Fritz!PCI v2.0. The card works without problems until I use concurrently the two channels. In that moment appears a little echo. I have reduced the tx gain but the echo not disappear. I see in messages: kernel: ECHOCAN: i:4000 TXBUF Overflow txbuflen:496 txcancellen:32 Some idea how resolve this problem? Thanks, -- Manuel ___ --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] 1.4.5
Keshav K. wrote: I have installed asterisk 1.4.5 on Centos 4.4 with the kernel, 2.6.9-55.ELsmp In starting it showed some errors related to gtk2, while running make. You are probably talking about output from the configure script. You can ignore it, as it is not actually a problem. I'll work on making that error output not actually show up, though. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] .call file
Paul wrote:
I'm going to top post in this situation.
Kevin - Commands that operate on the channel variables won't help if we
are using a call file. We will have a new channel.
Agreed, I misread and thought he was trying to generate a call file.
-Kevin
This syntax works with asterisk 1.2.x for me:
Application: AGI
Data: say_it.php|call_status_message
I have done other things where a bunch of parameters are stored in
postgres or mysql and the only parameter I pass via the call file is the
record key. The php script receives the key as a parameter and gets
everything else from the db. Something like this:
Application: AGI
Data: inform.php|68456943
Kevin Smith wrote:
Nitesh Divecha wrote:
Hello All,
Is there any way to pass additional parameters while calling AGI from
*.call file?
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php
Something like Data: recordvoice.php?id=3453name=asterisk
Cheers,
Nitesh
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I'm not 100% sure if you can pass it directly, but you can use the set
option in the call file to set local variables within Asterisk and then
pass them to the AGI script. So for your example it would be.
Set: name=asterisk
This will set the variable ${name} in asterisk and depending how your
script was created you should be able to grab the variable to use within
the script. If you are using say the PHP AGI you can use something like
the following:
$var = $agi-get_variable(name);
This will create an array with $var['data'] holding 'asterisk';
Now one more thing I am not sure of is for multiple variables (haven't
tried it yet ;D ). You may have to do it one of two ways.
Set: name=asterisk, id=3453
or
Set: name=asterisk
Set: id=3453
and if those don't work, just format it so you can filter it out with PHP.
Hopefully this will help.
Kevin
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Re: [asterisk-users] Zaptel Compilation
On Fri, Jun 29, 2007 at 07:40:30AM -0700, bilal ghayyad wrote: Hi Steve; I did what I told me below, and look like going fine but I do not know how can I know that zaptel compilation was implemented successfully specially I do not have a message in the end indicate this, Generally such commands exit with an error if there is any problem. No special message should be needed. If you have not recieved an error, the make command has finished successfully. If you have recieved an error you need to try to figure out what it is. (Which is why I find the handling of building h323 by failing make so buggy) please find below what the make and make install commands (for zaptel compilation) was ended by (please let me know if that is normal and the compilation was successfully done): This for make: make[2]: Leaving directory `/usr/src/asterisk-1.4/zaptel-1.4/xpp/utils' gcc -c -fPIC -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c gcc -c -fPIC -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c gcc -shared -Wl,-soname,libtonezone.so.1.0 -o libtonezone.so zonedata.lo tonezone.lo -lm make[1]: Leaving directory `/usr/src/asterisk-1.4/zaptel-1.4' This for make install: *** *** WARNING: *** If you had custom settings in /etc/modprobe.d/zaptel, *** they have been moved to /etc/modprobe.d/zaptel.bak. *** *** In the future, do not edit /etc/modprobe.d/zaptel, but *** instead put your changes in another file *** in the same directory so that they will not *** be overwritten by future Zaptel updates. *** What's it? Nothing more? What is the output of 'echo $?' (checking the exit status) immediately after 'make install'? -- Tzafrir Cohen icq#16849755jabber:[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] Polycom echo problem
I have three polycom 501 that are all hearing echo. The other party sounds fine but you can hear yourself rather well. The volume does help if lowered but that also makes the other party extremely quiet. Is there any way to control the gain of the mic or stop the microphone from picking up so much from the handset. It only happens while you are on the handset. ___ --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] 1.4.5
Keshav K. escribió: I have installed asterisk 1.4.5 on Centos 4.4 with the kernel, 2.6.9-55.ELsmp In starting it showed some errors related to gtk2, while running make. After updating gtk It has been installed and working , I have test few basic thing , which are fine , but it has echo very much.. i'll test it further and will update if i will face any issue. --Keshav Hi, I experienced the same but audio was choppy too. Probably my PC's hardware limitation. But the 600 echo test was awful. Regards, Francisco Neira Lima, Peru -05:00 UTC --- EC Red Internet [EMAIL PROTECTED] Inscríbete en www.ec-red.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] Polycom echo problem
I had the same situation and I had to replace my T1 card with the one with hardware echo canceller. All other solutions were failed. May be you need to do the same if you're on a PRI or using PSTN lines. If you're on a pure VoIP network, then its the phones. On 6/30/07, Jordan Novak [EMAIL PROTECTED] wrote: I have three polycom 501 that are all hearing echo. The other party sounds fine but you can hear yourself rather well. The volume does help if lowered but that also makes the other party extremely quiet. Is there any way to control the gain of the mic or stop the microphone from picking up so much from the handset. It only happens while you are on the handset. ___ --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 -- Zeeshan A Zakaria ___ --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] G729 , upgrade asterisk
I'm planning to upgrade my asterisk 1.4.4 to 1.4.6. usually for asterisk upgrade i delete modules directory and include, then compile the new version. Since i have couple of G729 Licenses on this server installed, would i need to call Digium to reactivate these Licenses? Is there any better and faster way of upgrade asterisk? Possibly without losing G729 License? Thanks! _ Play free games, earn tickets, get cool prizes! Join Live Search Club. http://club.live.com/home.aspx?icid=CLUB_wlmailtextlink___ --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] G729 , upgrade asterisk
Licenses are stored in /var/lib/asterisk/licenses, not in the module itself. Won't need any reactivation between versions either. There is no real need to delete the modules folder between minor versions like this, 'make install' will overwrite the modules and warn you if there are any extra ones in there (it should always warn about the g729 module). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AL Daei Sent: Saturday, June 30, 2007 18:12 To: asterisk-users@lists.digium.com Subject: [asterisk-users] G729 , upgrade asterisk I'm planning to upgrade my asterisk 1.4.4 to 1.4.6. usually for asterisk upgrade i delete modules directory and include, then compile the new version. Since i have couple of G729 Licenses on this server installed, would i need to call Digium to reactivate these Licenses? Is there any better and faster way of upgrade asterisk? Possibly without losing G729 License? Thanks! Play free games, earn tickets, get cool prizes! Join Live Search Club. Join Live Search Club! http://club.live.com/home.aspx?icid=CLUB_wlmailtextlink ___ --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] G729 , upgrade asterisk
Darryl Dunkin wrote: Licenses are stored in /var/lib/asterisk/licenses, not in the module itself. Won't need any reactivation between versions either. There is no real need to delete the modules folder between minor versions like this, 'make install' will overwrite the modules and warn you if there are any extra ones in there (it should always warn about the g729 module). Yeah, the only time you should delete *everything* from the modules directory, is when upgrading between major versions, such as from 1.2 to 1.4. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] Polycom echo problem
Zeeshan Zakaria wrote: I had the same situation and I had to replace my T1 card with the one with hardware echo canceller. All other solutions were failed. May be you need to do the same if you're on a PRI or using PSTN lines. If you're on a pure VoIP network, then its the phones. On 6/30/07, *Jordan Novak* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have three polycom 501 that are all hearing echo. The other party sounds fine but you can hear yourself rather well. The volume does help if lowered but that also makes the other party extremely quiet. Is there any way to control the gain of the mic or stop the microphone from picking up so much from the handset. It only happens while you are on the handset. If you're using FXO/analog lines, I'd recommend trying Octware's software echo canceler. I had the same problem on a recent installation and it fixed it. At $10.00 per channel, its a very good value, IMO. -- 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] awful list delays: 4 days!
Have a look below: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- It used to be easynews. Maybe your servers didn't take notice to the new DNS changes for a bit ? - Original Message - From: lenz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, June 30, 2007 12:40 PM Subject: Re: [asterisk-users] awful list delays: 4 days! It's messages from the list that get delivered after a few days. l. In data Fri, 29 Jun 2007 23:31:44 +0200, Mojo with Horan Company, LLC [EMAIL PROTECTED] ha scritto: Is it taking a while for _your_ messages to post to the list, or do you mean messages from the mailing list software take days to get to you? Moj -- Home of QueueMetrics - http://queuemetrics.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 ___ --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 wifi IP phone for asterisk (LINKSYSSUPPORT QUALITY)
Don't you just love outsourcing ? - Original Message - From: Michelle Dupuis [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, June 26, 2007 3:39 PM Subject: Re: [asterisk-users] Best wifi IP phone for asterisk (LINKSYSSUPPORT QUALITY) I haven't been too impressed with the WIP330 - but my experience with Linksys tech support has been disastrous! I spent approximately 50 minutes on hold, I was transferred between 4 different people (all of whom had a poor grasp of the English language), none of them understood the features of the phone. All they could do was read the user manual I already had in front of me. None of them could explain the auto-provisioning feature mentioned on the Linksys website, or on the phone menus, etc. None of them even understood what SIP and RTP protocols were. They were just there to read the manual to me. Here's a warning for the group...watch our for Linksys! -MD- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Seraphin Sent: Tuesday, June 26, 2007 4:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best wifi IP phone for asterisk On Tue, 26 Jun 2007, Hendrik Visage wrote: On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: HAve a look at the Linksys WIP 300 (or something) Can be charged from the USB port I have a WIP330 and it doesn't work. Maybe it needs a firmware upgrade. Maybe it's a defective unit and all the others work fine. I haven't called support yet because I haven't had the chance. Audio in one direction cuts out completely for about 4 seconds every 10 seconds during the call. For 10 seconds it works fine... audio in both directions... then 4 seconds of silence in one direction... then 10 seconds of normal, etc etc, repeating forever. Completely unusable as-is. All my other Linksys IP Phones work great, though. I only have one WIP330. I don't have any WIP300's. My recommendation: Whatever you go with, buy ONE first for testing to make sure you're happy with it BEFORE you buy a boatload of them. -- Nick ___ --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
