[Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem
Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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] Polycom IP 500 Phones - Button Assignment
Hello All, So far I have been unable to get the hard button labeled Voice Mail to conenct to Asterisk. I have followed all the Admin Guide instructions regarding the .cfg files and using up.bypassInstantMessage="1" up. to no avail. Has anyone been able to get a Polycom 500 to use the hardbutton to retrieve voice mail? ^Thanks, Wiley
Re: [Asterisk-Users] spa-3000 review?
Wolfgang S. Rupprecht wrote: Interesting. I'm at -current +/- a day and do see a NAK/retry-with-md5 exchange when I do a sip debug. The md5 authentication is also NAK-ed. Well you got farther than I got when I was having problems. :) My fear was that it was expecting the calling user to use their own username in the validation instead of asterisk using the shared secret with a shared user-id. Asterisk should use whatever credentials you define as HTTP Username/Password in the SPA-3000 configuration. -- PhoneBoy ___ 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] Re: MYSQL_FRIENDS and IAX problem
It seems that way, I asked the same question about a month ago, and no one cared to answer. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 07:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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
[Asterisk-Users] Asterisk NAT spa-2000
Hi All, I have a asterisk box that is now on its own static address on the net.it was originally behind a nat firewall. The problem I have is that the remote SPA-2000's that are behind nat firewalls now fail. here is relevent sip.con entry [2001] type=friend username=2001 host=dynamic defaultip=81.178.77.67 allow=ulaw dtmfmode=rfc2833 [EMAIL PROTECTED] context=sip callerid=James 2001 secret=hidden canreinvite=no allow=ulaw nat=yes qualify=yes I added the nat and qualify entries after hunting round google but still get this error, spot the no nat bit. to 81.178.77.67:34504 Retransmitting #2 (no NAT): OPTIONS sip:81.178.77.67:34504 SIP/2.0 Via: SIP/2.0/UDP 62.188.201.123:5060;branch=z9hG4bK68af34fa From: asterisk sip:[EMAIL PROTECTED];tag=as5582cfae To: sip:81.178.77.67:34504 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 18 Jul 2004 12:43:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 any ideas anyone thanks in advance Simon ___ 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] sent into invalid extension 's'
Hi,
On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new
Numbers. I changed the extensions in extension conf to match the new numbers. But i
always get:
Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel
'CAPI[contr1/89064934]/0' sent into invalid extension 's' in context 'default', but no
invalid handler
I only changed the MSNs in the extension.conf. It has worked with the old numbers from
German Telekom.
Any help?
Tom
[makeit]
exten = 932,1,Answer
exten = 932,2,Wait(1)
exten = 932,3,Background(own/wbebuz)
exten = 932,4,Queue(noc24)
exten = 933,1,Answer
exten = 933,2,Wait(1)
exten = 933,3,Background(own/wbebuz)
exten = 933,4,Queue(ebuz)
exten = 934,1,Answer
exten = 934,2,Wait(1)
exten = 934,3,Background(own/wblimtec)
exten = 934,4,Queue(limtec)
[nocnummern]
exten = 89064932,1,Goto,makeit|932|1
exten = 89064933,1,Goto,makeit|933|1
exten = 89064934,1,Goto,makeit|934|1
[default]
include = nocnummern
;exten = s,1,Answer
;exten = s,2,Background(own/tomnoc24)
;exten = s,3,Queue(noc24)
;exten = s,4,Hangup
exten = 2100,1,Dial(SIP/[EMAIL PROTECTED],60,Ttr)
exten = 2200,1,Dial(SIP/[EMAIL PROTECTED],60,Ttr)
;Wählen
exten = _99.,1,Dial(CAPI:${EXTEN:1},20,r)
exten = _99.,2,Playback(invalid)
exten = _99.,3,Hangup
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[Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All Total noob on the list so all help appreciated I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly... but I cannot get the phones to dial each other :( Initially I was getting a extension not found in local message (when dialling from console...from phone just engaged (busy) tone. when I add extension from console I now get a not found 404 messageI see that there was an earlier thread on the list that discussed removing the proxy forwarding from the phone settings and I've tried that from SIPDefault.cnf but it doesn't fix the problem. I've obviously missed something but am too inexperienced to spot it. P my files are as follows:- sipxx.cnf # Lounge Phone Settings # Line 1 Settings line1_name: 11; Line 1 Extension\User ID line1_displayname: Lounge1; Line 1 Display Name line1_authname: lounge11 ; Line 1 Registration Authentication line1_password: lounge; Line 1 Registration Password - sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: 137.222.10.60 ; SNTP Server IP Address sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: GMT ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when BST is in effect dst_start_month: April ; Month in which BST starts dst_start_day: 21 ; Day of month in which BST starts dst_start_day_of_week: Sun ; Day of week in which BST starts dst_start_week_of_month: 1 ; Week of month in which BST starts dst_start_time: 02 ; Time of day in which BST starts dst_stop_month: Oct ; Month in which BST stops dst_stop_day: 20 ; Day of month in which BST stops dst_stop_day_of_week: Sunday ; Day of week in which BST stops dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week of month dst_stop_time: 2 ; Time of day in which BST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic adjustment time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no user control) callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 proxy_backup: ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) proxy_emergency: ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled outbound_proxy: ; restricted to dotted IP or DNS A record only outbound_proxy_port: 5060 ; default is 5060 # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the
RE: [Asterisk-Users] Video/H323/SIP
Hi, -Original Message- MSN messenger 4.7 with any windows capturing device should work. Make sure you force the codecs properly, because MSN tries to negotiate some form of MJPEG which Asterisk doesn't support. How do you force the codecs? Do you do this in Messenger or Asterisk? Right now I have set videosupport=yes and allowed h261 and h261 in sip.conf. Are there any other settings I need to change? No, that should do. Make sure that you DO NOT SET 'allow=all' for your codecs, or MSN will try the wrong codecs. Florian ___ 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] Using Windows Messenger+Video in *
Hi, -Original Message- This is a little brief to say. I have had this working properly with recent asterisk boxes. A few things: Check if the [general] section has 'videosupport=yes' and if the sip peers are allowed to use h261 and h263 codecs. Best regards, Florian Do you think you could post your relevant .conf files? Is sip.conf the only one affected? Sip.conf is the only one affected for SIP-to-SIP calls. Iax.conf is also relevant if you want to use IAX links. Here is what I have in one of my setups: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = from-sip ; Default for local connections (which has no access) videosupport=yes [video2] type=friend username=video2 secret=hidden host=dynamic context=from-werkkamer callerid=Video 2 1222 canreinvite=no disallow=all allow=ulaw allow=alaw allow=speex allow=gsm allow=h261 allow=h263 ___ 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] Hotline
Hello There, I tried checking out for this feature , what i want to do is that as soon as the user picks up the handset , * waits for 10 secs and then dials a predefined number , its like the HOTLINE feature we have in normal POTs . Is it possible with Asterisk? If yes then how? Regards ~uppal ___ 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] Hotline
On Sunday 18 July 2004 09:36, Junaid Uppal wrote:
I tried checking out for this feature , what i want to do is that as
soon as the user picks up the handset , * waits for 10 secs and then
dials a predefined number , its like the HOTLINE feature we have in
normal POTs . Is it possible with Asterisk? If yes then how?
use immediate=yes in zapata.conf on the channel you want to be a hotline, and
then in the defined context something like
exten = s,1,Wait(10)
exten = s,2,Dial(${BATMAN},,T)
Read up on immediate mode and the dial command if you need more info.
-A.
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Re: [Asterisk-Users] Hotline
it can also be defined on some devices like the grandstreams.
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 9:42 AM
Subject: Re: [Asterisk-Users] Hotline
On Sunday 18 July 2004 09:36, Junaid Uppal wrote:
I tried checking out for this feature , what i want to do is that as
soon as the user picks up the handset , * waits for 10 secs and then
dials a predefined number , its like the HOTLINE feature we have in
normal POTs . Is it possible with Asterisk? If yes then how?
use immediate=yes in zapata.conf on the channel you want to be a hotline,
and
then in the defined context something like
exten = s,1,Wait(10)
exten = s,2,Dial(${BATMAN},,T)
Read up on immediate mode and the dial command if you need more info.
-A.
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[Asterisk-Users] Asterisk and zaptel on Fedora Core 2
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dear all. As I couldn't get to compile and run Asterisk 1.0RC1 on my default RedHat 9 I thought it was about time to upgrade to Fedora Core 2. Well, it was too late to realize the kernel 2.6 wasn't supported by Asterisk *officially* anyway. Here is what I did to get asterisk and zaptel to work under Fedora Core 2: I posted it on the wiki and here is an extract Getting asterisk to work on fedora core 2 is no problem. But getting zaptel to work is another issue. The kernel (2.6.5) source code provided with Fedora Core 2 is missing some auto-generated components. I found that the easiest way to get around all those issues was to download a new kernel source code like 2.6.7 from www.kernel.org. Here is the procedure: 1-Grab the 2.6.7 kernel source code and untar it (do not untar it in /usr/src, this is a very bad practice) 2-Copy the .config file from the default /usr/src/linux-2.6.5-1.358 into the 2.6.7 source code directory. 3-type; make menuconfig and make the necessary change for your hardware configuration. You could just leave it as it is as the default Fedora Core 2 contains everything. But having so much stuff in means much longer compilation time! Quit and save the .config file 4-Compile and install your kernel as describe there: http://www.digitalhermit.com/linux/Kernel-Build-HOWTO.html 5-Create a link linux-2.6 to your 2.6.7 linux kernel directory in /usr/src; something like: ln -s /data/work/src/linux-2.6.7 /usr/src/linux-2.6 6-Reboot with the new kernel 7-Get the latest asterisk, libpri and zaptel source code from the digium CVS directory 8-Go into the zaptel directory and type: make clean make linux26 make install make config 9-Edit the file /etc/init.d/zaptel and replace all: insmod with modprobe and rmmod with modprobe -r That's it. Make sure it works by starting the script /etc/init.d/zaptel start doing lsmod should show the wcfxs and zaptel module being installed. then install and run asterisk as usual. Hope all of this help Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+o0ZXeDVKqIr3GURAtHeAJsHbLo6Ty6TKNrhoFF7uvkSUuR9XQCeJ9Lr tmITAmxkoVHkVIS/uxOsEPw= =qJZf -END PGP SIGNATURE- ___ 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] Parking renamed to feature in 7/17/04 CVS
On 17/07/2004 at 20:25 Josh Roberson wrote: Seth Remington wrote: I just updated from CVS and noticed that Mark has renamed all of the parking related files (parking.conf, parking.h, res_parking.c) to features.conf, features.h, res_features.c respectively. The CVS log mentions that this is in preparation for some more (possibly post 1.0) feature additions. The header file still #define(s) _PARKING_H though so let the confusion ensue ;) Time to update the wiki. -Seth Actually, no, that was fixed also. -twisted Excellent! can't you do it so that each time you grab a new version from CVS it uses a random filename for each and every config, just to make sure.. possibly even using the wrong filename for the wrong configs... Andy ___ 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] sent into invalid extension 's'
The reason is in the error message. Try using extension number 89064934
instead of 934.
Chris.
Tom Fischer wrote (on Jul 18):
Hi,
On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved
new Numbers. I changed the extensions in extension conf to match the new numbers.
But i always get:
Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel
'CAPI[contr1/89064934]/0' sent into invalid extension 's' in context 'default', but
no invalid handler
I only changed the MSNs in the extension.conf. It has worked with the old numbers
from German Telekom.
Any help?
Tom
[makeit]
exten = 932,1,Answer
exten = 932,2,Wait(1)
exten = 932,3,Background(own/wbebuz)
exten = 932,4,Queue(noc24)
exten = 933,1,Answer
exten = 933,2,Wait(1)
exten = 933,3,Background(own/wbebuz)
exten = 933,4,Queue(ebuz)
exten = 934,1,Answer
exten = 934,2,Wait(1)
exten = 934,3,Background(own/wblimtec)
exten = 934,4,Queue(limtec)
[nocnummern]
exten = 89064932,1,Goto,makeit|932|1
exten = 89064933,1,Goto,makeit|933|1
exten = 89064934,1,Goto,makeit|934|1
[default]
include = nocnummern
;exten = s,1,Answer
;exten = s,2,Background(own/tomnoc24)
;exten = s,3,Queue(noc24)
;exten = s,4,Hangup
exten = 2100,1,Dial(SIP/[EMAIL PROTECTED],60,Ttr)
exten = 2200,1,Dial(SIP/[EMAIL PROTECTED],60,Ttr)
;W?hlen
exten = _99.,1,Dial(CAPI:${EXTEN:1},20,r)
exten = _99.,2,Playback(invalid)
exten = _99.,3,Hangup
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== [EMAIL PROTECTED]
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[Asterisk-Users] PhoneGaim?
I say on slashdot that the Linspire guys have released PhoneGaim. PhoneGaim is Gaim with SIP added on. Anyone want to add IAX2 as well... http://www.phonegaim.com/faq.html ___ 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] PSTN Gateway X101P
I am trying to setup a simple pstn gateway using Asterisk and a X100p card.
I have got everything installed using Redhat 9 and am able to load Asterisk.
I also configured sip and I am able to connect to the asterisk gateway with
Xlite on the windows side.
I am able to dial 1000 and get the welcome message.
What I am NOT able to do is dial a seven digit local or 10 digit long distance
number and make a phone call to the pstn using the x100p card.
I configured the zaptel.conf and zapta.conf files and when I do ztcfg -v
I get:
[EMAIL PROTECTED] asterisk]# ztcfg -v
Zaptel Configuration
==
1 channels configured.
It appears that I have the driver loaded correctly.
I edited the sample extensions.conf and changed the varible trunk to zap/1
Attached is my extensions.conf
When I dial 94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would be appreciated.
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the include command that includes contexts within
; other contexts. The #include command works in all asterisk configuration files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/1 ; Trunk interface
OUTGOING = Zap/1
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than General and Globals represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
;[context]
;exten = someexten,priority,application(arg1,arg2,...)
;exten = someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
; time range|days of week|days of month|months
;
;include = daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern. The most commonly used example is
; of course '9' like this:
;
;ignorepat = 9
;
; so that dialtone remains even after dialing a 9.
;
;
; Here are the entries you need to participate in the IAXTEL
; call routing system. Most IAXTEL numbers begin with 1-700, but
; there are exceptions. For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten = _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
;
; The SWITCH statement permits a server to share the dialplain with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A - B and B - A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch = IAX2/user:[EMAIL PROTECTED]/mycontext
[trunkint]
;
; International long distance through trunk
;
exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten =
[Asterisk-Users] quadbri NT_mode S-Bus Problem
I am running * with a Junghanns quadbri that should
allow us to integrate our ISDN house telephone system with VOIP. Preferably I
would like to run a setup, so that our internal ISDN phones on an S bus are not
aware that * is sitting in between.
With the configuration below I run into the
following problems:
1. On outbound calls, I get the normal rining call
progress tone althought the the other party has not even been reached. This
then changes from normal ringing suddenly to busy when the other party is
sending a busy signal. I'd rather have the call progress send a busy signal
right away.
2. Internal calls between to ISDN phones on the
S-bus is not possible. The phone rings but the call is dropped as soon as it is
answered.
Can the signalling= bri_net_ptmp be the cause and
how would I configure it for bri_net?
Does anyone have a working configuration that
overcomes thoses problems?
Regards,
Ben
; Zapata telephony interface
;
; Configuration file
[channels]
switchtype = euroisdn
overlapdial=no
echocancel=yes
echocancelwhenbridged=yes
pridialplan = unknown
prilocaldialplan = local
context=isdn-in
group = 1
signalling = bri_cpe_ptmp
channel = 1-2
context=local
signalling = bri_net_ptmp
group = 3
channel = 4-5
;
; extensions.conf
;
[local]
include = parkedcalls
include = ntout
include = conference
exten =
903,1,Dial(Zap/g2/9771762)
exten =
904,1,Dial(Zap/g2/9771707)
[ntout]
exten = s,1,DigitTimeout,3
exten = s,2,ResponseTimeout,5
exten =
_X.,1,Dial(Zap/g1/${EXTEN},,r)
exten = _X.,2,Congestion
[isdn-in]
exten =
9771762,1,Dial(Zap/g2/9771762)
exten =
9771707,1,Dial(Zap/g2/9771707)
RE: [Asterisk-Users] voicemail broadcast feature
Yes. I pulled the latest cvs and no cc in there at all anywhere. but the bug http://bugs.digium.com/bug_view_page.php?bug_id=0001361 shows that it is in fact committed to cvs. I cannot seem to reopen this bug to say that it is not really committed. I guess I will open a new bug report to do this. I believe we have a case of something being added to the wiki before it was added to the code :P If you peek into the apply_options() function in the app_voicemail.c file (updated today 7/17/04) it doesn't even check for the cc option. Options that are handled are: attach, serveremail, language, tz, delete, saycid, review, operator, envelope, callback, dialout, and exitcontext... but unfortunately NO cc. -Seth On Sat, 2004-07-17 at 16:21, Frank wrote: Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 = 1234,BroadCast Test,,,[EMAIL PROTECTED] . then many other voicemail boxes. - whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with cc=xxx, the vmail box does not get the cc. Got this right off the wiki. Hat am I missing? ___ 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] PSTN Gateway X101P
try puttin this in extensions.conf
[outgoing]
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
and into your siphones extensions definition
[sip]
include = outgoing
Adrià Vidal
[EMAIL PROTECTED] | http://adria.homeip.net | MSN
[EMAIL PROTECTED]
iChat [EMAIL PROTECTED] | FWD [EMAIL PROTECTED] | IAXTEL 1700 337 68
48
On Jul 18, 2004, at 5:12 PM, Jason Armentrout wrote:
1 channels configured.
It appears that I have the driver loaded correctly.
I edited the sample extensions.conf and changed the varible trunk to
zap/1
Attached is my extensions.conf
When I dial 94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would be appreciated.
___
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Re: [Asterisk-Users] PSTN Gateway X101P
I added
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
to the extensions.conf
but I am not sure I follow you on the second part, do you want me to add
include = outgoing
to my sip.conf file?? I did both of these changes, and I still have the same
problem.
Quoting Adria Vidal [EMAIL PROTECTED]:
try puttin this in extensions.conf
[outgoing]
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
and into your siphones extensions definition
[sip]
include = outgoing
Adrià Vidal
[EMAIL PROTECTED] | http://adria.homeip.net | MSN
[EMAIL PROTECTED]
iChat [EMAIL PROTECTED] | FWD [EMAIL PROTECTED] | IAXTEL 1700 337 68
48
On Jul 18, 2004, at 5:12 PM, Jason Armentrout wrote:
1 channels configured.
It appears that I have the driver loaded correctly.
I edited the sample extensions.conf and changed the varible trunk to
zap/1
Attached is my extensions.conf
When I dial 94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would be appreciated.
___
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Re: [Asterisk-Users] 7960 Dynamic DNS?
There are many dyn dns clients for Windoze availible and some for linux based computers. A few SOHO NAT routers support this also, but they are limited in scope and may not work for your situation. I think a workstation based solution is what you need if your router does not support it. Lyle - Original Message - From: Marty Mastera [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 16, 2004 8:15 PM Subject: RE: [Asterisk-Users] 7960 Dynamic DNS? snip Does anyone have any ideas on how to accomplish a dynamic dns registration without relying on a PC to do it? My router (Dell TrueMobile 2300) doesn't seem to offer this feature either. Marty ___ 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] sip-oh323
HI ALL; I have couple of ip phones connected to my asterisk box 1-cisco ata with sip protocol 2-sjphone with h323 protocol as I understand, asterisk isable to translate siph323 and vice versa ( am I right)???/ but when I try to connectfrom ATA toSJPHONE and vice versa it fails. plz help to find out more and appreciate an example config warmest regards mohammad
[Asterisk-Users] New G.729 codec and VLANS
The readme says that the license uses all network cards MACS What happens when VLANS are added or removed? Is it safe? ___ 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] Asterisk NAT spa-2000
I would comment out these lines in sip.conf ;externip=111.222.333.444 ;localnet=192.168.1.0 ;localmask=255.255.255.0 Then set nat=no -Original Message- From: Simon Chappell [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004 4:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk NAT spa-2000 Hi All, I have a asterisk box that is now on its own static address on the net.it was originally behind a nat firewall. The problem I have is that the remote SPA-2000's that are behind nat firewalls now fail. here is relevent sip.con entry [2001] type=friend username=2001 host=dynamic defaultip=81.178.77.67 allow=ulaw dtmfmode=rfc2833 [EMAIL PROTECTED] context=sip callerid=James 2001 secret=hidden canreinvite=no allow=ulaw nat=yes qualify=yes I added the nat and qualify entries after hunting round google but still get this error, spot the no nat bit. to 81.178.77.67:34504 Retransmitting #2 (no NAT): OPTIONS sip:81.178.77.67:34504 SIP/2.0 Via: SIP/2.0/UDP 62.188.201.123:5060;branch=z9hG4bK68af34fa From: asterisk sip:[EMAIL PROTECTED];tag=as5582cfae To: sip:81.178.77.67:34504 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 18 Jul 2004 12:43:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 any ideas anyone thanks in advance Simon ___ 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] PSTN Gateway X101P
On Jul 18, 2004, at 5:56 PM, Jason Armentrout wrote: to the extensions.conf but I am not sure I follow you on the second part, do you want me to add include = outgoing to my sip.conf file?? I did both of these changes, and I still have the same problem. must add include = outgoing into your extensions.conf file where the sip extensions are defined example [sip] ; include = fwd include = iaxtel include = stanaphone include = SIPphone include = fromiaxfwd include = from-iaxtel include = stana-incoming include = parkedcalls include = outgoing exten = 100,1,Dial(SIP/100,20,tr) exten = 100,2,Voicemail,100 exten = 100,3,Hangup Adrià Vidal [EMAIL PROTECTED] | http://adria.homeip.net | MSN [EMAIL PROTECTED] iChat [EMAIL PROTECTED] | FWD [EMAIL PROTECTED] | IAXTEL 1700 337 68 48 ___ 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] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
I just started out too and I can tell you it is easier to start from scratch with a good wiki then alter the demo files. Here is a wiki you can build a good working system with... http://www.wlug.org.nz/AsteriskSampleSetup For your ciscos search http://asterisk.xvoip.com/index.php Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004 5:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk Hi All Total noob on the list so all help appreciated I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly... but I cannot get the phones to dial each other :( Initially I was getting a extension not found in local message (when dialling from console...from phone just engaged (busy) tone. when I add extension from console I now get a not found 404 messageI see that there was an earlier thread on the list that discussed removing the proxy forwarding from the phone settings and I've tried that from SIPDefault.cnf but it doesn't fix the problem. I've obviously missed something but am too inexperienced to spot it. P my files are as follows:- sipxx.cnf # Lounge Phone Settings # Line 1 Settings line1_name: 11; Line 1 Extension\User ID line1_displayname: Lounge1; Line 1 Display Name line1_authname: lounge11 ; Line 1 Registration Authentication line1_password: lounge; Line 1 Registration Password - sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: 137.222.10.60 ; SNTP Server IP Address sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: GMT ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when BST is in effect dst_start_month: April ; Month in which BST starts dst_start_day: 21 ; Day of month in which BST starts dst_start_day_of_week: Sun ; Day of week in which BST starts dst_start_week_of_month: 1 ; Week of month in which BST starts dst_start_time: 02 ; Time of day in which BST starts dst_stop_month: Oct ; Month in which BST stops dst_stop_day: 20 ; Day of month in which BST stops dst_stop_day_of_week: Sunday ; Day of week in which BST stops dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week of month dst_stop_time: 2 ; Time of day in which BST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic adjustment time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no user control) callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 proxy_backup: ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) proxy_emergency: ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media
Re: [Asterisk-Users] Wo uses H323-phones with asterisk?
On Sat, Jul 17, 2004 at 10:35:58AM +0200, Christian Ekhart wrote: Hi, we successfully use innovaphone IP200 H.323 hardware phones with OH323/Asterisk. Calling/talking is OK, but call transfer does not work. Does anyone of you use H323-phones with asterisk AND IS ABLE to perform CALL TRANFERS?! Hello! A few months ago I tried to get an IP-200 to work with *. I had to use GnuGk where the IP-200 and * could register to. When using the R button on the phone to dial another call * would crash. So I am curious what versions of * and OH323 are you using? I am also interested in the configs. Maybe I have overlooked something back then... -Walter ___ 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] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
It doesn't look like you have a context set for phone1. Try putting context=sip in the phone1 section like you have in phone2. That'll put both in the same context of your extensions.conf file and should allow interaction between the two. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 7:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk Hi All Total noob on the list so all help appreciated I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly... but I cannot get the phones to dial each other :( Initially I was getting a extension not found in local message (when dialling from console...from phone just engaged (busy) tone. when I add extension from console I now get a not found 404 messageI see that there was an earlier thread on the list that discussed removing the proxy forwarding from the phone settings and I've tried that from SIPDefault.cnf but it doesn't fix the problem. I've obviously missed something but am too inexperienced to spot it. P my files are as follows:- sipxx.cnf # Lounge Phone Settings # Line 1 Settings line1_name: 11; Line 1 Extension\User ID line1_displayname: Lounge1; Line 1 Display Name line1_authname: lounge11 ; Line 1 Registration Authentication line1_password: lounge; Line 1 Registration Password - sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: 137.222.10.60 ; SNTP Server IP Address sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: GMT ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when BST is in effect dst_start_month: April ; Month in which BST starts dst_start_day: 21 ; Day of month in which BST starts dst_start_day_of_week: Sun ; Day of week in which BST starts dst_start_week_of_month: 1 ; Week of month in which BST starts dst_start_time: 02 ; Time of day in which BST starts dst_stop_month: Oct ; Month in which BST stops dst_stop_day: 20 ; Day of month in which BST stops dst_stop_day_of_week: Sunday ; Day of week in which BST stops dst_stop_week_of_month: 8 ; Week of month in which BST stops 8=last week of month dst_stop_time: 2 ; Time of day in which BST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) BST automatic adjustment time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) dnd_control: 0 ; Default 0 (0=off, 1=on, 2=off no user cntrl, 3=on no user control) callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 proxy_backup: ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) proxy_emergency: ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766)
[Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem
hmm - this is the bad thing about open source etc. Should we make a bugreport ? or are we just doing something wrong ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- usedcanon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] It seems that way, I asked the same question about a month ago, and no one cared to answer. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 07:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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 ___ 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] PSTN Gateway X101P
What I am NOT able to do is dial a seven digit local or 10 digit long distance number and make a phone call to the pstn using the x100p card. snip Attached is my extensions.conf When I dial 94341321 or 4341321 I just get a 404 error in Xlite. What am I doing wrong? Any help would be appreciated. Hey Jason In your extensions.conf, the [default] context only has the [demo] context included which provides no outbound dialing. Try adding an 'include =' line to your default context to allow for this. For example in extensions.conf, there is a context called [local] to allow for outbound dialing, so add 'include = local' under your [default] context... The other side of this is in sip.conf, where you tell the phone (or x-lite or whatever) which context to start in (from extensions.conf). Since you can already dial 1000 and get the demo, I assume that your sip.conf is configured to start in the [default] context in extensions.conf With that being the case, after adding the include = local to your [default] context, you should be able to dial your 7 digit number (you must dial 9 first). Marty ___ 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] New G.729 codec and VLANS
Anton Tinchev wrote: The readme says that the license uses all network cards MACS What happens when VLANS are added or removed? Is it safe? Also, in this day of motherboard-integrated NICs (even two or three), what will happen if the mobo dies and has to be replaced? ___ 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] quadbri NT_mode S-Bus Problem
What type is your ISDN house telephone system?
Without more specific information all we can do is guess...
For a sollution to 1 ... drop the r option of dial...
exten = _X.,1,Dial(Zap/g1/${EXTEN})
You might need pridialplan/prilocaldialplan set to local for local
calls... or both to unknown... just experiment with those values.
Regards
Ben Bosshardt wrote:
I am running * with a Junghanns quadbri that should allow us to
integrate our ISDN house telephone system with VOIP. Preferably I
would like to run a setup, so that our internal ISDN phones on an S
bus are not aware that * is sitting in between.
With the configuration below I run into the following problems:
1. On outbound calls, I get the normal rining call progress tone
althought the the other party has not even been reached. This then
changes from normal ringing suddenly to busy when the other party is
sending a busy signal. I'd rather have the call progress send a busy
signal right away.
2. Internal calls between to ISDN phones on the S-bus is not possible.
The phone rings but the call is dropped as soon as it is answered.
Can the signalling= bri_net_ptmp be the cause and how would I
configure it for bri_net?
Does anyone have a working configuration that overcomes thoses problems?
Regards,
Ben
; Zapata telephony interface
;
; Configuration file
[channels]
switchtype = euroisdn
overlapdial=no
echocancel=yes
echocancelwhenbridged=yes
pridialplan = unknown
prilocaldialplan = local
context=isdn-in
group = 1
signalling = bri_cpe_ptmp
channel = 1-2
context=local
signalling = bri_net_ptmp
group = 3
channel = 4-5
;
; extensions.conf
;
[local]
include = parkedcalls
include = ntout
include = conference
exten = 903,1,Dial(Zap/g2/9771762)
exten = 904,1,Dial(Zap/g2/9771707)
[ntout]
exten = s,1,DigitTimeout,3
exten = s,2,ResponseTimeout,5
exten = _X.,1,Dial(Zap/g1/${EXTEN},,r)
exten = _X.,2,Congestion
[isdn-in]
exten = 9771762,1,Dial(Zap/g2/9771762)
exten = 9771707,1,Dial(Zap/g2/9771707)
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[Asterisk-Users] sip-h323
hi all; hi DANIEL; I setup asterisk as a translator between sip-h323(I used oh323 not native). But there is a problem and it is as follows: whenI try to dial FIRST from sip UA to h323 client, or h323 client to sip UA , it is ok BUT the second try from any of them to another have no audio. any suggestion Regards
RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi Sean Both phones are set for context=sip in the sip.conf file. As I say the phones will both call out OK (I can dial the 500 test number and successfully connect to the remote PBX through my firewall). It's just that when I'm trying to call from phone to phone I'm getting the 404 not found error in the asteris verbose dialog. If anyone has a documented example of their 7960 config sipdefault.cnf and sipxipadd.cnf files together with their sip.conf and extensions.conf files I could have to test directly on my system I'd be appreciative to test them on my system. While the WiKi's are very useful as example files it would be great (and I may do it myself!!) if there was an up to date example file with all the options for each filed and a verbose description for the rational behind it (although I recognise that this is an 'in development' product and therefore the docs have to be done at the end!!). Part of the problem is there are so many dependencies that can affect the system including how the dhpcd server serves IP address's and associated files (for example the files have to be structured in a particular order on the tftpd server for the cisco's to pick them up correctly). Given this level of dependency I'm not sure where the break could be. The one thing I have noticed from the show sip peers field is that it's showing the phones as having a netmask of 255.255.255.255 although they're actually configyred for 255.255.255.0. P -Original Message- From: Sean Cheesman [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004, 11:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk It doesn't look like you have a context set for phone1. Try putting context=sip in the phone1 section like you have in phone2. That'll put both in the same context of your extensions.conf file and should allow interaction between the two. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 7:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk Hi All Total noob on the list so all help appreciated I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly... but I cannot get the phones to dial each other :( Initially I was getting a extension not found in local message (when dialling from console...from phone just engaged (busy) tone. when I add extension from console I now get a not found 404 messageI see that there was an earlier thread on the list that discussed removing the proxy forwarding from the phone settings and I've tried that from SIPDefault.cnf but it doesn't fix the problem. I've obviously missed something but am too inexperienced to spot it. P my files are as follows:- sipxx.cnf # Lounge Phone Settings # Line 1 Settings line1_name: 11 ; Line 1 Extension\User ID line1_displayname: Lounge1 ; Line 1 Display Name line1_authname: lounge11; Line 1 Registration Authentication line1_password: lounge ; Line 1 Registration Password - sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: 137.222.10.60 ; SNTP Server IP Address sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default)
Re: [Asterisk-Users] 7960 Dynamic DNS?
I can't think of any router that supports this You could put it in as a request to www.sveasoft.com for their firmware for the wrt54g (great box...runs linux and lots of features and functionality). P -Original Message- From: Lyle Giese [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004, 9:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7960 Dynamic DNS? There are many dyn dns clients for Windoze availible and some for linux based computers. A few SOHO NAT routers support this also, but they are limited in scope and may not work for your situation. I think a workstation based solution is what you need if your router does not support it. Lyle - Original Message - From: Marty Mastera [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 16, 2004 8:15 PM Subject: RE: [Asterisk-Users] 7960 Dynamic DNS? snip Does anyone have any ideas on how to accomplish a dynamic dns registration without relying on a PC to do it? My router (Dell TrueMobile 2300) doesn't seem to offer this feature either. Marty ___ 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] New G.729 codec and VLANS
You can re-register the codecs one time using other NICS. after that one time you need to contact Digium to be able to re-register, but the process is very easy! At 21:33 18.07.2004, you wrote: Anton Tinchev wrote: The readme says that the license uses all network cards MACS What happens when VLANS are added or removed? Is it safe? Also, in this day of motherboard-integrated NICs (even two or three), what will happen if the mobo dies and has to be replaced? ___ 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] Help! Unable to create channel of type SIP.
I have a SIP phone that can make calls but can't recieve calls. Can anyone suggest why?
sip show peers:
Name/username Host Dyn Nat ACL Mask Port Status 601/601 (Unspecified) D N 255.255.255.255 0 UNKNOWN
-- Executing Dial("SIP/206.132.91.139-0814c9f8", "SIP/601|20|r") in new stackJul 18 15:50:18 NOTICE[638991]: app_dial.c:689 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
RE: [Asterisk-Users] PSTN Gateway X101P
Thanks for the tip, that made things work, it is really difficult for me to understand the different config files and especially the extensions.conf, it is very confusing. I am trying to learn though. Now that I have got outgoing calls to work from the sip phone. How can I route incoming calls on the pstn line (x100p) to the sip phone? Thanks! Quoting Marty Mastera [EMAIL PROTECTED]: What I am NOT able to do is dial a seven digit local or 10 digit long distance number and make a phone call to the pstn using the x100p card. snip Attached is my extensions.conf When I dial 94341321 or 4341321 I just get a 404 error in Xlite. What am I doing wrong? Any help would be appreciated. Hey Jason In your extensions.conf, the [default] context only has the [demo] context included which provides no outbound dialing. Try adding an 'include =' line to your default context to allow for this. For example in extensions.conf, there is a context called [local] to allow for outbound dialing, so add 'include = local' under your [default] context... The other side of this is in sip.conf, where you tell the phone (or x-lite or whatever) which context to start in (from extensions.conf). Since you can already dial 1000 and get the demo, I assume that your sip.conf is configured to start in the [default] context in extensions.conf With that being the case, after adding the include = local to your [default] context, you should be able to dial your 7 digit number (you must dial 9 first). Marty ___ 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] PSTN Gateway X101P
Thanks for the tip, that made things work, it is really difficult for me to understand the different config files and especially the extensions.conf, it is very confusing. I am trying to learn though. Now that I have got outgoing calls to work from the sip phone. How can I route incoming calls on the pstn line (x100p) to the sip phone? Thanks! First, I would dial the telephone number of the line plugged into the X101P and make sure that the demo answers to verify that things are working correctly...assuming that works, you just need to modify your extensions.conf a little bit... Your [default] context includes [demo] which has an answer line in it, followed by the rest of the items necessary to playback the demo. So if you want an incoming call to ring directly to your x-lite, I would remove the include for [demo] from your [default] context (but leave the include for [local] so that you can make outbound calls!...then inside your [default] context (just below the include for [local] for example) add lines that will answer the phone and ring your x-lite: (note that below, the SIP/1000 is just an example...the '1000' should be whatever name you gave your x-lite in sip.conf) exten = s,1,Wait exten = s,2,Answer exten = s,3,Dial(SIP/1000,20,r) Save the changes and reload asterisk, try calling the line connected to the X101P and if your x-lite has registered with asterisk correctly, it should ring there...look on the wiki (www.voip-info.org) for the specific syntax of the Dial command and it's options, also the above is a very basic config, with no timeouts specified, etc...it should work, but should/could be made more robust after you get it working initially. Marty ___ 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] Help! Unable to create channel of type SIP.
Your phone isn't registered. Ie Host
(Unspecified) so it has no idea where to send the call. Set your phone to
register and then asterisk can find it.
bkw
- Original Message -
From:
Joe
Babstock
To: [EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 2:57 PM
Subject: [Asterisk-Users] Help! Unable to
create channel of type SIP.
I have a SIP phone that can make calls but can't recieve calls. Can
anyone suggest why?
sip show peers:
Name/username
Host Dyn Nat
ACL
Mask
Port Status
601/601
(Unspecified) D N
255.255.255.255 0
UNKNOWN
-- Executing Dial("SIP/206.132.91.139-0814c9f8",
"SIP/601|20|r") in new stackJul 18 15:50:18 NOTICE[638991]: app_dial.c:689
dial_exec: Unable to create channel of type 'SIP' == Everyone is
busy/congested at this time
__Do You
Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
RE: [Asterisk-Users] PSTN Gateway X101P
Thanks Marty, That works now, the caller id on Xlite only shows the name for some reason, not the number, but anyway it now rings in. When I call the pstn number, the zaptel picks up the line on the first ring and then forwards it to the sip phone and rings it. Is there anyway to prevent the zaptel from picking up the line until the sip phone actully answers the call. This way I could answer the phone either locally on a regular analog handset or through the sip phone. The way it is now, it only rings my phones in the house 1 time. Jason Quoting Marty Mastera [EMAIL PROTECTED]: Thanks for the tip, that made things work, it is really difficult for me to understand the different config files and especially the extensions.conf, it is very confusing. I am trying to learn though. Now that I have got outgoing calls to work from the sip phone. How can I route incoming calls on the pstn line (x100p) to the sip phone? Thanks! First, I would dial the telephone number of the line plugged into the X101P and make sure that the demo answers to verify that things are working correctly...assuming that works, you just need to modify your extensions.conf a little bit... Your [default] context includes [demo] which has an answer line in it, followed by the rest of the items necessary to playback the demo. So if you want an incoming call to ring directly to your x-lite, I would remove the include for [demo] from your [default] context (but leave the include for [local] so that you can make outbound calls!...then inside your [default] context (just below the include for [local] for example) add lines that will answer the phone and ring your x-lite: (note that below, the SIP/1000 is just an example...the '1000' should be whatever name you gave your x-lite in sip.conf) exten = s,1,Wait exten = s,2,Answer exten = s,3,Dial(SIP/1000,20,r) Save the changes and reload asterisk, try calling the line connected to the X101P and if your x-lite has registered with asterisk correctly, it should ring there...look on the wiki (www.voip-info.org) for the specific syntax of the Dial command and it's options, also the above is a very basic config, with no timeouts specified, etc...it should work, but should/could be made more robust after you get it working initially. Marty ___ 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] 7960 Dynamic DNS?
I had a Netgear WGR614 802.11g Wireless Router for a short time period, it did support automatic dyndns updates, which was very handy. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7960 Dynamic DNS? I can't think of any router that supports this You could put it in as a request to www.sveasoft.com for their firmware for the wrt54g (great box...runs linux and lots of features and functionality). P -Original Message- From: Lyle Giese [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004, 9:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7960 Dynamic DNS? There are many dyn dns clients for Windoze availible and some for linux based computers. A few SOHO NAT routers support this also, but they are limited in scope and may not work for your situation. I think a workstation based solution is what you need if your router does not support it. Lyle - Original Message - From: Marty Mastera [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 16, 2004 8:15 PM Subject: RE: [Asterisk-Users] 7960 Dynamic DNS? snip Does anyone have any ideas on how to accomplish a dynamic dns registration without relying on a PC to do it? My router (Dell TrueMobile 2300) doesn't seem to offer this feature either. Marty ___ 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 ___ 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] quadbri NT_mode S-Bus Problem
What type is your ISDN house telephone system?
Without more specific information all we can do is guess...
Our system is a just the basic subscription to SWISSCOM, which is the main
phone company in Switzerland. We have BRI with 2 Channels which can be used
simulaniously and a Siemens NT that has only the function of feeding our
S-bus with 4 phones connected.
For a sollution to 1 ... drop the r option of dial...
exten = _X.,1,Dial(Zap/g1/${EXTEN})
I will give it a try.
You might need pridialplan/prilocaldialplan set to local for local
calls... or both to unknown... just experiment with those values.
I am still looking for any documentation regarding the use of
pridialplan/prilocaldialplan. I don't know how to find out what SWISSCOM
requires.
Thanks for your help.
Ben
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[Asterisk-Users] Brain-dead Grandstream BT102?
Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] New G.729 codec and VLANS
Also, in this day of motherboard-integrated NICs (even two or three), what will happen if the mobo dies and has to be replaced? The same thing that would happen if the NIC died. IMHO it's a good thing to tie to the NIC, because the chances of the MOBO dieing is not that extreme. If it does die, than just call digium and they'll re-license it. Now it would be nice if when you install the codec, there was a 3 or 4 day period where the license wasn't needed. After the 4 days is up, then it requires a lic. key. This would be useful in the event that your * switch dies on a friday evening. - Brent ___ 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] New G.729 codec and VLANS
Marc Storck wrote: You can re-register the codecs one time using other NICS. after that one time you need to contact Digium to be able to re-register, but the process is very easy! That's good to know, thanks! ___ 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] New G.729 codec and VLANS
Brent Franks wrote: The same thing that would happen if the NIC died. IMHO it's a good thing to tie to the NIC, because the chances of the MOBO dieing is not that extreme. If it does die, than just call digium and they'll re-license it. Now it would be nice if when you install the codec, there was a 3 or 4 day period where the license wasn't needed. After the 4 days is up, then it requires a lic. key. This would be useful in the event that your * switch dies on a friday evening. Well, personally I won't have to worry about that, because my systems will be completely redundant (even redundant G.729 licenses). The hardware and licenses are really not that expensive (a 3GHz P4 server with a T-1 card and 23 G.729 licenses is well under $2000). ___ 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] Re: Re: MYSQL_FRIENDS and IAX problem
Bug report might be a good idea, I just dropped the issue as I could do without using IAX. I am sure others may not have that flexibility. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 19:10 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem hmm - this is the bad thing about open source etc. Should we make a bugreport ? or are we just doing something wrong ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- usedcanon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] It seems that way, I asked the same question about a month ago, and no one cared to answer. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 07:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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 ___ 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] error 1 and 2 during make of asterisk with SUSE 8.2 and 9.1
Paul wrote: Hi, i'm traying to compile asterisk on my pc, a laptop whit SUSE 9.1 and a desktop with SUSE 8.2, with a teles S0 16/3 PnP. With Kernel 2.4 (Desktop) Asterisk run but it's umpossible to compile the driver ISDN-utils for Teles. With kernel 2.6 I can't compile zaptel (not necessary with my laptop) and asterisk, in both cases I receve errors during make or make linux26 (I saw the notes on http://www.voip-info.org/wiki+Asterisk+Zaptel+Installation). These r my notes from compiling on SUSE 9.1 Bit painful until u know what to do :-) Install the kernel souces from yast Then you need to install this rpm which is ONLY on the DVD, not on the CDs - sigh kernel-syms-2.6.4-52.i586.rpm Then run the yast online updater to get the latest kernels and sources reboot then in /usr/src/linux make cloneconfig make prepare make modules Then make a symlink from /usr/src/linux to /usr/src/linux-2.6 Then you can build all the * stuff ___ 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] Brain-dead Grandstream BT102?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sunday 18 July 2004 05:52 pm, Bruce Komito wrote: Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? I thought there was a default it reverted to if reset. But alas that probably only applies when you have functioning firmware. Sorry, no solution from me but calling Grandstream. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA+wKAljK16xgETzkRAmAQAKClsfGW8weEuD2AgZtkkDDGvjRs8QCfcVMC wvo9ztUHNjxNlC4ImsGVCMg= =uwRh -END PGP SIGNATURE- ___ 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] chan_capi won't compile
I am trying to compile chan_capi 3.3.4a, but I end up with lots of gibberish. Near the top it states that capi20.h doesn't exist. Searching for the file, several show up: # find / -name capi20.h -print /usr/src/linux-2.4.21-144/include/config/isdn/capi/capi20.h /usr/src/linux-2.4.21-231-include/smp/include/config/isdn/capi/capi20.h /usr/src/linux-2.4.21-231-include/psmp/include/config/isdn/capi/capi20.h /usr/src/linux-2.4.21-231-include/default/include/config/isdn/capi/capi20.h /usr/src/linux-2.4.21-231-include/debug/include/config/isdn/capi/capi20.h /usr/src/linux-2.4.21-231-include/smp4G/include/config/isdn/capi/capi20.h /usr/src/linux-2.4.21-231-include/athlon/include/config/isdn/capi/capi20.h How do I tell the script to look in one of these locations, or is there another step that I missed? Below follows the beginning of the log. It goes on for several hundred lines with lots of errors. asterix:/usr/src/chan_capi-0.3.4a # make install gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c In file included from /usr/include/linux/kernelcapi.h:13, from /usr/include/linux/capi.h:18, from chan_capi.c:34: /usr/include/linux/list.h:563:2: warning: #warning don't include kernel headers in userspace chan_capi.c:35:20: capi20.h: No such file or directory In file included from chan_capi.c:38: chan_capi_pvt.h:92: error: parse error before _cword chan_capi_pvt.h:92: warning: no semicolon at end of struct or union chan_capi_pvt.h:189: error: parse error before '}' token chan_capi.c:41: error: parse error before ast_capi_MessageNumber chan_capi.c:41: warning: type defaults to `int' in declaration of `ast_capi_MessageNumber' chan_capi.c:41: warning: data definition has no type or storage class chan_capi.c:103: error: parse error before _capi_put_cmsg chan_capi.c:103: error: parse error before '*' token chan_capi.c:103: warning: return type defaults to `int' chan_capi.c: In function `_capi_put_cmsg': chan_capi.c:104: error: `MESSAGE_EXCHANGE_ERROR' undeclared (first use in this function) chan_capi.c:104: error: (Each undeclared identifier is reported only once chan_capi.c:104: error: for each function it appears in.) chan_capi.c:104: error: parse error before error chan_capi.c:109: error: `error' undeclared (first use in this function) chan_capi.c:109: warning: implicit declaration of function `capi20_put_cmsg' chan_capi.c:109: error: `CMSG' undeclared (first use in this function) -- Thor Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ 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] New G.729 codec and VLANS
Anton Tinchev wrote: The readme says that the license uses all network cards MACS The MAC address is unique a 6 byte address assigned to every 802-family (802.1 Ethernet, 802.11 wireless, etc.) network interface. What happens when VLANS are added or removed? Nothing... VLANs have absolutely no effect of MAC addresses; a VLAN is just a virtual partition within a switch. Is it safe? Completely. Nick ___ 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] chan_capi-0.3.4a
Hallo, due to everchanging CVS, chan_capi-0.3.4a doesn't compile anymore with new cvs my solution was to chande chan_capi.c the line 21 from #include asterisk/parking.h to #include asterisk/features.h now chan_capi compiles again and seems back on duty again. Hope this help. Diego ___ 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] Help. New SIP hardphone.
I have an Avaya 4602SW SIP phone. They just released the SIP firmware for it the other day. I have it working with my Asterisk, but have a couple issues. My setup is like this: Avaya 4602 phone at home behind router and Asterisk server is straight on the Internet. My phone registers with Asterisk and works fine, but after a while when I pick up the handset and dial a number after I get to the last digit it just beeps at me like its out of service or has become unregistered. I am just guessing that the phone is becoming unconnected from Asterisk b/c in the CLI I see a lot of: -- Got SIP response 481 Call Does Not Exist back from my.home.ip.address But this doesnt appear in the CLI until several minutes after the phone is turned on. When I reset the phone it dials out just fine. The other thing is this. When I look in my outgoing log on my router which my phone is connected to I see: 192.168.1.52(phone IP) asterisk.public.ip 5060 But when I do sip show peers it shows: Name/usernameHostDyn Nat ACL Mask Port Status 2002/2002home.ext.ip D N 255.255.255.255 1029 Unmonitored Why does the wireless router at home show it going out 5060, but Asterisk shows it on port 1029? As well, I also get the following: Jul 18 02:11:22 WARNING[1133718080]: chan_sip.c:601 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for mailto:[EMAIL PROTECTED] for seqno 102 (Non-critical Request) In all, the phone is great, the sound quality is superb, but I dont want to have to reset it every 30 minutes or so just to use it. Any help will be well appreciated.
RE: [Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem
On 03:33 PM 7/18/2004, usedcanon wrote: Bug report might be a good idea, I just dropped the issue as I could do without using IAX. I am sure others may not have that flexibility. Umar. -Original Message- Subject: [Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem hmm - this is the bad thing about open source etc. Should we make a bugreport ? or are we just doing something wrong ? It seems that way, I asked the same question about a month ago, and no one cared to answer. Umar. -Original Message- Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem Hi, Are there realy no-one who can help here Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? I believe that 'ast_data' is the solution to this problem, and will probably obsolete mysql friends. However, I could be incorrect in that manner. There are folks on this list who would be much better informed to say whether or not it will obsolete mysql friends. -Chris ___ 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] chan_capi won't compile
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 19/07/2004, at 9:08 AM, Thor Atle Rustad wrote: I am trying to compile chan_capi 3.3.4a, but I end up with lots of gibberish. Near the top it states that capi20.h doesn't exist. Searching for the file, several show up: Make sure that you've created a link from /usr/src/linux-2.4.21 to /usr/src/linux ln -s /usr/src/linux-2.4.21 /usr/src/linux then recompile asterisk Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+w0hXeDVKqIr3GURAoupAJ48PcOTSr+/Sq9a8KPhN06s27PdEQCdEXpr jUh01lQeb6H5v5MVRJVFp7Y= =NlKe -END PGP SIGNATURE- ___ 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] Polycom IP 500 Voicemail
Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones.cfg files and using up.bypassInstantMessage="1" up.in the XML to no avail. Has anyone been able to get a Polycom 500 to use the hardbutton to retrieve voice mail and drop directly into voice mail without going through all the menus? Thanks, Wiley
[Asterisk-Users] ChanIsAvail issue
Hello
I am trying to setup ChanIsAvail function in the
extensions.conf file so that user should use the available channel to call out,
but immediately after the function like, zap channel hangup.
Here is the copy of my extensions.conf file and
messages display on consol while making the call.
Please help me to fingure out this
issue.
Thanks
Deepak
Extension.conf :
exten =
_9NXX,1,ChanIsAvail(${TRUNK})exten =
_9NXX,2,NoOP,${AVAILCHAN}exten =
_9NXX,3,Cut(TheChannel=AVAILCHAN,,1)exten =
_9NXX,4,NoOP,${TheChannel}exten =
_9NXX,5,Dial(${TheChannel}/${EXTEN:${TRUNKMSD}})exten =
_9NXX,6,Hangup
Log File:
-- Executing
ChanIsAvail("SIP/201-57f5", "Zap/g1") in new stack --
Hungup 'Zap/1-1' -- Executing NoOp("SIP/201-57f5",
"Zap/1-1") in new stack -- Executing Cut("SIP/201-57f5",
"TheChannel=AVAILCHAN||1") in new stack -- Executing
NoOp("SIP/201-57f5", "Zap/1") in new stack -- Executing
Dial("SIP/201-57f5", "Zap/1/2353070") in new stackJul 18 16:57:43
NOTICE[1200825920]: app_dial.c:689 dial_exec: Unable to create channel of type
'Zap' == Everyone is busy/congested at this time
-- Executing Hangup("SIP/201-57f5", "") in new stack == Spawn
extension (office, 92353070, 6) exited non-zero on
'SIP/201-57f5'
Re: [Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem
I believe that 'ast_data' is the solution to this problem, and will probably obsolete mysql friends. However, I could be incorrect in that manner. There are folks on this list who would be much better informed to say whether or not it will obsolete mysql friends. -Chris I did not tests with iaxfriends, but I tested some with sipfriends. I'm afraid that the support for sipfriends is not complete, because AFAIK, the additional parameters of friend can't be set, such as defaultip, nat, pickupgroup or callgroup. I dont know if ast_data bring some solution to this. Regards, Gus ___ 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] New G.729 codec and VLANS
Nicholas Bachmann wrote: Anton Tinchev wrote: The readme says that the license uses all network cards MACS The MAC address is unique a 6 byte address assigned to every 802-family (802.1 Ethernet, 802.11 wireless, etc.) network interface. What happens when VLANS are added or removed? Nothing... VLANs have absolutely no effect of MAC addresses; a VLAN is just a virtual partition within a switch. In linux VLAN appears as completely different network interface Is it safe? Completely. Adding or removing NIC? Nick ___ 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] Polycom IP 500 Voicemail
Wiley E. Siler wrote: Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using up.bypassInstantMessage=1 up.oneTouchVoicemail=1 in the XML to no avail. Has anyone been able to get a Polycom 500 to use the hardbutton to retrieve voice mail and drop directly into voice mail without going through all the menus? We programmed line 3 (line 6 on the IP 600s) on each phone with its own context/registration and set the IP 500 to auto dial into voicemail. extensions.conf: [voicemail] exten = 5501,1,voicemailmain2,[EMAIL PROTECTED] The phone.cfg file has a setting for autodial. I assume you can get a phone registered, but make sure dtmfmode is set to inband and set a mailbox= line to get MWI working. -rb ___ 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] Echo problem update - POSSIBLE SOLUTION
So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different PC with the same cards and Pstn lines? snip No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? Nope. the intent of that post was only to suggest that echo resolution varies by system, and has nothing to do with how fancy/speedy of a Compaq/Dell/HP/IBM/insert-your-favorite-box-here you might be considering or have available, or how much you spent for it. The T2240 with tdm-x100p cards in one US case does not have echo after the echotraining=800 implementation. Don't read anything more into it then just that. (The echotraining=800 was enough of a change for that exact system implementation to function well. The next one may not.) Some strong arguments have been made off-list the existing echo cancellation function is highly dependent upon interrupt latency, motherboard chipset in use, PCI controller, and/or other system-level items that might even include driver inefficiencies of the NIC card. Its way to early to pin the issue any closer, and might even involve more then one item. (Gary Mart is focusing on this and I'm sure he would appreciate any technical/programming help he can get. Now I wish I wouldn't have let those skills go years ago.) Swapping motherboards can impact echo but doing so does not address the root cause, only the symptoms. It would be nice to know XXX board works and YYY board does not, but the professional approach should focus on the underlying issue(s) and correcting/compensating for those, if possible. It could be something as simple as a linux installation default (eg, assuming 33mhz buss, choice of drivers), or as complex as rewriting how the cancellation algorithm functions in general. It is known that a lot of implementations don't have echo, and apparently those boxes are using internal system resources that fall within the tolerances of the existing cancellation routines AND those boxes have been correctly interfaced to their pstn. Why others don't needs to be identified, and unfortunately, is not a simple task. In the past eight months we've all listened to suggestions that include killing the system's GUI interface, don't share interrupts, reverse tip ring, etc, etc. However, it now _appears_ those were probably addressing the symptom and not the root cause. It's still most appropriate to ensure the pstn interfacing is implemented correctly including source of T1 sync, impedance matching, adjust gain parameters to reasonable levels, use of proper interface cards for your country's pstn standards, etc. Rich ___ 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] CID, international style?
I'm thinking of doing an app to work with the CID that's gotten from the Zap channel. All the CID's I see from within the US are 10 digit numbers. I'm out in the rural areas of the US, and no-one ever calls me from overseas. If they did, what would the CID look like? What does the CallerID look like overseas? How many countries provide it? murf ___ 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 progress detection
Hello, I haven't seen any recent posts on call progress detection, so here's a question: How would one accomplish an automated outbound dialing application using *, whereby a requirement is to wait for the greeting to complete (live person, answering machine, voicemail) before delivering the message? For example, playing a 'reminder' message to a list of recipients. I know its possible using telephony boards (ie. Dialogic/Intel), but don't know about *. I have experimented with callprogress=yes in zapata.conf, but not sure if that was intended to cover what i describe above. Regards, Steve ___ 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 progress detection
On Sun, 2004-07-18 at 20:38, Stephen David wrote: Hello, I haven't seen any recent posts on call progress detection, so here's a question: How would one accomplish an automated outbound dialing application using *, whereby a requirement is to wait for the greeting to complete (live person, answering machine, voicemail) before delivering the message? For example, playing a 'reminder' message to a list of recipients. I know its possible using telephony boards (ie. Dialogic/Intel), but don't know about *. I have experimented with callprogress=yes in zapata.conf, but not sure if that was intended to cover what i describe above. callprogress is to detect pickup, ringing, hangup, and busy signal on analog lines that don't support a complex enough signalling to support a computer on the other side. What you need is something like a ecording looking for silence post answer. AGI supports record with silence detection. Once you detect the specified amount of silence, you can play your message. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] LAN Switch w/ QoS
Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ 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] chan_capi won't compile
Make sure that you've created a link from /usr/src/linux-2.4.21 to /usr/src/linux ln -s /usr/src/linux-2.4.21 /usr/src/linux then recompile asterisk The symlinks were already there. # ls -ld /usr/src/linux* lrwxrwxrwx1 root root 25 Jul 19 03:46 /usr/src/linux - /usr/src/linux-2.4.21-231 drwxr-xr-x 16 root root 600 Jul 19 03:45 /usr/src/linux-2.4.21-144 drwxr-xr-x 15 root root 464 Jul 18 22:00 /usr/src/linux-2.4.21-215 drwxr-xr-x 18 root root 728 Jul 18 22:00 /usr/src/linux-2.4.21-231 drwxr-xr-x8 root root 192 Jul 18 21:58 /usr/src/linux-2.4.21-231-include lrwxrwxrwx1 root root 24 Jul 18 22:00 /usr/src/linux-include - linux-2.4.21-231-include Recompiled Asterisk. Chan_capi still won't compile. Thor ___ 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] LAN Switch w/ QoS
On Sun, 18 Jul 2004, Michael Welter wrote: Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. You can get Cisco 2950s for about $600/24 ports. ___ 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] LAN Switch w/ QoS
On Jul 18, 2004, at 7:14 PM, [EMAIL PROTECTED] wrote: On Sun, 18 Jul 2004, Michael Welter wrote: Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. You can get Cisco 2950s for about $600/24 ports. And 48 ports from Dell for about the same price. I haven't used any of their latest round of switches, but their older ones were decent for the price. Cisco's switches are almost certainly better-made, but Dell's not *usually* that bad. Scott ___ 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] TE405P
I'm installing TE405P card. This is my zaptel.conf. -- span=1,0,0,ccs,hdb3,crc4span=2,1,0,ccs,hdb3,crc4span=3,0,0,ccs,hdb3,crc4span=4,0,0,ccs,hdb3,crc4 loadzone = usdefaultzone= us-- When i modprobe wct4xxp, -- PCI: Found IRQ 11 for device 02:0b.0PCI: Sharing IRQ 11 with 00:1d.7Found TE410P at base address ed80, remapped to e0951000TE410P version c01a009bFALC version: 0005, Board ID: 00Reg 0: 0x1ad4b800Reg 1: 0x1ad4b000Reg 2: 0x07fc07fcReg 3: 0xReg 4: 0xReg 5: 0xReg 6: 0xc01a009bReg 7: 0x1000Reg 8: 0xReg 9: 0x00ffReg 10: 0xTE410P: Launching card: 0TE410P: Setting up global serial parametersFound a Wildcard: Wildcard TE410P-XilinxRegistered tone zone 0 (United States / North America)TE410P: Span 1 configured for ESF/B8ZSTE410P: Span 2 configured for ESF/B8ZSSPAN 2: Primary Sync SourceTE410P: Span 3 configured for ESF/B8ZSTE410P: Span 4 configured for ESF/B8ZSwct4xxp: Setting yellow alarm on span 1wct4xxp: Setting yellow alarm on span 2wct4xxp: Setting yellow alarm on span 3wct4xxp: Setting yellow alarm on span 4 -- I have two questions. - Is TE410P is same as TE405P, or did I received different card? - zaptel.conf is configured CCS/HDB3. But It's configured as ESF/B8ZS. Hong
[Asterisk-Users] Adding voice mail box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA+zjhljK16xgETzkRAh8jAKCJ7iJhFBVRxBFzbl8cGziqbnUjoQCdEzbb oTA7sXW1EXmmDGpUXrPf174= =zANK -END PGP SIGNATURE- ___ 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] LAN Switch w/ QoS
I have been quite happy with our HP 2848 GigE switches that we put in for our desktops a few months ago. I have also used the 2650 48 10/100 + 2 GigE switches before. We are looking at the 2650-PWR for our VoIP deployment (only about 60 phones for our USGS/U of A mixed department). Harry On Sun, 2004-07-18 at 19:39, Scott Laird wrote: On Jul 18, 2004, at 7:14 PM, [EMAIL PROTECTED] wrote: On Sun, 18 Jul 2004, Michael Welter wrote: Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. You can get Cisco 2950s for about $600/24 ports. And 48 ports from Dell for about the same price. I haven't used any of their latest round of switches, but their older ones were decent for the price. Cisco's switches are almost certainly better-made, but Dell's not *usually* that bad. Scott ___ 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 -- Harry McGregor, Computing Manager Tucson Support Group - U.S. Geological Survey University of Arizona - Environment and Natural Resource Building 520-670-5574 (office) - [EMAIL PROTECTED] 520-661-7875 (Cell) - [EMAIL PROTECTED] The opinions/statements expressed herein are my own and should not be taken as a position, opinion, or endorsement of the University of Arizona or the U.S. Geological Survey. ___ 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] Adding voice mail box
Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve Look for your controb/script directory. The script is called 'addmailbox'. Regards, Gus ___ 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] Adding voice mail box
On Sunday 18 July 2004 11:21 pm, CW_ASN wrote: Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve Look for your controb/script directory. The script is called 'addmailbox'. Regards, Gus Nah, I cannot do that! It's a bit too obvious... Thx! -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] PSTN Gateway X101P
When I call the pstn number, the zaptel picks up the line on the first ring and then forwards it to the sip phone and rings it. Is there anyway to prevent the zaptel from picking up the line until the sip phone actully answers the call. This way I could answer the phone either locally on a regular analog handset or through the sip phone. The way it is now, it only rings my phones in the house 1 time. Jason Hey Jason, glad things are working...I think I understand your problem and the short answer is no - there isn't a way to ring the x-lite without asterisk answering the call first (if I'm wrong about this, someone please correct me!). It sounds like your analog telephone isn't connected into the asterisk box, but instead just plugged into a standard wall outlet somewhere, connected directly to the pstn. If this is the case, you will be limited b/c asterisk must answer the call before it can do any other processing such as ring another phone, etc...you might be able to configure asterisk to answer after 5 rings or something, giving you a chance to answer the analog phone first, but most people would probably do the following: The way around this is to connect your analog phone into asterisk and have asterisk ring the analog phone and the x-lite simultaneously, giving you the choice of how to answer it. There are a couple of ways to do this, such as a Digium TDM400B pci card with 1 FXS module installed in it (to which you would connect the phone), or a SIP (or H.323, or IAX) to FXS adapter such as the cisco ata 286 or the sipura 2000, etc.. (various models are described on the wiki)... There are plenty of advantages to this such as music on hold, the ability to transfer calls between x-lite and the analog phone, and plenty more as described on the wiki.. Marty ___ 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] Echo problem update - POSSIBLE SOLUTION
Thanks for that. Like many I believe * is unusable in production until these echo issues are quoshed are resolved. Lets hope someone takes up the bounty offer.Rich Adamson [EMAIL PROTECTED] wrote: So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different PC with the same cards and Pstn lines? No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes?Nope. the intent of that post was only to suggest that echo resolutionvaries by system, and has nothing to do with how fancy/speedy of a Compaq/Dell/HP/IBM/ you might be considering or have available, or how much you spent for it. The T2240 with tdm-x100p cards in "one US case" does not have echo afterthe echotraining=800 implementation. Don't read anything more into itthen just that. (The echotraining=800 was enough of a change for thatexact system implementation to function well. The next one may not.)Some strong arguments have been made off-list the existing echo cancellation function is highly dependent upon interrupt latency,motherboard chipset in use, PCI controller, and/or other system-level items that might even include driver inefficiencies of the NIC card. Its way to early to pin the issue any closer, and might even involvemore then one item. (Gary Mart is focusing on this and I'm sure hewould appreciate any technical/programming help he can get. Now Iwish I wouldn't have let those skills go years ago.)Swapping motherboards can impact echo but doing so does not addressthe root cause, only the symptoms. It would be nice to know XXX boardworks a nd YYY board does not, but the professional approach shouldfocus on the underlying issue(s) and correcting/compensating for those,if possible. It could be something as simple as a linux installation default (eg, assuming 33mhz buss, choice of drivers), or as complex as rewriting how the cancellation algorithm functions in general.It "is" known that a lot of implementations don't have echo, andapparently those boxes are using internal system resources that fall within the tolerances of the existing cancellation routines ANDthose boxes have been correctly interfaced to their pstn. Why others don't needs to be identified, and unfortunately, is not asimple task.In the past eight months we've all listened to suggestions thatinclude killing the system's GUI interface, don't share interrupts, reverse tip ring, etc, etc. However, it now _appears_ those wereprobably addressing the symptom and not the root cause.It 's still most appropriate to ensure the pstn interfacing is implemented correctly including source of T1 sync, impedance matching,adjust gain parameters to reasonable levels, use of proper interfacecards for your country's pstn standards, etc.Rich___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself
[Asterisk-Users] SIP to H323 call timeout
Hi all, I have the following setup: UAs SER -- ASTERISK --GNUGK - GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out immediately, so call setup is never completed. On GNUGK the call request comes in followed by a normal call drop. Any ideas on what could be the problem ?? My asterisk configuration, debug and console output are as follow : SIP.CONF == [general] port = 5080 bindaddr = 10.10.1.170 context = to_GNUGK disallow=all allow=g729 H323.CONF === [general] port = 1720 allow = g729 gatekeeper = 64.80.103.12 allowgkrouted = yes context = to_SER EXTENSIONS.CONF [general] static = yes writeprotect = yes [to_GNUGK]] exten = _.,1,Dial(h323/[EMAIL PROTECTED]:1720,60,C) [to_SER] exten = _.,1,Dial(SIP/[EMAIL PROTECTED]:5060,60) DEBUG File == Jul 15 16:14:10 DEBUG[65541]: Check for res for Jul 15 16:14:10 DEBUG[65541]: is not a local user Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop: sip:[EMAIL PROTECTED];ftag=661806388;lr=on Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username: 15613021234 Jul 15 16:14:10 DEBUG[311316]: [EMAIL PROTECTED]:1720, timeout=0. Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t Jul 15 16:14:23 DEBUG[311316]: [EMAIL PROTECTED]:1720, timeout=0. Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter Jul 15 16:14:31 DEBUG[311316]: is not a local user Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found CONSOLE Output == *CLI -- Executing Dial(SIP/-08121388, h323/[EMAIL PROTECTED]:1720|60|C) in new stack -- Called [EMAIL PROTECTED]:1720 == No one is available to answer at this time -- Timeout on SIP/-08121388 == CDR updated on SIP/-08121388 _ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus ___ 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] GUI based.. or ??
Hi, I am Abhishek from India. I am have studying Cisco VOIP since a couple of months.Searching for Soft PBX somenthing like (Cisco Callmanager) i came accros this Asterisk. I have to provide a a solution to a clinet where he wants a connectivity between his 3 offices across the WAN with a very limited amount of budget.Since i am not aware abt this product much, but was able to foind out the features of the product and was satisfied also, So i just wanted to know from u ppl (since u ppl are expert in this) that : 1.)does this product has got a GUI interface .?? 2.)Can we integrate Cisco or any other H/w with this.? 3.)it looks like freeware..isnt it.? Please do let me the details abt the same.. I ll be really greatful Thanking You, Regards Abhishek Katta - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 8:45 PM Subject: Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different PC with the same cards and Pstn lines? snip No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? Nope. the intent of that post was only to suggest that echo resolution varies by system, and has nothing to do with how fancy/speedy of a Compaq/Dell/HP/IBM/insert-your-favorite-box-here you might be considering or have available, or how much you spent for it. The T2240 with tdm-x100p cards in one US case does not have echo after the echotraining=800 implementation. Don't read anything more into it then just that. (The echotraining=800 was enough of a change for that exact system implementation to function well. The next one may not.) Some strong arguments have been made off-list the existing echo cancellation function is highly dependent upon interrupt latency, motherboard chipset in use, PCI controller, and/or other system-level items that might even include driver inefficiencies of the NIC card. Its way to early to pin the issue any closer, and might even involve more then one item. (Gary Mart is focusing on this and I'm sure he would appreciate any technical/programming help he can get. Now I wish I wouldn't have let those skills go years ago.) Swapping motherboards can impact echo but doing so does not address the root cause, only the symptoms. It would be nice to know XXX board works and YYY board does not, but the professional approach should focus on the underlying issue(s) and correcting/compensating for those, if possible. It could be something as simple as a linux installation default (eg, assuming 33mhz buss, choice of drivers), or as complex as rewriting how the cancellation algorithm functions in general. It is known that a lot of implementations don't have echo, and apparently those boxes are using internal system resources that fall within the tolerances of the existing cancellation routines AND those boxes have been correctly interfaced to their pstn. Why others don't needs to be identified, and unfortunately, is not a simple task. In the past eight months we've all listened to suggestions that include killing the system's GUI interface, don't share interrupts, reverse tip ring, etc, etc. However, it now _appears_ those were probably addressing the symptom and not the root cause. It's still most appropriate to ensure the pstn interfacing is implemented correctly including source of T1 sync, impedance matching, adjust gain parameters to reasonable levels, use of proper interface cards for your country's pstn standards, etc. Rich ___ 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] GUI based.. or ??
Abhishek, In reverse order 3/ yes it is freeware, though some of the termination boards are available for sale from www.digium.com 2/ yes you can interface to Cisco handsets running SIP. 1/ Does it have a gui interface - the short answer is no. The longer answer is depending on what you mean, if you mean programming - then no though a number of people have developed sql interfaces. If you mean softphones then yes there are a number of software based phones such as x-lite. I hope this answers some of your questions, keep looking and asking this is a good product for you to learn on and to research further. Cheers, Dean Sydney, Australia -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abhishek Katta Sent: Tuesday, 2 March 1999 3:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] GUI based.. or ?? Hi, I am Abhishek from India. I am have studying Cisco VOIP since a couple of months.Searching for Soft PBX somenthing like (Cisco Callmanager) i came accros this Asterisk. I have to provide a a solution to a clinet where he wants a connectivity between his 3 offices across the WAN with a very limited amount of budget.Since i am not aware abt this product much, but was able to foind out the features of the product and was satisfied also, So i just wanted to know from u ppl (since u ppl are expert in this) that : 1.)does this product has got a GUI interface .?? 2.)Can we integrate Cisco or any other H/w with this.? 3.)it looks like freeware..isnt it.? Please do let me the details abt the same.. I ll be really greatful Thanking You, Regards Abhishek Katta - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 8:45 PM Subject: Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different PC with the same cards and Pstn lines? snip No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? Nope. the intent of that post was only to suggest that echo resolution varies by system, and has nothing to do with how fancy/speedy of a Compaq/Dell/HP/IBM/insert-your-favorite-box-here you might be considering or have available, or how much you spent for it. The T2240 with tdm-x100p cards in one US case does not have echo after the echotraining=800 implementation. Don't read anything more into it then just that. (The echotraining=800 was enough of a change for that exact system implementation to function well. The next one may not.) Some strong arguments have been made off-list the existing echo cancellation function is highly dependent upon interrupt latency, motherboard chipset in use, PCI controller, and/or other system-level items that might even include driver inefficiencies of the NIC card. Its way to early to pin the issue any closer, and might even involve more then one item. (Gary Mart is focusing on this and I'm sure he would appreciate any technical/programming help he can get. Now I wish I wouldn't have let those skills go years ago.) Swapping motherboards can impact echo but doing so does not address the root cause, only the symptoms. It would be nice to know XXX board works and YYY board does not, but the professional approach should focus on the underlying issue(s) and correcting/compensating for those, if possible. It could be something as simple as a linux installation default (eg, assuming 33mhz buss, choice of drivers), or as complex as rewriting how the cancellation algorithm functions in general. It is known that a lot of implementations don't have echo, and apparently those boxes are using internal system resources that fall within the tolerances of the existing cancellation routines AND those boxes have been correctly interfaced to their pstn. Why others don't needs to be identified, and unfortunately, is not a simple task. In the past eight months we've all listened to suggestions that include killing the system's GUI interface, don't share interrupts, reverse tip ring, etc, etc. However, it now _appears_ those were probably addressing the symptom and not the root cause. It's still most appropriate to ensure the pstn interfacing is implemented correctly including source of T1 sync, impedance matching, adjust gain parameters to reasonable levels, use of proper interface cards for your country's pstn standards, etc. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] Polycom IP 500 Voicemail
I have a solution that allows me to assign a soft key with no problems. However, it seems like a waste the the hard button labeled Voice Mail is not dialing right into voice mail. Is there a known way yo do this? I have tried everything in the manual but it doesn't seem to work. I have IP 500s and I want to be able to use all three display lines for just lines on the phone. Also, do you know if it is possible to program the buttons along the bottom of the screen like normal soft buttons? And finally... Is there a way to make the system dial without having to hit the Send key after dialing a number? Thanks for the tips! Wiley -Original Message- From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004 5:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley E. Siler wrote: Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using up.bypassInstantMessage=1 up.oneTouchVoicemail=1 in the XML to no avail. Has anyone been able to get a Polycom 500 to use the hardbutton to retrieve voice mail and drop directly into voice mail without going through all the menus? We programmed line 3 (line 6 on the IP 600s) on each phone with its own context/registration and set the IP 500 to auto dial into voicemail. extensions.conf: [voicemail] exten = 5501,1,voicemailmain2,[EMAIL PROTECTED] The phone.cfg file has a setting for autodial. I assume you can get a phone registered, but make sure dtmfmode is set to inband and set a mailbox= line to get MWI working. -rb ___ 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] Asterisk Control Script
Does anyone know where I can find a list of all the control scripts? I want to write a standard windows tool that will allow you to pregenerate the configuration for your Asterisk install and them press one button to have it log into your boxand upload the scripts. Of course, I will let everyone know when it is complete. Thanks, Wiley
Re: [Asterisk-Users] Adding voice mail box
Dont have to.. just add it to the voicemail.conf and it will auto do everything for you. bkw - Original Message - From: Steve [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 9:58 PM Subject: [Asterisk-Users] Adding voice mail box -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA+zjhljK16xgETzkRAh8jAKCJ7iJhFBVRxBFzbl8cGziqbnUjoQCdEzbb oTA7sXW1EXmmDGpUXrPf174= =zANK -END PGP SIGNATURE- ___ 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] GR-303 and _FXS_ support!
For those who don't watch asterisk-cvs, it appears that markster has begun (and possibly) completed adding GR-303 FXS support to Asterisk. This means that Asterisk could be used as an access concentrator off of a class 5 switch, which gives us a higher-level alternative between using single PRIs and going all the way to SS7. I for one am very interested in pursuing this option as soon as someone out there has tested it on a live connection... don't think I can afford to be the guinea pig, though, unless my telco really wants to have some fun :-) ___ 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
