[Asterisk-Users] Re: PSTN Call drops randomly

Alex, Chris and Eric: Based on your kind suggestions, I ade the following modifications in the zapata.conf: 1. busydetect=no 2. commented out busycount=xx 3. comented out switchtype=national I did make a 30 minute and an hour call on two servers during the day before modifications. Did not disco

Re: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions

On Fri, 23 Apr 2004, Paul Tyreman wrote: > Would it be possible for you to provide some more info on this. There are two[1] SCCP channels available for Asterisk: - chan_skinny: part of Asterisk (?), basic support, limited development - chan_sccp: derived from chan_skinny, separately downloaded,

RE: [Asterisk-Users] Fax problem

Use ulaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela Sent: Friday, April 23, 2004 7:52 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fax problem Hi, We have a machine with an *'s with Digium TDM400P and connected wit other machine wi

Re: [Asterisk-Users] newbie install problems

On Fri, 23 Apr 2004, Rubens Zupelli Filho wrote: > I'm using a Debian kernel 2.4.22 with all (second the archives/docs) > pre-reqs packages. Quick and dirty: instead of trying to compile/build, just do this: apt-get -t sarge install asterisk/unstable Remember to have the appropriate entry for un

[Asterisk-Users] newbie install problems

Hi I'm new with asterisk, and I´m trying to install it with a X100P card. I stuck on some problems/issues. Could someone clarify me some points ? 1. I did a cvs checkout, did a 'make' and 'make install' . From the make output seems that some headers (.h) files for some codecs are missing. Is th

Re: [Asterisk-Users] IAXPHONE failures in calls to Cisco Phones

MLS Drop for SysAdmin wrote: Has anyone else experienced this, or does anyone have idea what's wrong? Check your codecs. I'm not intimately familiar with the 7960s, but it sure smells like that is a likely possibility. B. ___ Asterisk-Users mailing l

Re: [Asterisk-Users] Festival problems

On Saturday 24 April 2004 01:06, Jeff Workman wrote: > I tried it without the quotes as well, same thing. > > > Be sure you start Festival "before" starting *. > > Did that as well. Not sure what the problem is. I've just been installing Festival this evening and got it working after a couple o

[Asterisk-Users] TekDigitel iPRO and *?

Has anyone used TekDigitel iPRO with * ? http://www.tekdigitel.com/website/htmlPages/content/products/product_introductions/introduction_to_V-SERVER_iPRO_Dual_Ethernet.htm It looks like combo FXO/FXS device which speaks H323 only. Does it work well (or at all) with * ? -Dan ___

[Asterisk-Users] IAXPHONE failures in calls to Cisco Phones

I have been operating a functional asterisk system using Fedora in a 500 MHz Pentium III Stations are Cisco 7960s and Grandstream 102s.  We needed to identify a software based phone to handle traveling users, so we tried IAXPHONE's latest version. Interestingly, calls from the IAX client to 7960

Re: [Asterisk-Users] Festival problems

--On Friday, April 23, 2004 5:13 PM -0600 Rich Adamson <[EMAIL PROTECTED]> wrote: My /etc/asterisk/festival.conf looks like this: My conf looks the same. My extension looks like this: exten => 603,1,Answer() exten => 603,2,Festival('this is a test testing 1 2 3') Try syntax like the followin

[Asterisk-Users] Call Queues, Call groups

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] Cisco phones

Paul Tyreman wrote: All I can find on that Cisco website is this: http://www.cisco.com/pcgi-bin/cpn/cpn_match_result.pl?CurPosition=0&Direction=&ResultType=EC&search_id=156576&tab_name=findsp&country_id=GB

[Asterisk-Users] Hangup in AGI

If I call the Hangup command from AGI directly of via EXEC Hangup it does not work. If shows on the console but it does not hangup. It continues on to the rest of the priorities in the dialplan.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://

RE: [Asterisk-Users] zaprtc on 2.6

The warning is harmless. I built on the existing work for compiling modules for 2.6, which required the source in linux-2.6 (which you could mod in the makefile), but you also need for the makefile in ../zaptelrtc to setup the rest of the build environment. Using the command make -C /usr/src/linu

Re: [Asterisk-Users] WARNING[1074420448]

listas iPfone wrote: plase somebody help me... You should help yourself first and read the asterisk/channels/h323/README. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUB

Re: [Asterisk-Users] zaprtc on 2.6

On Friday 23 April 2004 22:58, Dan Austin wrote: > I should add a small readme to the patch. The 2.6 series build process > creates a symlink to identify the correct irq_vectors.h during the build > process then clears it once the build is complete. I modified the file > ../include/asm/irq.h to h

Re: [Asterisk-Users] Festival problems

> My /etc/asterisk/festival.conf looks like this: My conf looks the same. > My extension looks like this: > > exten => 603,1,Answer() > exten => 603,2,Festival('this is a test testing 1 2 3') Try syntax like the following (this works): exten => 3913,2,Festival(mary had a little lamb) Be sure

RE: [Asterisk-Users] zaprtc on 2.6

I should add a small readme to the patch. The 2.6 series build process creates a symlink to identify the correct irq_vectors.h during the build process then clears it once the build is complete. I modified the file ../include/asm/irq.h to have an absolute reference to irq_vectors.h to resolve tho

RE: [Asterisk-Users] MP3 encoding of Monitor files

Try LAME with "-m m" (for mono encoding). I'm guessing that the encoder isn't recognizing the channel-count properly, thinks it's dealing with a stereo file, and then splits the PCM data into two channels, effectively making the file double the speed. Can Monitor record raw PCM (no WAV header)?

[Asterisk-Users] zaprtc on 2.6

Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machi

[Asterisk-Users] Asterisk configuration inside a DMZ w/SIP

Hello all,   I’m having a nightmare of a time trying to get stable results with SIP clients on Asterisk.  I can’t seem to find a configuration that works!  In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall.   Originally, I had configured Asterisk to run

Re: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear

On Apr 22, 2004, at 23:48, Olle E. Johansson wrote: Geert Nijpels wrote: Ian White wrote: On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report

Re: [Asterisk-Users] UK ISDN PRI Problems

Hi Chris, > I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which > is currently working happily with an SDX Index phone system. I have to I can't see particular problems with your config, but I have a few comments: [snip] > I have a Digium E100p card which is configured in zap

Re: [Asterisk-Users] call initiation

On Fri, 2004-04-23 at 15:36, Roger wrote: > Steven Critchfield wrote: > > >I'm assuming you mean asterisk when you say pbx, > > > Since this is the asterisk mailing list - yes. > > >and I am assuming you > >mean SIP phones from the 'send' or 'dial' comment. > > > > > Yes. That I should of spe

Re: [Asterisk-Users] Problem with instalation T100P

Steven's a bit touchy, but he's right. You've got a kernel mismatch here. Probably you need to recompile the modules under your new kernel. John Bartosz Jozwiak wrote: On Fri, 2004-04-23 at 13:04, Bartosz Jozwiak wrote: On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote: When I do modeprobe

Re: [Asterisk-Users] Polycom registration

Keep reading the instructions. You simply must invest some time here. On the page it says: "To get the full documentation and recent firmware releases for these phones go to this website: http://www.freedomphones.net/polycom/files/ (cache) The firmware files also contain some default configura

Re: [Asterisk-Users] zaptel on Fedora (Core 1) RedHat Linux-2.4

On Sat, 24 Apr 2004 01:21:09 +0530, Satish Kumar wrote > Hi! > > I have fresh installation of Fedora (Core 1) RedHat Linux-2.4.22-1.2115.nptl > The kernel source is in place /usr/src/linux-2.4.22-1.2115.nptl > The config file for the kernel is in place /boot/config-2.4.22-1.2115.nptl

Re: [Asterisk-Users] call initiation

Steven Critchfield wrote: I'm assuming you mean asterisk when you say pbx, Since this is the asterisk mailing list - yes. and I am assuming you mean SIP phones from the 'send' or 'dial' comment. Yes. That I should of specified. This sounds like a problem with the dialplan in the individual

[Asterisk-Users] MGCP problem

Title: MGCP problem The latest cvs mgcp code seems to be slightly broken.  I have two gateways and both of them only report their ip address to * instead of a hostname.  When I audit the endpoints, * properly places the ip address in brackets and the audit is successful.  The phones connected

RE: [Asterisk-Users] PSTN Call drops randomly - Email found in subject

Set busydetect=no in your zapata.conf file. That should stop the random hang-ups. If you really need busy detection, try setting busycount=8 or even 10. If you still get random hang-ups, turn off busy detection and turn on call progress. May help the situation. Gregory P. Scasny Golden Technol

[Asterisk-Users] WARNING[1074420448]

Hello all,   I just installed the h.323 drivers, make all the process etc.. and i get that error message and asterisk dont load:  [chan_h323.so]Apr 23 17:16:39 WARNING[1074420448]: loader.c:239 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file ordire

Re: [Asterisk-Users] PSTN Call drops randomly

Set callprogress=no and busycount=6 or higher in /etc/asterisk/zapata.conf On Fri, 2004-04-23 at 14:38, Shahid Mahmood wrote: > Dear List members, > After succesfully installing the * on a couple of systems, and putting > them on test, I observed that there is an intermittent call drop on > PSTN l

Re: [Asterisk-Users] zaptel on Fedora (Core 1) RedHat Linux-2.4

Just making sure... Rule one for getting kernel source for anything Do this before copying the config from /boot/. make mrproper This will clean out the source tree. This may help also... (I always make sure this is done.) Look at the Makefilemake sure the EXTRAVERSION matches you r

Re: [Asterisk-Users] call initiation

Others correct me if I'm wrong, but I believe that: Presumably, you have lines like: exten => _XXX,1,Dial(SIP/${EXTEN}) exten => _9NXX,1,Dial(ZAP/1/${EXTEN:1}) exten => _91NXXNXX,1,Dial(Zap/1/${EXTEN:1}) if none of your internal extensions start with a nine (ie if they are in the range 00

RE: [Asterisk-Users] 3com SIP phone working with asterisk

I just tried basic calling...price-wise, around $100 or so. Lisa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: Friday, April 23, 2004 11:26 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3com SIP phone working with aster

Re: [Asterisk-Users] PSTN Call drops randomly

Shahid, Looking below here are a few thoughts: Why are you defining switchtype and signalling for a switch? You don't need these for an X100P card. check your card make sure it's on it's own interrupt (/proc/interrupts) add echotraining=1 with your other echo commands. I had a problem like t

Re: [Asterisk-Users] call initiation

On Fri, 2004-04-23 at 14:39, Roger wrote: > Users withing the office can dial a 3 digit extension and that will ring > a phone. The problem I'm running into is you have to press xxx then > press 'send or 'dial'. The pbx doesn't recognize a 3 digit number as an > internal extension and automati

RE: [Asterisk-Users] call initiation

Roger, Maybe you are using extensions like "_9." try to put de complete number in your estension.conf ej; exten => _9XXX,1,Dial(. exten => 101,1,Dial(Zap/1) in that case send congestion if the 3 digits extensions are not in extensions.conf. Regards, Pedro J. Vela Ru

Re: [Asterisk-Users] call initiation

Roger wrote: Users withing the office can dial a 3 digit extension and that will ring a phone. The problem I'm running into is you have to press xxx then press 'send or 'dial'. The pbx doesn't recognize a 3 digit number as an internal extension and automatically dial it the user has to initi

[Asterisk-Users] Busy error

Hi, When have a incoming call from E1 to a extension FXS, and this extension is busy, the incoming call recive ring tone, and it is wrong. What can I do? Thanks in advance Pedro Here is the trace: asterisk-1*CLI> < Protocol Discriminator: Q.931 (8) len=41 < Call Ref: len= 2 (reference 66/0x42

Re: [Asterisk-Users] PSTN Call drops randomly

Do you need the busy detect feature as this often can cause random drop outs. Try changing busydetect=yes to busydetect=no in zapata.conf. Assuming that asterisk still works (i.e. it detects when people hang up and when lines are busy (most lines will do it ok but some won't (e.g. ntl cable in

[Asterisk-Users] zaptel on Fedora (Core 1) RedHat Linux-2.4

Hi!I have fresh installation of Fedora (Core 1) RedHat Linux-2.4.22-1.2115.nptlThe kernel source is in place /usr/src/linux-2.4.22-1.2115.nptlThe config file for the kernel is in place /boot/config-2.4.22-1.2115.nptlAs per zaptel instruction, i first created the kernel config file as follo

[Asterisk-Users] call initiation

Users withing the office can dial a 3 digit extension and that will ring a phone. The problem I'm running into is you have to press xxx then press 'send or 'dial'. The pbx doesn't recognize a 3 digit number as an internal extension and automatically dial it the user has to initiate that call.

[Asterisk-Users] PSTN Call drops randomly

Dear List members, After succesfully installing the * on a couple of systems, and putting them on test, I observed that there is an intermittent call drop on PSTN line. The systems are - Dell Optiplex P3/500MHz/128MB - Built-in ethernet - 1 X100P (Motorolla chip) card on PCI - 10G HDD etc. - Aster

Re: [Asterisk-Users] IAX clients are Unmonitored / UNREACHABLE

It doesn't matter what ports you run at. But the only way to make it work with different ports right now is to change the driver source and recompile. That's what I meant by enabling the code. Regarding the qualify, Asterisk monitors the connection to the host by sending probes or pings and uses

Re: [Asterisk-Users] Polycom registration

John Baker wrote: Try following the instructions at http://www.voip-info.org/wiki-Polycom+Phones I think you don't have your MACADDRESS.cfg file set right. I've never used the web interface. The problem is I don't seem to have a reference to the format of that file or what options I can put

Re: [Asterisk-Users] Zaphfc

In this case is ok but if the number is a mobile phone is wrong infact for example a mobile phone cid=347xx---> + 0 = 0347xx is wrong (the correct is 347) Thank's Tiziano At 18.54 23/04/2004 +0200, you wrote: You can do something like : [incoming] exten => s,1,Answer exten => s,2

[Asterisk-Users] UK ISDN PRI Problems

Advance apologies for the length of this mail; I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have

Re: [Asterisk-Users] Problem with instalation T100P

> On Fri, 2004-04-23 at 13:04, Bartosz Jozwiak wrote: > > > On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote: > > > > When I do modeprobe wct1xxp I get it : > > > > > > > > modprobe wct1xxp > > > > /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol > > > > create_proc_entry_R1b235e62 > > >

Re: [Asterisk-Users] Problem with instalation T100P

On Fri, 2004-04-23 at 13:04, Bartosz Jozwiak wrote: > > On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote: > > > When I do modeprobe wct1xxp I get it : > > > > > > modprobe wct1xxp > > > /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol > > > create_proc_entry_R1b235e62 > > > > > > > > > /

Re: [Asterisk-Users] Problem with instalation T100P

Is this a new kernel? Did you recompile your modules under the new kernel after making it? John Bartosz Jozwiak wrote: When I do modeprobe wct1xxp I get it : modprobe wct1xxp /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol create_proc_entry_R1b235e62 /lib/modules/2.4.18-386/misc/zapt

Re: [Asterisk-Users] Polycom registration

Try following the instructions at http://www.voip-info.org/wiki-Polycom+Phones I think you don't have your MACADDRESS.cfg file set right. I've never used the web interface. If it still doesn't work after that, write back. John P.S. Make sure you use a good xml editor when fixing up the cfg

Re: [Asterisk-Users] Problem with instalation T100P

> On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote: > > When I do modeprobe wct1xxp I get it : > > > > modprobe wct1xxp > > /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol > > create_proc_entry_R1b235e62 > > > > > /lib/modules/2.4.18-386/misc/zaptel.o: insmod > > /lib/modules/2.4.18-386

Re: [Asterisk-Users] Problem with instalation T100P

On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote: > When I do modeprobe wct1xxp I get it : > > modprobe wct1xxp > /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol > create_proc_entry_R1b235e62 > /lib/modules/2.4.18-386/misc/zaptel.o: insmod > /lib/modules/2.4.18-386/misc/zaptel.o fail

[Asterisk-Users] Fax problem

Hi, We have a machine with an *'s with Digium TDM400P and connected wit other machine with *'s an TDM400P too. Well, I have a fax connected to each machine, and the protocol in the middle is IAX2 alaw. The fax between two fax, on in each machine, not work. The fax answer, but error in comm. Whic

[Asterisk-Users] Festival problems

After patching and installing Festival, I am unable to get it to do anything useful. I get the following error message on the * console when I dial the test extension: Parsing '/etc/asterisk/festival.conf': Found Apr 23 13:43:06 WARNING[1226062640]: app_festival.c:382 festival_exec: Strings do

[Asterisk-Users] oh323 goes silent after 5 seconds

I have this problem trying to talk to an ADDPAC gateway using oh323, when I call the sound is great for the first 5 seconds then it goes almost silent... all you can hear are some clicks every once in a while.   Anybody seen this can point me to some config settings to change?  

[Asterisk-Users] Exception flag warnings

I keep seeing the following errors in my asterisk logs: Apr 23 12:13:36 WARNING[1226062640]: Exception flag set on 'SIP/Phone1-c016', but no exception handler Apr 23 12:23:37 WARNING[1268026160]: You might not have the soxmix installed and available in the path, please check. The soxmix one i

Re: [Asterisk-Users] MP3 encoding of Monitor files

lame did the same thing. The reason I ask this on the asterisk list is that .wav files I record from other sources encode just fine. I think the hitch is the sample rates produced by asterisk. File recorded by gnome sound recorder (lame/bladeenc encode just fine): RIFF (little-endian) data, WAVE

[Asterisk-Users] Problem with instalation T100P

When I do modeprobe wct1xxp I get it : modprobe wct1xxp /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol create_proc_entry_R1b235e62 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_unregister_Re139a4b3 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol __pollwait_Rdead

Re: [Asterisk-Users] Polycom registration

Roger wrote: I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_re

[Asterisk-Users] Polycom registration

I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Regist

RE: [Asterisk-Users] Cisco phones

Well, After a lot of searching (and a few calls to Cisco) here is what I came up with. When you buy a Cisco IP phone used in order to legally use it you must first re-license the firmware through an authorized Cisco Provider. Once you re-license the phone you can then get the service contrac

Re: [Asterisk-Users] Zaphfc

You can do something like : [incoming] exten => s,1,Answer exten => s,2,SetCallerID(0${CALLERID}) enten => s,3, There is maybe a better way to do the samething. At 18:40 23/04/2004, you wrote: How i can obtain a complete caller ID from ISDN zaphfc in italy because i obtain a caller id without

[Asterisk-Users] Planning Asterisk

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line

[Asterisk-Users] Zaphfc

How i can obtain a complete caller ID from ISDN zaphfc in italy because i obtain a caller id without a initial 0 (for example cid=305001010 the correct number is 0305001010)   Thank's Tiziano

Re: [Asterisk-Users] Cisco phones

All I can find on that Cisco website is this:   http://www.cisco.com/pcgi-bin/cpn/cpn_match_result.pl?CurPosition=0&Direction=&ResultType=EC&search_id=156576&tab_name=findsp&country_id=GB   I can't see the likes of BT, O2, Vodaphone etc wanting to deal with me !         -Original Message--

Re: [Asterisk-Users] Cisco phones

Paul Tyreman wrote: I have bough a cisco phone on eBay to use with Asterisk, but according to that website, you need a contract with Cisco systems to upgrade the phone to work with SIP. I am guessing the phone that I get won't come with that as it was used with the cisco call manager software

Re: [Asterisk-Users] X100P Answer

You've probably got callerID enabled in zapata.conf. That will cause a wait of several rings whilst * looks for the caller ID info. Since this only works in the US (or pkaces with similar phone systems), disabling it in other territories saves the ring delay. Make sure you have this in zapata

Re: [Asterisk-Users] MP3 encoding of Monitor files

use lame Il ven, 2004-04-23 alle 17:33, Mike Machado ha scritto: > I have having problems trying to take a file recorded with Monitor and > convert it to MP3. When I use 'play' to play the .wav file, it sounds > fine. After bladenc'ing it, it plays at lightening speed, and the voices > are all hig

Re: [Asterisk-Users] MP3 encoding of Monitor files

On Fri, 2004-04-23 at 10:33, Mike Machado wrote: > I have having problems trying to take a file recorded with Monitor and > convert it to MP3. When I use 'play' to play the .wav file, it sounds > fine. After bladenc'ing it, it plays at lightening speed, and the voices > are all high pitch. I tried

Re: [Asterisk-Users] Cisco phones

On Fri, 23 Apr 2004, Paul Tyreman wrote: > I have three questions to ask about this: > 1) How do I know if my phone qualifies for a service contrct ? When you (try to) buy your service contract, you will need to give the model and serial number of the item you are trying to include into your co

[Asterisk-Users] H323 error

While calling to H323 peer     *CLI> 1:22:59.944   H225 Caller:81e5c48   assert.cxx(105)   PWLib   Assertion fail: Invalid array element, file /root/pwlib/include/ptlib/array.h, line 1183, Error=115   bort, ore dump, gnore?*CLI>     *CLI> show versionAsterisk CVS-04/22/04-23:56:0

Re: [Asterisk-Users] * INSTRUCTIONS

On Fri, 23 Apr 2004 03:55:57 -0400, tmpm <[EMAIL PROTECTED]> wrote: >Might I humbly request someone, somewhere in the community establish a >"dummies guide to asterisk" kind of site, that explains in detail what the >cryptic scripts actually do, line by line. >The Wiki is helpful, but unless you

Re: [Asterisk-Users] Extension buttons

On 06:36 AM 4/23/2004, John Todd wrote: >Is it possible (ignoring Asterisk for the minute) for Polycom phones >to indicate visually (on the LCD or on a lighted "extension" button >or something) that a particular line is in use? I would expect this >method to be via NOTIFY or SUBSCRIBE calls from a

Re: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions

Hi,   Would it be possible for you to provide some more info on this.   I have just bought a Cisco 7960 on eBay, but only now has the reality of needing a login to upgrade to SIP become clear.   Can you tell me how you managed to get your phone going on Asterisk without the image change ?  

Re: [Asterisk-Users] smallest phone

If you do that, you'll have to carry around a wireless access point as well. Nathan - Original Message - From: "Miguel Cavazos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, April 23, 2004 10:08 AM Subject: Re: [Asterisk-Users] smallest phone > why not wisip? its size its l

RE: [Asterisk-Users] Cisco phones

>Does anyone have a part number or know of anywhere in the UK that resells >the image or the license or both? Matt, I have tried www.cisilion.com/ for a price on the license, but so far have not had a reply. These are the only place I have found to sell the license. The support contracts are a

[Asterisk-Users] MP3 encoding of Monitor files

I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that

Re: [Asterisk-Users] smallest phone

> why not wisip? its size its like a regular cellphone and it uses wifi Because it sucks ass? Check the archives for some very valid gripes about the device. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinf

Re: [Asterisk-Users] 3com SIP phone working with asterisk

interesting... did you tried all the function? ie, can you put a call on hold, and more important do blind & supervised transfer? what about the prices? more or less, just to have an idea... tnx, Matteo Il ven, 2004-04-23 alle 17:08, Lisa Xie ha scritto: > Hello everyone, > > I just like to let

Re: [Asterisk-Users] Cisco phones

I have three questions to ask about this:   1)   How do I know if my phone qualifies for a service contrct ?   2)   Where do I buy a service contract from ?   3)   How will Cisco know that I have downloaded a image that I don't have a licence for ?   Thanks, Paul.     -Original Message--

Re: [Asterisk-Users] X100P Answer

On Fri, 23 Apr 2004 14:55:52 +0100, Mark Olliver wrote Hi, I seam to have a problem working out how to get my X100P to answer after 1 ring. Currently it is working fine and connects to the switchboard menu correctly but just does it after 4 rings, which I would prefer if we could reduce. I fo

[Asterisk-Users] SIP to H323 with no joy

Greetings and salutations to all... I'm having a bit of a problem getting a SIP phone (Xten) to call an H323 Cisco ATA-186. Both devices can call into the * and get the demo, voicemail, etc... I'm pretty sure my problem is in my configs as it feels like a stupid error and to prove this to mys

[Asterisk-Users] Info abaut zaphfc

I'm trying to correct the cid for italy because when arrive a call cid display the number without the initial 0 and when i want to redial the missed call i can't because the number is wrong     Thank's Tiziano

[Asterisk-Users] CLI command

Hi all,   Here is a simple question. How can I know if a call is in pass-thru mode, i.e. * is not in the media path???   Thanks.   Ben

Re: [Asterisk-Users] smallest phone

why not wisip? its size its like a regular cellphone and it uses wifi Miguel Cavazos On Fri, 2004-04-23 at 08:00, Chris Hirsch wrote: > Tim Sailer wrote: > > >Folks, > > I'm looking for a SIP or IAX phone for field techs to take with them > >when out on service calls. The regular desktop phones

[Asterisk-Users] 3com SIP phone working with asterisk

Hello everyone, I just like to let you know that I tested Asterisk with 3COM SIP phones and it worked fine. The 3Com phones are old ones with the same look of NBX 2102 phone but different product number: P/N: 655005001 Rev B There is no special set up except that I have to specifically put allo

Re: [Asterisk-Users] Cisco phones

Does anyone have a part number or know of anywhere in the UK that resells the image or the license or both? > On Fri, 2004-04-23 at 03:12, Paul Tyreman wrote: > > Why is there such a variation in price between what the two of you > > have paid to get the SIP image for a 7960 phone ? $8 would be

Re: Off Topic: RE: [Asterisk-Users] :)

On Fri, Apr 23, 2004 at 09:11:48AM +0200, Dave Cotton said: > On Fri, 2004-04-23 at 08:43 +0200, Florian Overkamp wrote: > > Thanks to this message where a virus chose to use my from-address to send > > its crap from I am now being harassed with many many virus warning messages. > > > > A call to

Re: [Asterisk-Users] X100P Answer

On Fri, 23 Apr 2004, Mark Olliver wrote: > I seam to have a problem working out how to get my X100P to answer after > 1 ring. Currently it is working fine and connects to the switchboard > menu correctly but just does it after 4 rings, which I would prefer if > we could reduce. Try this: zapata.c

Re: [Asterisk-Users] chan_capi

Andrea, Here is a little patch for compiling chan_capi.0.3.1 with latest asterisk CVS. I could read in the lists that a new chan_capi.0.3.2 will soon arrive. In the wait time you can use this patch. put the patch in the chan_capi directory and tip: # patch -p1 < patch.chan_capi-against-0.3.1.di

Re: [Asterisk-Users] Cisco phones

On Fri, 23 Apr 2004, Paul Tyreman wrote: > So are you telling me that to be legal, I need to pay $105, but could > get away with $8 ? *IF* your phone qualifies for service contract (which is US$ 8), yes. You still will have an illegal copy, and you can also be charged later for all the software yo

Re: [Asterisk-Users] Cisco phones

On Fri, 2004-04-23 at 09:11, Paul Tyreman wrote: > If the $8 service contract only gives you access to the image, but you > aren't really allowed to use it, then why do Cisco offer that contact > in the first place ? Support contracts give you access to all Cisco firmware. > So are you telling me

Re: [Asterisk-Users] X100P Answer

On Fri, 23 Apr 2004 14:55:52 +0100, Mark Olliver wrote > Hi, > > I seam to have a problem working out how to get my X100P to answer > after 1 ring. Currently it is working fine and connects to the > switchboard menu correctly but just does it after 4 rings, which I > would prefer if we could re

Re: [Asterisk-Users] X100P Answer

in the [context] set in zaptel.conf ; exten => 6165551212,1,NoOp exten => 6165551212,2,Wait,2; seconds to wait before pickup exten => 6165551212,3,Answer ; On Fri, 23 Apr 2004, Mark Olliver wrote: > > Hi, > > I seam to have a problem working out how to get my X100P to answer after > 1 ring.

Re: [Asterisk-Users] smallest phone

On Fri, Apr 23, 2004 at 08:37:42AM -0500, Eric Wieling wrote: > On Fri, 2004-04-23 at 00:39, James H. Thompson wrote: > > A standard butt set (e.g. http://www.sandman.com/pdf/page81.pdf) combined with a > > Grandstream (very > > small) or Sipura ATA would make a pretty small combination and be use

Re: [Asterisk-Users] zaphfc

Try with : channel => 1-2 Regards, At 11:40 20/04/2004, you wrote: Hello, Here it goes: zaptel.conf: --- span=1,1,3,ccs,ami bchan=1-2 dchan=3 --- zapata.conf --- switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local echocancel=yes immediate=yes group = 1 context=local c

Re: [Asterisk-Users] Cisco phones

If the $8 service contract only gives you access to the image, but you aren't really allowed to use it, then why do Cisco offer that contact in the first place ?   So are you telling me that to be legal, I need to pay $105, but could get away with $8 ?         -Original Message-From:

Re: [Asterisk-Users] Cisco phones

On Fri, 23 Apr 2004, Paul Tyreman wrote: > What website do I have to go to in order to buy a SIP image update ? When I bought mine, I did a Google search on their part number: SW-SM-UL-7960 (Cisco SIP license for 7960 IP Phone) Also, read this message: http://lists.digium.com/pipermail/asterisk-u

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