Re: [Asterisk-Users] Play message and dial extensions simultaneously
Hi, I've been looking for such solution without lucks. The prompts (either by Playback or Background or app_queues) will have to complete before the Dial cmd kicks in, which takes a lot of time. Please let me know if you become aware of any solutions for this apparently obvious problem. Regards, H. On 11/10/05, C F [EMAIL PROTECTED] wrote: For what purpose? Have you tried: exten = s,1,Dial(SIP/,15Local/[EMAIL PROTECTED]) exten = 123,1,Background(custom/msg1) It might not work, I have never tried something like this, but it might work. On 11/8/05, Mike Clark [EMAIL PROTECTED] wrote: Ok, this has to be simple and I'm just not seeing it. On and inbound call, I want to play a specific message while simultaneously ringing extensions. Its basically like music on hold and queues, but I need the message to always start from the beginning, not just play from where the MOH process happens to be at that time. I tried Googling, but no luck. I did try exten = 1,1,Answer exten = 1,2,Wait(1) exten = 1,3,BackGround(custom/msg1) exten = 1,4,Dial(SIP/,15) but it played the entire message before dialing. Thanks, Mike Clark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Test environment for a Predictive Dialer
Hi Markus I did the same to test one application on intelligent cards from Pikatechnologies. I had no E1 in office so I set up one Asterisk box with TE110P to simulate a CO. The only thing is to change support for protocol. In ZAPATA.CONF of one system use(the real PABX): signalling = bri_cpe on the other(the CO simulator) signalling = bri_net Then use a cross E1 cable. Hope it helps! Ciao Mauro Message: 11 Date: Wed, 9 Nov 2005 18:45:37 +0100 From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Test environment for a Predictive Dialer To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hello all, I'm thinking about to set up a test environment for a predictive dialler with two asterisk machines. Each Asterisk should use a Digium TE110P card. One machine should work as predictive dialler; the other box should simulate the PSTN. - Is it in general possible to interconnect the two asterisk machines in that way? Do I need any hardware in between to connect the two TE110P cards? - Can I simulate the PSTN with a Digium TE110P card? Thanks and Regards Markus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] groupware + unified messagerie +Asterisk
Yes I know this project however my goal would be something like this : [FAX] PSTN--[VOICE]--ASTERISK--(e)groupware [SMS] | Mail Server So (e)groupware' clients should be able to send/receive voice messages fax and sms from/to e-mail click to dial contacts in address book and more :) What do you think of this project ? Regards Harry --- Robert Rozman [EMAIL PROTECTED] a écrit : Hi, I guess you know this project, but just in case: http://jivesoftware.org/asterisk-im/ IMHO, Egroupware would be best groupware solution to start on, but they have little interest in doing that (searching their mailing list for voip returned 2 hits...). We will gradually start working on merging java sip client with Asterisk-IM client and see what will come out Regards, Rob. - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 10, 2005 5:25 AM Subject: Re: [Asterisk-Users] groupware + unified messagerie +Asterisk harry gaillac wrote: it's no what i expect the easier solution you provide the more customers you get ! Indeed. However, I tend to be of the opinion that you should have enough money in the bank for a full year of wages for someone if you take on extra staff. While this may make my growth slower, at least I can honestly guarantee my staff's continued employment! So, to cut a long story short, I don't have enough staff to write an infinitely configurable one, as I currently have my books pretty crammed with jobs. If you have any questions though and want to develop one yourself, I'm more than happy to help you! :D -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 NAT register
I'm unable to register soft phone on * that is behind NAT. I have SP on public address 195.29.109.0 which is dynamically changed. * is in private address 10.0.0.81 that is behind NAT on address xxx.xxx.xxx.xxx When I try to register this is message that I receive on * CLI # Testing 195.29.109.0 with 10.0.0.0 Target address 195.29.109.0 is not local, substituing externip And that is all. When I enter sip show peers I get Name/username HostDyn Nat ACL Mask PortStatus 2150/2150 (Unspecified) D 255.255.255.255 0 Unmonitored This is how my sip.conf looks like. [general] nat=yes externip = xxx.xxx.xxx.xxx ; here I have my public IP fromdomain = mydomain.hr localnet = 10.0.0.0/255.255.255.0 port=5060 bindaddr=0.0.0.0 context=sip srvlookup=yes dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw allow=alaw musicclass=default [2150] type=friend username=2150 secret=2150 host=dynamic mailbox=2150 Have I done something wrong or is there I haven't done? Please help. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] groupware + unified messagerie +Asterisk
Thanks for your advises it's no what i expect the easier solution you provide the more customers you get ! I don't agree you ! the best solution you provide the more customers you get (apache projects) ! Indeed. However, I tend to be of the opinion that you should have enough money in the bank for a full year of wages for someone if you take on extra staff. A commercial solution would be a better choice ! While this may make my growth slower, at least I can honestly guarantee my staff's continued employment! So, to cut a long story short, I don't have enough staff to write an infinitely configurable one, as I currently have my books pretty crammed with jobs. I agree you I don't ask you to write this project . asterisk hylafax (e)groupware have been written why not provide an open source solution to improve the use of asterisk for the users . If you have any questions though and want to develop one yourself, I'm more than happy to help you! thank you for your assistance Regards Harry PS: What about presence/IM may i load the lastest asterisk on cvs ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CentOS vs. Vanilla Kernel
Julian - What hardware are you using? Proc, RAM, SCSI or IDE, etc. The reason I ask is that I have multiple hardware platforms, all on FC1 or FC4, and none of them hit 100% for each IRQ. I am usually in the high 98% with the occasional 100% on P3 servers (give or take 1 Gig RAM, 1 Gig CPU). Two servers are dual p3 1.2 with 2 Gigs Ram. Since CentOS is brought up, maybe my OS is the culprit...far fetched? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, November 10, 2005 12:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CentOS vs. Vanilla Kernel Not a problem that I've had :) Linux foxtrot.tessera.co.uk 2.6.9-22.0.1.EL #1 Thu Oct 27 12:26:11 CDT 2005 i686 i686 i386 GNU/Linux Opened pseudo zap interface, measuring accuracy... 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% --- Results after 24 passes --- Best: 100.00 -- Worst: 100.00 -- Average: 100.00 [EMAIL PROTECTED] zaptel]# Julian. [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote on 11/07/2005 01:17:31 PM: HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Will you be using Zaptel hardware? The only way I can get zttest results of 100% is with a CentOS 2.4 kernel. Any CentOS 2.6 kernel I've tried (Uni, SMP, with IOAPIC enabled or disabled) gave me 99.99% at best... Tim Massey -- -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Can't create iax channel
The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. Based on your post, seems that you have an issue with codecs more than creating an IAX trunk. What version of Asterisk are you using? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Gemmell Sent: Thursday, November 10, 2005 12:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can't create iax channel Hi all Could somebody please give me an idea as to whats wrong here. I'm trying to connect 2 servers using IAX, I'm not trunking them because I read that you need zaptel hardware installed at both sides to do the trunking. Theregistration seems to have worked as the output of iax show peers on the side I'm working from is as follows Name/UsernameHost Mask Port Status wayne165.165.164.87 (D) 255.255.255.255 4569 Unmonitored and on the other side iax2 show users shows Username SecretAuthen Def.Context A/C Codec Pref waynepassword 001 default No Host When trying to call from this side to that side I get the following -- Executing Dial(SIP/301-2d50, IAX2/wayne:[EMAIL PROTECTED]/204) in new stack Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: chan_iax2.c:7745 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/wayne-5 -- Hungup 'IAX2/wayne-5' Nov 10 08:37:21 NOTICE[30785]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion(SIP/301-2d50, ) in new stack == Spawn extension (from-internal, 204, 2) exited non-zero on 'SIP/301-2d50' Any ideas? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 1.2.0-RC1 Crashing with g729 codec and ATA 186
Gervais de Montbrun ha scritto: I downloaded the chan_sccp as you suggested, but it does not seem to support my Cisco 12 SP+. I can see that it would support the ata, but if it doesn't support my other phone, then I need the skinny protocol and then can't use sccp... :-( the 12SP should work Do you know if I can get it to work with both my Cisco 12 SP+ and my ATA-186? Well you just need to change the default tcp port you can use chan_sccp on port 2000 and chan_skinny on port 2001 Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Play message and dial extensions simultaneously
You can play music instead of providing a ringtone. ( I think it's the M option for the dial command) We used this for a reception solution so that the caller would not know that they were not being ignored. PaulH - Original Message - From: Hugh Jackman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 10, 2005 7:04 PM Subject: Re: [Asterisk-Users] Play message and dial extensions simultaneously Hi, I've been looking for such solution without lucks. The prompts (either by Playback or Background or app_queues) will have to complete before the Dial cmd kicks in, which takes a lot of time. Please let me know if you become aware of any solutions for this apparently obvious problem. Regards, H. On 11/10/05, C F [EMAIL PROTECTED] wrote: For what purpose? Have you tried: exten = s,1,Dial(SIP/,15Local/[EMAIL PROTECTED]) exten = 123,1,Background(custom/msg1) It might not work, I have never tried something like this, but it might work. On 11/8/05, Mike Clark [EMAIL PROTECTED] wrote: Ok, this has to be simple and I'm just not seeing it. On and inbound call, I want to play a specific message while simultaneously ringing extensions. Its basically like music on hold and queues, but I need the message to always start from the beginning, not just play from where the MOH process happens to be at that time. I tried Googling, but no luck. I did try exten = 1,1,Answer exten = 1,2,Wait(1) exten = 1,3,BackGround(custom/msg1) exten = 1,4,Dial(SIP/,15) but it played the entire message before dialing. Thanks, Mike Clark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Can't create iax channel
On Thursday 10 November 2005 10:55, Jason Walker wrote: The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. I don't know where I read it, apparently it is needed for timing or something, could be in the old handbook or hitchikers guide to asterisk as I havn't got far enough into the new handbook to comment. Based on your post, seems that you have an issue with codecs more than creating an IAX trunk. Thanks, yes I was disallowing all codecs, :( -- Cheers Wayne ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 OH-323 module-Inbound Call dropped due to in-call-rate violation (1.55)
Hi to all,
i have installed the latest CVS asterisk version as well as the
asterisk-oh323-0.7.3.
I have also installed the openh323-v1_17_2 and pwlib-v1_9_1 ( i also tried
the Mimas patched oh323 and pwlib but they did not behave well as far as the
gatekeeper registration was concerned).
The problem i now have, is that when i call from a h323 terminal
(netmeeting) to an Asterisk registered SIP client i get the following:
Nov 9 18:19:56 WARNING[20122] chan_oh323.c: Inbound call
'ip$192.168.1.1:10235/23826-488a9126' dropped due to in-call-rate violation
(1.55)
---where 192.168.1.1 is the asterisk server.
The oh323.conf is as follows:
; Configuration file of OpenH323 channel driver
;
;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=1
tcpEnd=2
;
; Configure the UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
; rtp.conf
;
udpStart=1
udpEnd=2
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
; lowdelay, throughput, reliability, mincost, none
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
;outboundMax=100
;inboundMax=100
;simultaneousMax=100
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=2
;crlThreshold=30
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=10
libTraceLevel=10
libTraceFile=/var/log/asterisk/oh323.log
;
; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is the
zone name.
; Valid values for this option are:
; DISABLE,
; DISCOVER,
; gatekeeper's DNS name,
; gatekeeper's ip,
; GKID:gatekeeper's id
; gatekeeper's id@gatekeeper's name or address
;
gatekeeper=192.168.2.1
;gatekeeper=DISCOVER
;
; Set the gatekeeper password. If used, it enables H.235 access to
gatekeeper.
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout. Before the expiration of
; the timeout, a re-registration is attempted.
;
gatekeeperTTL=600
;
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
; Q931- Q.931 Keypad Information Element
; STRING - H.245 string
; TONE- H.245 tone
; RFC2833 - RFC2833
; INBAND -
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Default language
;
language=en
;
; Default Music-On-Hold class
;
musiconhold=default
;
; Set the default context of H.323 calls.
;
context=voip-h323
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
alias=asterisk
gw=12345678
;alias=123
;
; Aliases/prefixes routed in all-aliases context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in more-aliases context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in all-prefixes context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in more-stuff context.
;
;context=more-stuff
;alias=664
;gwprefix=02
;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
; Define the codec list of the channel driver.
; Every codec option may have a frames option
; associated with it.
; Valid values for the codec option are:
; G711U - G.711 u-Law
; G711A - G.711 A-Law
; G7231 - G.723.1(6.3k)
; G72316K3- G.723.1(6.3k)
; G72315K3- G.723.1(5.3k)
; G7231A6K3 - G.723.1A(6.3k)
;
[Asterisk-Users] Asterisk 1.0.9 + TE210 --- Long
Hi all,
I am trying to test Asterisk with TE210 and SPANDSP. So i connect
back-to-back (with E1 crossover cable) the two E1 ports of the TE210
and it seems that everything is fine. The i create a script that
calls an extension that starts the rxfax application and initiate
another extension that starts the txfax application. What i expect is
to start sending a fax from the first E1 and to receive it from the
other. But the result is to open the appropriate channels, move
normall to the next priorities of each extension that means start
sending fax and receiving fax but nothing more, freeze there
Doing a pri intense debug at the spans i can see that the T203
counter restarts all the time.
Find bellow configuration.
Please help!
--- SCRIPT ---
#!/usr/bin/perl
use Asterisk::Manager;
$|++;
my $astman = new Asterisk::Manager;
$astman-user('admin');
$astman-secret('secret');
$astman-host('localhost');
$astman-connect || die $astman-error . \n;
$astman-setcallback('Hangup', \hangup_callback);
$astman-setcallback('DEFAULT', \default_callback);
print $astman-sendcommand( Action = 'Originate',
Callerid = SLOT1,
Channel = 'Zap/g1/getfax',
Exten = 'sendfax',
Context = 'Outgoing',
Priority = '1' );
$astman-eventloop;
$astman-disconnect;
sub hangup_callback {
printf(hangup callback\n);
}
sub default_callback {
my (%stuff) = @_;
foreach (keys %stuff) {
printf(%s: %s\n, $_, $stuff{$_});
}
printf(\n);
}
--- RESULT WHEN RUNNING THE SCRIPT ---
EventNewchannelChannelZap/3-1StateRsrvdCallerIDunknownUniqueid1131547210.4CallerID:
SLOT1
Event: Newcallerid
Uniqueid: 1131547210.4
Channel: Zap/3-1
CallerID: SLOT1
Event: Newcallerid
Uniqueid: 1131547210.4
Channel: Zap/3-1
CallerID: SLOT1
Event: Newstate
Uniqueid: 1131547210.4
Channel: Zap/3-1
State: Dialing
CallerID: unknown
Event: Newchannel
Uniqueid: 1131547210.5
Channel: Zap/34-1
State: Ring
Event: Newexten
Channel: Zap/34-1
Context: Incoming
Extension: getfax
Application: SetVar
Uniqueid: 1131547210.5
AppData: FAXFILE=/tmp/1131547210.5.tiff
Priority: 1
Event: Newexten
Channel: Zap/34-1
Context: Incoming
Extension: getfax
Uniqueid: 1131547210.5
Application: RxFAX
AppData: /tmp/1131547210.5.tiff
Priority: 2
CallerID: unknown
Event: Newstate
Channel: Zap/34-1
State: Up
Uniqueid: 1131547210.5
CallerID: SLOT1
Event: Newstate
Channel: Zap/3-1
State: Up
Uniqueid: 1131547210.4
Event: Newexten
Channel: Zap/3-1
Context: Outgoing
Extension: sendfax
Uniqueid: 1131547210.4
Application: SetVar
AppData: SENDFAX=/tmp/sendfax.tiff
Priority: 1
Event: Newexten
Channel: Zap/3-1
Context: Outgoing
Extension: sendfax
Uniqueid: 1131547210.4
Application: TxFAX
AppData: /tmp/sendfax.tiff|caller
Priority: 2
* Stays there
--- ZAPATA.CONF ---
[trunkgroups]
; define any trunk groups
[channels]
switchtype=euroisdn
;pridialplan=national
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=no
rxgain=0.0
txgain=0.0
;faxdetect=both
; Span 1
context=Outgoing
group=1
;signalling=pri_net
signalling=pri_cpe
channel = 1-15
channel = 17-31
; Span 2
context=Incoming
group=2
signalling=pri_net
;signalling=pri_cpe
channel = 32-46
channel = 48-62
--- ZAPTEL.CONF ---
#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=span num,timing,line build out (LBO),framing,coding[,yellow]
#
# The timing parameter determines the selection of primary, secondary, and
# so on sync sources. If this span should be considered a primary sync
# source, then give it a value of 1. For a secondary, use 2, and so on.
# To not use this as a sync source, just use 0
#
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1: 133-266 feet (DSX-1)
# 2: 266-399 feet (DSX-1)
# 3: 399-533 feet (DSX-1)
# 4: 533-655 feet (DSX-1)
# 5: -7.5db (CSU)
# 6: -15db (CSU)
# 7: -22.5db (CSU)
#
# The framing is one of d4 or esf for T1 or cas or ccs for E1
#
# Note: d4 could be referred to as sf or superframe
#
# The coding is one of ami or b8zs for T1 or ami or hdb3 for E1
#
# E1's may have the additional keyword crc4 to enable CRC4 checking
#
# If the keyword yellow follows, yellow alarm is transmitted when no
# channels are open.
#
#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4
#
# Next come the dynamic span definitions, in the form:
# dynamic=driver,address,numchans,timing
#
# Where driver is the name of the driver (e.g. eth), address is the
# driver specific address (like a MAC for eth), numchans is the number
# of channels, and timing is a timing priority, like for a
[Asterisk-Users] Asterisk 1.0.9 + TE210 --- Long
Hi all,
I am trying to test Asterisk with TE210 and SPANDSP. So i connect
back-to-back (with E1 crossover cable) the two E1 ports of the TE210
and it seems that everything is fine. The i create a script that
calls an extension that starts the rxfax application and initiate
another extension that starts the txfax application. What i expect is
to start sending a fax from the first E1 and to receive it from the
other. But the result is to open the appropriate channels, move
normall to the next priorities of each extension that means start
sending fax and receiving fax but nothing more, freeze there
Doing a pri intense debug at the spans i can see that the T203
counter restarts all the time.
Find bellow configuration.
Please help!
--- SCRIPT ---
#!/usr/bin/perl
use Asterisk::Manager;
$|++;
my $astman = new Asterisk::Manager;
$astman-user('admin');
$astman-secret('secret');
$astman-host('localhost');
$astman-connect || die $astman-error . \n;
$astman-setcallback('Hangup', \hangup_callback);
$astman-setcallback('DEFAULT', \default_callback);
print $astman-sendcommand( Action = 'Originate',
Callerid = SLOT1,
Channel = 'Zap/g1/getfax',
Exten = 'sendfax',
Context = 'Outgoing',
Priority = '1' );
$astman-eventloop;
$astman-disconnect;
sub hangup_callback {
printf(hangup callback\n);
}
sub default_callback {
my (%stuff) = @_;
foreach (keys %stuff) {
printf(%s: %s\n, $_, $stuff{$_});
}
printf(\n);
}
--- RESULT WHEN RUNNING THE SCRIPT ---
EventNewchannelChannelZap/3-1StateRsrvdCallerIDunknownUniqueid1131547210.4CallerID:
SLOT1
Event: Newcallerid
Uniqueid: 1131547210.4
Channel: Zap/3-1
CallerID: SLOT1
Event: Newcallerid
Uniqueid: 1131547210.4
Channel: Zap/3-1
CallerID: SLOT1
Event: Newstate
Uniqueid: 1131547210.4
Channel: Zap/3-1
State: Dialing
CallerID: unknown
Event: Newchannel
Uniqueid: 1131547210.5
Channel: Zap/34-1
State: Ring
Event: Newexten
Channel: Zap/34-1
Context: Incoming
Extension: getfax
Application: SetVar
Uniqueid: 1131547210.5
AppData: FAXFILE=/tmp/1131547210.5.tiff
Priority: 1
Event: Newexten
Channel: Zap/34-1
Context: Incoming
Extension: getfax
Uniqueid: 1131547210.5
Application: RxFAX
AppData: /tmp/1131547210.5.tiff
Priority: 2
CallerID: unknown
Event: Newstate
Channel: Zap/34-1
State: Up
Uniqueid: 1131547210.5
CallerID: SLOT1
Event: Newstate
Channel: Zap/3-1
State: Up
Uniqueid: 1131547210.4
Event: Newexten
Channel: Zap/3-1
Context: Outgoing
Extension: sendfax
Uniqueid: 1131547210.4
Application: SetVar
AppData: SENDFAX=/tmp/sendfax.tiff
Priority: 1
Event: Newexten
Channel: Zap/3-1
Context: Outgoing
Extension: sendfax
Uniqueid: 1131547210.4
Application: TxFAX
AppData: /tmp/sendfax.tiff|caller
Priority: 2
* Stays there
--- ZAPATA.CONF ---
[trunkgroups]
; define any trunk groups
[channels]
switchtype=euroisdn
;pridialplan=national
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=no
rxgain=0.0
txgain=0.0
;faxdetect=both
; Span 1
context=Outgoing
group=1
;signalling=pri_net
signalling=pri_cpe
channel = 1-15
channel = 17-31
; Span 2
context=Incoming
group=2
signalling=pri_net
;signalling=pri_cpe
channel = 32-46
channel = 48-62
--- ZAPTEL.CONF ---
#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=span num,timing,line build out (LBO),framing,coding[,yellow]
#
# The timing parameter determines the selection of primary, secondary, and
# so on sync sources. If this span should be considered a primary sync
# source, then give it a value of 1. For a secondary, use 2, and so on.
# To not use this as a sync source, just use 0
#
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1: 133-266 feet (DSX-1)
# 2: 266-399 feet (DSX-1)
# 3: 399-533 feet (DSX-1)
# 4: 533-655 feet (DSX-1)
# 5: -7.5db (CSU)
# 6: -15db (CSU)
# 7: -22.5db (CSU)
#
# The framing is one of d4 or esf for T1 or cas or ccs for E1
#
# Note: d4 could be referred to as sf or superframe
#
# The coding is one of ami or b8zs for T1 or ami or hdb3 for E1
#
# E1's may have the additional keyword crc4 to enable CRC4 checking
#
# If the keyword yellow follows, yellow alarm is transmitted when no
# channels are open.
#
#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4
#
# Next come the dynamic span definitions, in the form:
# dynamic=driver,address,numchans,timing
#
# Where driver is the name of the driver (e.g. eth), address is the
# driver specific address (like a MAC for eth), numchans is the number
# of channels, and timing is a timing priority, like for a
[Asterisk-Users] sorry for posting many times
Sorry for posting many times ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H263 algoritm in 1.2.0.rc1
I have just upgraded my server to Asterisk 1.2.0.rc1 from the beta1 release. Most seems to work just fine, except for endpoints trying to use h263 as video algorithm. Result: Audio is ok, video NOK. Anyone else with the same problem? Tips on how to fix it? Trond ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IM / presence asterisk-1.2-RC1
Hello, Does asterisk's team will improve IM and presence in asterisk-1.2 ! Send Sip MESSAGE is impossible. When the buddies status change nothing is happened. How asterisk's team plan to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 trancoder
Hi asterisk lovers, Does anyone know a good trancoder to produce g729 files from gsm or wav. Regards, Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CentOS vs. Vanilla Kernel
It's all in the email, just look a little lower ;) hint: HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 Jason Walker wrote: Julian - What hardware are you using? Proc, RAM, SCSI or IDE, etc. The reason I ask is that I have multiple hardware platforms, all on FC1 or FC4, and none of them hit 100% for each IRQ. I am usually in the high 98% with the occasional 100% on P3 servers (give or take 1 Gig RAM, 1 Gig CPU). Two servers are dual p3 1.2 with 2 Gigs Ram. Since CentOS is brought up, maybe my OS is the culprit...far fetched? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, November 10, 2005 12:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CentOS vs. Vanilla Kernel Not a problem that I've had :) Linux foxtrot.tessera.co.uk 2.6.9-22.0.1.EL #1 Thu Oct 27 12:26:11 CDT 2005 i686 i686 i386 GNU/Linux Opened pseudo zap interface, measuring accuracy... 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% --- Results after 24 passes --- Best: 100.00 -- Worst: 100.00 -- Average: 100.00 [EMAIL PROTECTED] zaptel]# Julian. [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote on 11/07/2005 01:17:31 PM: HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Will you be using Zaptel hardware? The only way I can get zttest results of 100% is with a CentOS 2.4 kernel. Any CentOS 2.6 kernel I've tried (Uni, SMP, with IOAPIC enabled or disabled) gave me 99.99% at best... Tim Massey -- -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TDM400 FXO Screech
A nasty screech. That's what callers here sometimes when they dial into my FXO port from the PSTN. But usually, it works OK. Is this common? That was fairly common on the original TDM cards (rev E/F) with older drivers. The problem would usually show up after the card has been in operation for a week or so (time varied). Stopping asterisk and restarting the driver usually corrected the problem. If your card is an early revision, call digium support and explain what's happening and they'll RMA the card. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] H263 algoritm in 1.2.0.rc1
Please post a bug on bugs.digium.com with a full sip debug trace with verbosity of at least 4 and a debug level of at least 4 so we can track down and fix any possible bug before 1.2 is released. Thanks. On 11/10/05, Trond Andersen [EMAIL PROTECTED] wrote: I have just upgraded my server to Asterisk 1.2.0.rc1 from the beta1 release. Most seems to work just fine, except for endpoints trying to use h263 as video algorithm. Result: Audio is ok, video NOK. Anyone else with the same problem? Tips on how to fix it? Trond ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Playtone on answering the phone
Quoting Matt Riddell [EMAIL PROTECTED]: They are not DTMF tones they are 1100Hz, 400Hz and 440Hz tones, used in call shop systems. They monitor call progress and trigger billing. Regards Obelix Obelix wrote: Quoting Matt Riddell [EMAIL PROTECTED]: Is there a way of converting the play tone to a gsm file which can be played using the A option? Sure, if you send me the dtmf tones you need and I'll mail you some gsm files. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Zaptel T1 Timing Source
Yes, but the problem is, I think from a T1 theoretical perspective, that because the T1s are from different providers, their timings may be different. I would assume that I need to be able specify a timing source per provider. Correct? No, all real telco's will sync against a higher level clock, so they are already in sync. You only need to pick one that you sync from; all others become alternates should the primary source fail. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IM / presence asterisk-1.2-RC1
Harry, The monitoring of buddies on Polycom phones is possible with the release candidate for v1.2. We've asked for a sip debug/trace from you to try and troubleshoot your problem, and you haven't provided that to date. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: Hello, Does asterisk's team will improve IM and presence in asterisk-1.2 ! Send Sip MESSAGE is impossible. When the buddies status change nothing is happened. How asterisk's team plan to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Zaptel T1 Timing Source
Rich Adamson wrote: Yes, but the problem is, I think from a T1 theoretical perspective, that because the T1s are from different providers, their timings may be different. I would assume that I need to be able specify a timing source per provider. Correct? No, all real telco's will sync against a higher level clock, so they are already in sync. You only need to pick one that you sync from; all others become alternates should the primary source fail. Public telephone exchanges normally contain an atomic (rhubidium) clock. The clock in a T1 or E1 from a telco is, therefore, extremely accurate. Even if two telcos in different parts of the world don't sync together (some do, and some don't) their clocks are still, essentially, in sync. This also means that if you derive all your VoIP timing from a public E1 or T1 clock, and a box the other side of the world does the same, the VoIP packets exchanged between them should show no noticable timing shifts. :-) Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ITS Telecom Hardware
Hi, Has anybody tried using ITS Telecom Analog::GSM gateway devices with * ? http://www.its-tel.com/main/home/doc.asp?mCatID=1977mCatPID=1972tpMID=0 They appear to be very favourably priced... Rgds Pete ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ericsson pabx and digium card TE110P
Olivier; Merci pour ta réponse, le problème était au niveau de mon zapata.conf, il fallait que je rajoute la fonction overlap=yes, parcontre je ne passe pas par France Telecom je remplace plutot France telecom, en gros: Pabx ericssonconnecté avecsa carte E1 directement sur asterisk avec la carte E1 TE110P, parcontre j'ai un autre problème quand l'utilisateur compose rapidement son numéro asterisk route que quelque digits et non pas la totalité, je pense que mon problème la et dans le fichier extension.conf, car si le users compose la totalité du numéro durant 5s l'appel passe sans aucun problème, je ne sais pas comment lui dire sur le fichier extension.conf d'attendre plus de 5s avant d'envoyer l'appel :-( ma régle de routage sur le fichier extension est : exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) Merci de ton aide Si le schéma est bien le suivant :E1(Frannce-telecom) -- PABX --- Asterisk.Et que les appels entrants sont transmis à * avec seulement 4 digits, c'est plus un problème d'opérateur que de PABX.En effet, traditionnellement France-Telecom n'envoie sur une E1 louéeaux entreprises que les 4 derniers digits des appels entrants. Ce qui engénéral permet de savoir vers quel poste envoyer l'appel entrant. Cordialement,Le mar 08/11/2005 à 06:33, Chee Foong a écrit : Did you verify with the pbx engineer on how many digits the pbx are sending? Usually this should be the setting in the pbx. CCF -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]]On Behalf Of vador loupe Sent: Sunday, October 30, 2005 10:23 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] ericsson pabx and digium card TE110P Hi; Could some one help me: I have a problème to make call from my pabx ericsson i receive juste 4 digit from ericssonto my asterisk any idea??? thanks zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=fr defaultzone=fr zapata.conf: [channels] language=fr switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown hidecallerid=no threewaycalling=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no context=entrant group = 0 signalling=pri_net channel = 1-15 channel = 17-31 __ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 DHCP and Polycom boot server
I've been trying to set up my Polycom phones to get the boot server info (tftp-server-address) from DHCP on a Cisco router. I've previously just specified it manually on the phone, and that works well enough, but I need to change now (because of the number and geographic locations of the phones). I can actually get it to work just fine (using option 66 on the Cisco router), if I change the DHCP menu on the Polycom phone to show BootSrv Type: String. That's great, but that's not a default setting, and I don't want to have to change any settings on the phone. I want the phones to be able to provision fully, out-of-the-box, with nothing but the info from DHCP. If I leave the default setting (BootSrv Type: IP Address), and tell the Cisco router to send the boot serverinfo as an IP rather than as a string, nothing happens. The phone just says Could not contact boot server, using existing configuration, but according to the FTP logs and ethereal, the phone doesn't actually try to contact the boot server at all. I've tried various version of the bootrom, but nothing has worked so far. Has anybody gotten this to work? (Cisco router DHCP and Polycom boot server) If you want this to be anywhere near reliable, then consider switching from tftp to ftp per the Polycom warnings. The phones want to check the timestamps of various files to determine whether to read/implement that file, and tftp does not support that. You might get it to work the first time, but subsequent changes to those files will not be read by the phone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP NAT register
I have solved one part of the problem. I'm able to register. I'm able to call SIP phones and I can hear them. The only problem is that they can't hear me. So, this is the situation. Softphone_1 (on public IP) = Internet = Router = * (private IP) = Softphone_2 (private IP) SP_1 can call and hear what SP_2 is saying. SP_2 can't hear what SP_1 is saying. The guy that works with firewall says that UDP ports 1-10005 are opened (the ports that I have configured in rtp.conf file). Can it be anything else than firewall? I don't want bother him if I'm not totally sure that the problem is in firewall. Thank you. Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parčina Sent: 10. studeni 2005 9:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP NAT register I'm unable to register soft phone on * that is behind NAT. I have SP on public address 195.29.109.0 which is dynamically changed. * is in private address 10.0.0.81 that is behind NAT on address xxx.xxx.xxx.xxx When I try to register this is message that I receive on * CLI # Testing 195.29.109.0 with 10.0.0.0 Target address 195.29.109.0 is not local, substituing externip And that is all. When I enter sip show peers I get Name/username HostDyn Nat ACL Mask PortStatus 2150/2150 (Unspecified) D 255.255.255.255 0 Unmonitored This is how my sip.conf looks like. [general] nat=yes externip = xxx.xxx.xxx.xxx; here I have my public IP fromdomain = mydomain.hr localnet = 10.0.0.0/255.255.255.0 port=5060 bindaddr=0.0.0.0 context=sip srvlookup=yes dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw allow=alaw musicclass=default [2150] type=friend username=2150 secret=2150 host=dynamic mailbox=2150 Have I done something wrong or is there I haven't done? Please help. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IM / presence asterisk-1.2-RC1
I did it !?
//
Connected to Asterisk 1.2.0-rc1 currently running on
serveur1 (pid = 1125)
Verbosity is at least 4
serveur1*CLI sip show subscriptions
Peer UserCall ID Extension
Last state Type
192.168.0.21 86 f1682d8d-8f 84
Idle xpidf+xml
192.168.0.21 86 5f32aec-95b 85
Idle xpidf+xml
192.168.0.20 84 cb424ae1-e4 86
Idle xpidf+xml
192.168.0.20 84 715fac66-a9 87
Idle xpidf+xml
4 active SIP subscriptions
serveur1*CLI
//
serveur1*CLI sip show peers
Name/username HostDyn Nat ACL
Port Status
87/87 192.168.0.21 D N
5060 OK (84 ms)
86/86 192.168.0.21 D N
5060 OK (97 ms)
85/85 192.168.0.20 D N
5060 OK (87 ms)
84/84 192.168.0.20 D N
5060 OK (96 ms)
4 sip peers [4 online , 0 offline]
serveur1*CLI
///
my sip.conf:
[general]
context=local ; Default context for incoming calls
; if asterisk was compiled with OSP support.
realm=nxs.yi.org; Realm for digest authentication
; defaults to asterisk
; Realms MUST be globally unique according to
RFC
3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0
binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
tos=lowdelay;
lowdelay,throughput,reliability,mincost,none
maxexpirey=3600 ; Max length of incoming
registration we allow
defaultexpirey=1000 ; Default length of
incoming/outoing registration
allow=all ; First disallow all codecs
musicclass=default ; Sets the default music on hold
class for all SIP calls
language=fr ; Default language setting for all
users/peers
rtptimeout=60 ; Terminate call if 60 seconds of no
RTP activity
tpholdtimeout=300 ; Terminate call if 300 seconds of
no RTP activity
useragent=Asterisk PBX ; Allows you to change the
user agent string
dtmfmode = rfc2833 ; Set default dtmfmode for sending
DTMF. Default: rfc2833
promiscredir = no ; If yes, allows 302
or REDIR to non-local SIP address
nat=yes
qualify=500
[84]
type=friend
secret=84
context=local
host=dynamic
mailbox=84
allow=all
[85]
type=friend
secret=85
context=local
host=dynamic
mailbox=85
allow=all
[86]
type=friend
secret=86
context=local
host=dynamic
mailbox=86
allow=all
[87]
type=friend
secret=87
context=local
host=dynamic
mailbox=87
allow=all
//
my extension.conf
;
[general]
;
static=yes
writeprotect=no
switch = Realtime/[EMAIL PROTECTED]
;
[globals]
;
[local]
exten = 80,1,Answer
exten = 80,2,Dial(Zap/g2,14)
exten = 80,3,VoiceMail(u80)
exten = 80,103,VoiceMail(b80)
exten = 84,hint,Sip/84
exten = 84,1,Answer
exten = 84,2,Dial(Sip/84,10)
exten = 84,3,VoiceMail(u84)
exten = 84,103,VoiceMail(b84)
exten = 85,hint,Sip/85
exten = 85,1,Answer
exten = 85,2,Dial(Sip/85,10)
exten = 85,3,VoiceMail(u85)
exten = 85,103,VoiceMail(b85)
exten = 86,hint,Sip/86
exten = 86,1,Answer
exten = 86,2,Dial(Sip/86,10)
exten = 86,3,VoiceMail(u86)
exten = 86,103,VoiceMail(b86)
exten = 87,hint,Sip/87
exten = 87,1,Answer
exten = 87,2,Dial(Sip/87,10)
exten = 87,3,VoiceMail(u87)
exten = 87,103,VoiceMail(b87)
include = mailbox
include = apps
include = pstn
[mailbox]
exten = 700,1,VoiceMailMain()
[pstn]
exten = s,1,Answer
exten = s,2,Goto(local,84,1)
include = outgoing-pstn
[outgoing-pstn]
ingnorepat = 0
exten = _0,1,Dial(Zap/g1/${EXTEN:1})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1})
exten = _0.,3,Hangup
//
Regards
Harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
Harry,
The monitoring of buddies on Polycom phones is
possible with the
release candidate for v1.2. We've asked for a sip
debug/trace from you
to try and troubleshoot your problem, and you
haven't provided that to
date.
On 11/10/05, harry gaillac [EMAIL PROTECTED]
wrote:
Hello,
Does asterisk's team will improve IM and presence
in
asterisk-1.2 !
Send Sip MESSAGE is impossible.
When the buddies status change nothing is
happened.
How asterisk's team plan to solve this problem ?
Regards
Harry
___
Appel audio GRATUIT
Re: [Asterisk-Users] Asterisk Crashing (high load issues)
I would say your best bet is to change your system into a distributed dialing system. We did this with Vicidial and have installations on multiple servers with over 100 agents all working off of the same lists and campaigns. A distributed system will also allow for more redundancy and less total downtime if one server goes down. We noticed the same kind of limitations you are and now do a max of 40 agents per server, and when we need more capacity we just add another server. MATT--- On 11/10/05, Kyle Hagan [EMAIL PROTECTED] wrote: Kyle Hagan wrote: We purchased a new Dual Xeon 3ghz, 2gb ram to upgrade our 3ghz Pentium 1gb ram, that has been having load issues due to our growing company. We are having problems... We use a predictive dialer that we custom programmed in perl. It basically drops, moves, files into the callout directory and uses queues to transfer to agents when someone picks up. Oh, we are running HEAD version. Kyle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues with one Agent set to DND
I have a question. Is there any way to have a caller entering a Queue to go to voicemail if there is only one Agent and that extension has the phone set to DND? We have one extension that is the primary service technician and have it set to always be a member / logged in, so he cannot just logout when he goes to lunch. The phone rings when he is at lunch and drives people crazy. I tried setting DND on, when a call comes into the queue it shows his extension as do not disturb and sets it to BUSY, but the call is still on hold. I would think that if there is only one agent and that agent is set to DND the call should proceed as if there were no agents logged in. - James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] R2-Digital (Q.421)
Hi, I tried hunting for a little more info. I think all that happens with this is they use the Q.421 spec for handling the ABCD bits, and then simply send the DNIS through as DTMF after the seize if acknowledged. That means they loose some of the functionality of real R2 signalling - e.g. no busy, NU, or congestion detection. It wouldn't take a lot of work to implement that. Regards, Steve Steve Underwood wrote: Hi Jesus, The Cisco kit, and one or two other products, offer an R2 digital using DTMF mode, but this is the first time I have heard of it being used. The spec for this is definitely not Q.421. That spec does not mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU specs, as far as I can tell. Without a spec, or any equipment to play with, there isn't a lot I can do right now. Steve Jesus Mogollon wrote: Hi Steve: Thanks for your help. I really appreciate it.. My provider is CANTV in Venezuela. There's a venezuelan variant in the code and I'm using that. Incoming works perfectly, outgoing is not working. I'm being told that incoming is MFCR2 but outgoing is R2-Digital with DNIS DTMF. There is a Cisco router working and it's using the following: r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS What's the equivalent in libmfcr2 and Unicall? Again, thank you for your help and your code! Jesus Mogollon 2005/11/5, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi Jesus, FX is not a variant of R2. It is a completely different signalling protocol. This means your service provider is using R2 for some of your channels, and providing all your incoming calls on those channels. It is use FX signalling for other channels, and you must make your outgoing calls there. Someone else told be about a similar configuration. I think they were able to use chan_zap for the other channels, and make use of its FX signalling features. I am not sure how that works, as FX signalling over E1s is far from standardised. Regards, Steve Jesus Mogollon wrote: Steve: That's exactly what I'm using. Incoming calls work like a charm but when I try calling I get a protocol error. My provider says that for outgoing I need to use fx signalling. I see that in unicall.conf there's such a thing as protocolvariant=fx but if I uncomment that line, unicall gives me an error. Any ideas? Thanks for your help... 2005/11/4, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Jesus Mogollon wrote: Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are to use Q421 R2 Digital signalling. Any help is appreciated. Jesus Mogollon See http://www.soft-switch.org http://www.soft-switch.org Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 p2p
Hello I am still new to Asterisk, but looking at some products to offer small and medium sized buisnesses. Is it possibel to have the sip ends talk directly to eachother? Have authorisation and call setup on the asterisk, but leave the actual conversation p2p? BR Amund Nygaard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] DTMF method AVT
What kind of DTMF method signaling is AVT ? The sipura admin manual refers to AVT as a.k.a rfc2833. My Sippura seems to support only InBand, AVT, INFO, InBand+Info, Auto INFO does not work with Asterisks voicemail system so it is useless for me. Auto works just fine for me. InBand - I have a problem with this one when I try to connect to a bank automated systems (some of them don't recognize this one). When I set asterisk to rfc2833 dialing out works perfectly but when somebody tries to call me and dial an internal extension, it is a 50/50 (sometimes it works and sometimes it gives me invalid extension). I'm using Asterisk-1.0.8 Upgrade to something recent. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ITS Telecom Hardware
how much is that per pc.? On 11/10/05, Pete Barnwell [EMAIL PROTECTED] wrote: Hi, Has anybody tried using ITS Telecom Analog::GSM gateway devices with * ? http://www.its-tel.com/main/home/doc.asp?mCatID=1977mCatPID=1972tpMID=0 They appear to be very favourably priced... Rgds Pete ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New revision of my MFC/R2 software available
Hi, Users of my MFC/R2 software may be interested to know that new versions are available. These fix a bug where a timer was not always correctly cancelled. The result could be the locking up of a channel. You can download the updates from http://www.soft-switch.org Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura 2000
I followed your steps to the letter but after resetting to factory defaults unfortunately it still doesn't record the configuration changes I do. 2005/11/9, Adam Moffett [EMAIL PROTECTED]: If you unplug the ethernet cable on a Sipura SPA and then reset the power it'll boot up in a diagnostic mode. When you pick up the phone that's connected to it you'll get a dialtone and there are speical codes you can dial to do various things. It would be helpful if you told us where you got the box from. If it was used with an itsp, they have probably configured it to disallow config changes. Sipura has provided a number of ways for itsp's to secure their products and it is very possible to disable changes for each config parameter in the box. Other options in the box force it to resync the configuration with the itsp after any changes are made, and also check for changes after xxx number of seconds (I think the default was 3600 seconds). If you can't make changes in admin mode, then you either have a box that was preconfigured by an itsp, or, possibly a defective box. Someone else suggested you reset the box to factory defaults by using an attached touchtone phone. If you can't do that either, then its fairly obvious the box was secured by your previous itsp. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 and VPN
Anyone out there got a SIP phone (mine's a Cisco 7940) to work through a VPN with a Netscreen 5gt? It has always worked for me with any ScreenOS version 4.x. I had the need to upgrade it to ScreenOS 5.x and it breaks the phone. Here's the goofy part, it works enough to still register with the phone system and check if there is voicemail waiting. But I get no audio on outbound calls. Inbound calls seem to work OK. The netscreen is not in NAT mode, but in route mode. When the phone system talks to the phone at home, it uses the home LAN address. In debug mode, the phone system doesn't seem to notice anything is wrong. I don't know if this means anything or not, but... On the phone system, if I do a nmap -sU -p5060 homephoneip it comes back with the port is open. If I do the same thing from my home PC and nmap the SIP port on the phone system, it comes back open|filtered which I think means no UDP packet is returning. SSH to the phone system works fine from home. I also noticed that NTP os broken on the phone, so something is wrong with UDP. I found a really good article from someone having the same issues but it made no difference for me. I have a support contract through Juniper, but they still have not found any resolution. Here's the sip.conf section. I tried some variations with canreinvite and some things, but it didn't help. This has worked for me over a year like this. Anyone got any ideas? Thanks! Mark [1426] type=friend username=123456 secret=123456 host=dynamic ;canreinvite=no ;disallow=all ;allow=ulaw,alaw ;dtmfmode=inband ;nat=never context=office [EMAIL PROTECTED] linelabel=First Last callerid=First Last 1426 line = 1426 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 p2p
yes From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Amund Nygaard Sent: Thursday, November 10, 2005 8:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call p2p Hello I am still new to Asterisk, but looking at some products to offer small and medium sized buisnesses. Is it possibel to have the sip ends talk directly to eachother? Have authorisation and call setup on the asterisk, but leave the actual conversation p2p? BR Amund Nygaard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Redirect/Transfer
I have a question which may be about the SIP protocol, or may be about SIP features supported in Asterisk, I don't know. Let's say I have three Asterisk boxes, A, B and C, which pass calls to each other using SIP. A call comes into box A from somewhere, and A determines that the call should be routed to box B. When box B receives the call, it does some operations internally, and decides that in fact the call should be handled by box C instead. I know B could easily dial a new call to C and pass the contents of the call back and forth between A and C. However, is it possible for box B to redirect the original call to box C so that A is talking directly to C, and B is no longer involved? In fact, A and C might not be Asterisk, but other kinds of SIP switch. Box B definitely is Asterisk, and is the box over which I have control. Thanks in advance for any ideas. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Queues with one Agent set to DND
James Armstrong wrote: I have a question. Is there any way to have a caller entering a Queue to go to voicemail if there is only one Agent and that extension has the phone set to DND? We have one extension that is the primary service technician and have it set to always be a member / logged in, so he cannot just logout when he goes to lunch. The phone rings when he is at lunch and drives people crazy. I tried setting DND on, when a call comes into the queue it shows his extension as do not disturb and sets it to BUSY, but the call is still on hold. I would think that if there is only one agent and that agent is set to DND the call should proceed as if there were no agents logged in. - James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users James, You should try to use normal sip channels instead of agents and define them as members of a queue, so you are able to set a voicemail when busy/unavailable. Check http://www.voip-info.org/wiki/view/Asterisk+call+queues * *// Members are those channels that are active answering the Queue. It can be agents or normal channels, like sip/snom23 Regards, Juan Manuel Coronado Z. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 still no rtp traffic
Hi all, i'm still experiencing a one way call only between a ipPhone and an analog one through a oh323 channel between my asterisk and a Nortel GK. Doing some sniffing and some debug with ethereal and tcpump i can say (i hope, as newby to say the right thing) that i can't see any rtp traffic between the asterisk and the nortel. In the analog phone (in the outside telecom world) i can't ear nothing said in the ipPhone. Viceversa in the ipPhone (Mitel one) i can ear the voice comming from the outside world. In my sip.conf [419] callerid=0432281316 TEST test 419 type=friend username=419 secret=password host=dynamic nat=yes canreinvite=no reinvite=no disallow=all allow=ulaw allow=gsm ;allow=alaw dtmfmode=rfc2833 context=out callgroup=1 pickupgroup=1 There's no rtp traffic from the phone or from the asterisk to the GK. The GK stays on the intranet even if it has a internet looking ip. ipPhone 10.24.3.40 asterisk 10.24.2.253 GK 80.74.178.196 Issuing on asterisk rtp debug [2]WrapH323EndPoint::AnswerCall: Request to answer call ip$80.74.178.196:34404/1169 Got RTP packet from 10.24.3.40:20012 (type 0, seq 14, ts -1120604096, len 160) [2]WrapH323EndPoint::AnswerCall: Call answered [ip$80.74.178.196:34404/1169] Got RTP packet from 10.24.3.40:20012 (type 0, seq 15, ts -1120603936, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 16, ts -1120603776, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [3]WrapH323EndPoint::OpenAudioChannel: Direction = RECODER, Buffer = 320 [2]WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8, TimeUnits 8 [2]WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 [2]WrapH323EndPoint::OpenAudioChannel: Packet size: 160 [2]WrapH323EndPoint::OpenAudioChannel: Frames per packet: 20 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k [3]WrapH323EndPoint::OpenAudioChannel: The sound channel with the application is asterisk-oh323(fd=42) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 42, mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize 160 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound channel Asterisk for recording using 1x320 byte buffers. [3]WrapH323Connection::OnEstablished: WrapH323Connection [ip$80.74.178.196:34404/1169] established (FastStartDisabled/H245Tunneling) [3]WrapH323EndPoint::OnConnectionEstablished: Connection [ip$80.74.178.196:34404/1169] established. [3]WrapH323EndPoint::GetConnectionInfo: [ip$80.74.178.196:34404/1169] RTP Media: 10.24.2.253:21002-0.0.0.0:0 Got RTP packet from 10.24.3.40:20012 (type 0, seq 17, ts -1120603616, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 18, ts -1120603456, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] Got RTP packet from 10.24.3.40:20012 (type 0, seq 19, ts -1120603296, len 160) [3]WrapH323EndPoint::OpenAudioChannel: Direction = PLAYER, Buffer = 320 [2]WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8, TimeUnits 8 [2]WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 [2]WrapH323EndPoint::OpenAudioChannel: Packet size: 160 [2]WrapH323EndPoint::OpenAudioChannel: Frames per packet: 20 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k [3]WrapH323EndPoint::OpenAudioChannel: The sound channel with the application is asterisk-oh323(fd=40) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 40, mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize 160 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound channel Asterisk for playing using 1x320 byte buffers. [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26203, ts 160, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [5]PAsteriskSoundChannel::Read: Data read [320 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 20, ts -1120603136, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26204, ts 320, len 160) [5]PAsteriskSoundChannel::Read: Data read [320 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 21, ts -1120602976, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet
Re: [Asterisk-Users] ad hoc conferencing-reg
I've think I've been working on the same thing. Many SIP phones have a
built in conferencing feature...but they may not all work the same and
may have all different instructions. So doing it in asterisk is
preferable to me so I can give users one set of instructions for it.
It's not a simple straightforward thing like threewaycalling= on in
zapata.conf. For SIP you have to create an extension that executes a
macro which dynamically creates a meetme conference or adds a caller to
an existing one. Then you create an extension that goes to that macro.
Person A can then call person B, transfer person B to the conference
extension, call Person C, transfer Person C to the conference extension,
then call the conference extension to add themselves to the conference.
At least that's the ideaI haven't quite got it working perfectly ;)
First I enabled blindxfer in features.conf
Then in extensions.conf created an extension for conferences...it's 999
for me but it could be anything.
Then I added this NWayCall macro below. This is a modified version of
something I saw on Voip-info.org. When this macro is called, it first
checks to see if the caller was transfered to it or called the extension
directly. If they were transfered here, it gets the name of the SIP
user that transfered them, then checks to see if a conference with that
name exists. If the conference doesn't exist it creates one, otherwise
it adds the transferred person to the conference. If you weren't
transfered to this extension (as in, you called it directly) it adds you
to the conference.
Last time I tried this was last week, and I've been busy with other
things since. When I tried it, it worked but it was very twitchy. Any
improvements you can come up with would be appreciated.
Or if anyone has an entirely better way to do this, I'm listening.
exten = 999,1,Macro(NWayCall)
[macro-NWayCall]
exten = s,1,Noop(${BLINDTRANSFER})
exten = s,2,Gotoif($[${BLINDTRANSFER} !=
]?s-TRANSFERED|1:s-NOTTRANSFERED|1)
exten = s-TRANSFERED,1,GoTo(s-SIPHOLDER|1)
exten = s-SIPHOLDER,1,Cut(CONFHOLDER=BLINDTRANSFER,/,2)
exten = s-SIPHOLDER,2,Cut(CONFHOLDER=CONFHOLDER,-,1)
exten = s-SIPHOLDER,3,Goto(s-USERJOIN|1)
exten = s-USERJOIN,1,MeetMe(${CONFHOLDER},dwxM)
exten = s-USERJOIN,2,Hangup()
exten = s-NOTTRANSFERED,1,GoTO(s-SIP2HOLDER|1)
exten = s-SIP2HOLDER,1,Cut(CONFHOLDER=CHANNEL,/,2)
exten = s-SIP2HOLDER,2,Cut(CONFHOLDER=CONFHOLDER,-,1)
exten = s-SIP2HOLDER,3,Goto(s-CHECKCONFEXIST|1)
exten = s-CHECKCONFEXIST,1,MeetmeCount(${CONFHOLDER},CONFCOUNT)
exten = s-CHECKCONFEXIST,2,GotoIf($[${CONFCOUNT} =
]?s-INVALID|1:s-CONFNOTEMPTY|1)
exten = s-CONFNOTEMPTY,1,Gotoif($[${CONFCOUNT}
0]?s-HOLDERJOIN|1:s-INVALID|1)
exten = s-HOLDERJOIN,1,Meetme(${CONFHOLDER},qdAx)
exten = s-INVALID,1,Playtones(info)
exten = s-INVALID,2,Wait(10)
exten = s-INVALID,3,Hangup()
Hi all
How to configure adhoc conferencing in asterisk for
sip phones.pls give me if any document for that.
regards
ramakrishnan.n
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Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1
This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity tpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=Asterisk PBX ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP Redirect/Transfer
Olle has said he has a working patch for this scenario, but it will be a couple of weeks yet before it's ready to be merged into the HEAD tree so it will be a post 1.2 thing. On 11/10/05, Tony Mountifield [EMAIL PROTECTED] wrote: I have a question which may be about the SIP protocol, or may be about SIP features supported in Asterisk, I don't know. Let's say I have three Asterisk boxes, A, B and C, which pass calls to each other using SIP. A call comes into box A from somewhere, and A determines that the call should be routed to box B. When box B receives the call, it does some operations internally, and decides that in fact the call should be handled by box C instead. I know B could easily dial a new call to C and pass the contents of the call back and forth between A and C. However, is it possible for box B to redirect the original call to box C so that A is talking directly to C, and B is no longer involved? In fact, A and C might not be Asterisk, but other kinds of SIP switch. Box B definitely is Asterisk, and is the box over which I have control. Thanks in advance for any ideas. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP and VPN
ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have listed below. set policcy id 1001 from Trust to Trust Local Remote SIP permit log count set policy id 1001 application IGNORE set policy id 1002 from Trust to Trust Remote Local SIP permit log count set policy id 1002 application IGNORE I am running 5.2r1 without any issues but I have turned off any application deep scanning. unset alg sql unset alg q931 unset alg h245 unset alg ras unset alg sip -Chip -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Johnson Sent: Thursday, November 10, 2005 9:15 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP and VPN Anyone out there got a SIP phone (mine's a Cisco 7940) to work through a VPN with a Netscreen 5gt? It has always worked for me with any ScreenOS version 4.x. I had the need to upgrade it to ScreenOS 5.x and it breaks the phone. Here's the goofy part, it works enough to still register with the phone system and check if there is voicemail waiting. But I get no audio on outbound calls. Inbound calls seem to work OK. The netscreen is not in NAT mode, but in route mode. When the phone system talks to the phone at home, it uses the home LAN address. In debug mode, the phone system doesn't seem to notice anything is wrong. I don't know if this means anything or not, but... On the phone system, if I do a nmap -sU -p5060 homephoneip it comes back with the port is open. If I do the same thing from my home PC and nmap the SIP port on the phone system, it comes back open|filtered which I think means no UDP packet is returning. SSH to the phone system works fine from home. I also noticed that NTP os broken on the phone, so something is wrong with UDP. I found a really good article from someone having the same issues but it made no difference for me. I have a support contract through Juniper, but they still have not found any resolution. Here's the sip.conf section. I tried some variations with canreinvite and some things, but it didn't help. This has worked for me over a year like this. Anyone got any ideas? Thanks! Mark [1426] type=friend username=123456 secret=123456 host=dynamic ;canreinvite=no ;disallow=all ;allow=ulaw,alaw ;dtmfmode=inband ;nat=never context=office [EMAIL PROTECTED] linelabel=First Last callerid=First Last 1426 line = 1426 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Test environment (Windows Softphone)
Assuming an XP or 2003 box, I use the free xlite client. Create a user for each instance that you want to run. Right click on the shortcut and select run as... enter the username and password of the account, setup the settings for the phone, and repeat the process for each additional instance. If you have a well designed audio card driver, point the phones at the same input and feed a source into it. I have successfully had 48 of these running on a P4 box for testing. HTH BEN Marcus Deluigi (intern) wrote: Hi! I want to test asterisk with about 10 Softphones (on windows) with just one windows machine (in the best case). I'm thinking of a softphone, that I can run in multiple instances on one computer and that can be configured to play a file on an incoming call or to make a call after some time... Does anyone know such a softphone or can anybody give me another solution? Greetings, Marcus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] terminal emulation application that uses SIP
I am in search for a terminal emulation application like securecrt, putty, or penguin that can use SIP. It can be either linux or windows application. Thanks, Chip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ITS Telecom Hardware
Looks interesting. Analog single port only, though, so you would be subject to the vagrancies of a TDMXXX analog card. A VoiceBlue gateway is SIP so you can do IP-only until it hit the GSM network, and they aren't that expensive, $2500 US. My VoiceBlue is stuck in customs! Chomping at the bit to get it. s -Original Message- From: Pete Barnwell [mailto:[EMAIL PROTECTED] Sent: Thursday, November 10, 2005 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ITS Telecom Hardware Hi, Has anybody tried using ITS Telecom Analog::GSM gateway devices with * ? http://www.its-tel.com/main/home/doc.asp?mCatID=1977mCatPID=1972tpMID=0 They appear to be very favourably priced... Rgds Pete ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phones no longer register - except one?
Hi I've got an interesting problem. A few days ago (maybe even a week or two) all my sip phones lost registrations with my asterisk box. All that is but one. The asterisk box is out on the internet, I have two phones at my location and 1 at another separate location. The only phone that remains registered is an Integrated Networks IN002 (or something like that). This is at my location. I also have a grandstream GXP-2000 that will not register. This is also at my location. I have tried xlite and sjphone (on my desktop and mobile phone (via wireless) respectivley) to test, These also fail to register. I have a Budgetone 102 at another location which also fails to register. There is nothing on the command line apart from the IN002 phone registering and talking to the * server. It dials in and out fine. The only thing I can see that mine and the remote location have in common is the ISP that provides DSL (plusnet), and I was wondering if they were limiting traffic as they have recently announced their own telephony service. But I doubt it and if that was the case then why does the IN002 register and not the budget tone? Its a crazy paranoid theory, but I can't think of anything else. The * is 1.0.9 and was working perfectly when I upgraded from a CVS version. Any ideas - I must be missing something obvious - but I've not changed anything since the upgrade. Any why one phone? Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Queues with one Agent set to DND
Tried that. The queue has a static agent of SIP/107. When calling the queue it shows 107 as being BUSY (DND enabled). The caller just stays in the queue. What I really need is to have the caller stay in queue when the extension is busy (because that is that queues are all about), but have the caller leave the queue if DND is enabled in the database for that extension and there is only one extension as an Agent. - James Juan Manuel Coronado Z. wrote: James Armstrong wrote: I have a question. Is there any way to have a caller entering a Queue to go to voicemail if there is only one Agent and that extension has the phone set to DND? We have one extension that is the primary service technician and have it set to always be a member / logged in, so he cannot just logout when he goes to lunch. The phone rings when he is at lunch and drives people crazy. I tried setting DND on, when a call comes into the queue it shows his extension as do not disturb and sets it to BUSY, but the call is still on hold. I would think that if there is only one agent and that agent is set to DND the call should proceed as if there were no agents logged in. - James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users James, You should try to use normal sip channels instead of agents and define them as members of a queue, so you are able to set a voicemail when busy/unavailable. Check http://www.voip-info.org/wiki/view/Asterisk+call+queues * *// Members are those channels that are active answering the Queue. It can be agents or normal channels, like sip/snom23 Regards, Juan Manuel Coronado Z. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Simple Dial for If Busy Send to Voicemail
I am looking for a simple dial plan for if my zap channel is busy/unavailable send to Voicemail. I couldnt find anything simple online. -chip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Queues with one Agent set to DND
Hello James, you could approach this problem in many a way. I'd suggest to make your support guy log on to the queue using AgentCallBack and enforce joinempty=no in the queue itself. When your agent goes to lunch, he logs off and people cannot join the support queue anymore, so you move them to voicemail or play a message. This way you will also gather quite a number of stats on the support queue that will be very useful when people will keep complaining that they never get an answer. :-) Yours, l. On Thu, 10 Nov 2005 16:12:39 +0100, James Armstrong [EMAIL PROTECTED] wrote: Tried that. The queue has a static agent of SIP/107. When calling the queue it shows 107 as being BUSY (DND enabled). The caller just stays in the queue. What I really need is to have the caller stay in queue when the extension is busy (because that is that queues are all about), but have the caller leave the queue if DND is enabled in the database for that extension and there is only one extension as an Agent. - James Juan Manuel Coronado Z. wrote: James Armstrong wrote: I have a question. Is there any way to have a caller entering a Queue to go to voicemail if there is only one Agent and that extension has the phone set to DND? We have one extension that is the primary service technician and have it set to always be a member / logged in, so he cannot just logout when he goes to lunch. The phone rings when he is at lunch and drives people crazy. I tried setting DND on, when a call comes into the queue it shows his extension as do not disturb and sets it to BUSY, but the call is still on hold. I would think that if there is only one agent and that agent is set to DND the call should proceed as if there were no agents logged in. - James -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cell phone as digital trunk line
Goal: I would like to use the cheap cellular phone from my family share plan to add an * trunk. With this, nights and weekends are free as is cell(*) to cell. As an * noob, I have been scouring the threads for information on using a cell phone as a trunk (not a handset). Aside from using an analog connection: (Cell+cradel-to-*-FXO) or a fixed wireless adapter (like Telular), has anyone been successful connecting a cellphone via USB or Bluetooth? I've read a bit on the blue_chan but an mot 100% sure of its capabilities. thanks, paul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] R2-Digital (Q.421)
Just to clarify this in my head :-).. So... They are using E1/R2 (the R2 Digital)in fact, for all the line signaling (nothing unusual) The register signaling, that I was under impression would be MF in each timeslot (MFC5C in .br, not sure if the same in others), is in fact DTMF in this trunk, and only to provide DNIS ? (in Brazil R2, the register signaling has some collect call information and etc). Steve Underwood wrote: Hi, I tried hunting for a little more info. I think all that happens with this is they use the Q.421 spec for handling the ABCD bits, and then simply send the DNIS through as DTMF after the seize if acknowledged. That means they loose some of the functionality of real R2 signalling - e.g. no busy, NU, or congestion detection. It wouldn't take a lot of work to implement that. Regards, Steve Steve Underwood wrote: Hi Jesus, The Cisco kit, and one or two other products, offer an R2 digital using DTMF mode, but this is the first time I have heard of it being used. The spec for this is definitely not Q.421. That spec does not mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU specs, as far as I can tell. Without a spec, or any equipment to play with, there isn't a lot I can do right now. Steve Jesus Mogollon wrote: Hi Steve: Thanks for your help. I really appreciate it.. My provider is CANTV in Venezuela. There's a venezuelan variant in the code and I'm using that. Incoming works perfectly, outgoing is not working. I'm being told that incoming is MFCR2 but outgoing is R2-Digital with DNIS DTMF. There is a Cisco router working and it's using the following: r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS What's the equivalent in libmfcr2 and Unicall? Again, thank you for your help and your code! Jesus Mogollon 2005/11/5, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi Jesus, FX is not a variant of R2. It is a completely different signalling protocol. This means your service provider is using R2 for some of your channels, and providing all your incoming calls on those channels. It is use FX signalling for other channels, and you must make your outgoing calls there. Someone else told be about a similar configuration. I think they were able to use chan_zap for the other channels, and make use of its FX signalling features. I am not sure how that works, as FX signalling over E1s is far from standardised. Regards, Steve Jesus Mogollon wrote: Steve: That's exactly what I'm using. Incoming calls work like a charm but when I try calling I get a protocol error. My provider says that for outgoing I need to use fx signalling. I see that in unicall.conf there's such a thing as protocolvariant=fx but if I uncomment that line, unicall gives me an error. Any ideas? Thanks for your help... 2005/11/4, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Jesus Mogollon wrote: Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are to use Q421 R2 Digital signalling. Any help is appreciated. Jesus Mogollon See http://www.soft-switch.org http://www.soft-switch.org Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT'd SIP extension, no audio
Hi folks, I have an asterisk server behind a NAT'd gateway that is using iptables. Internally, I have no problems connecting to asterisk. I would like to be able to use a sip softphone from outside the gateway, and become an extension on my asterisk PBX. I have a laptop running X-Lite. When I connect it internally, the extension works fine. When I got outside my gateway, to another network on the internet (that is itself NAT'd behind a Belkin wiresless router), and I also change the sip extension in the asterisk dialplan to have nat=yes, then I hear no voice. Note that I can dial, and call will be connected; for example, if I dial into voicemail, I can enter my password and see in the asterisk logs that it went into the voice mail app. However I hear silence. If I dial the extension, it rings until it is picked up, and after that there is silence. Here are the iptables commands in my current setup (that don't have audio): $iptables -A FORWARD -i eth0 -p udp --dport 5060:5080 -j ACCEPT $iptables -t nat -A PREROUTING -i eth0 -p udp -d x.x.x.x --dport 5060:5080 -j DNAT --to-destination 192.168.1.40:5060:5080 $iptables -A FORWARD -i eth0 -p tcp --dport 5060:5080 -j ACCEPT $iptables -t nat -A PREROUTING -i eth0 -p tcp -d x.x.x.x --dport 5060:5080 -j DNAT --to-destination 192.168.1.40:5060:5080 $iptables -A FORWARD -i eth0 -p udp --dport 8000:2 -j ACCEPT $iptables -t nat -A PREROUTING -i eth0 -p udp -d x.x.x.x --dport 8000:2 -j DNAT --to-destination 192.168.1.40:8000:2 $iptables -A FORWARD -i eth0 -p tcp --dport 8000:2 -j ACCEPT $iptables -t nat -A PREROUTING -i eth0 -p tcp -d x.x.x.x --dport 8000:2 -j DNAT --to-destination 192.168.1.40:8000:2 192.168.1.40 is the address of my Asterisk server. x.x.x.x is my external IP address. I got these commands by copying commands I have successfully used to forward the ports used for VNC, and because I saw stuff on the internet that said I needed to hand the RTP ports as well as SIP. I have both UDB and TCP in there because I some people have told me UDP only was needed and others told me TCP was needed. Here is the section in sip_additional.conf that defines the extension: [908] username=908 type=friend secret= record_out=Always record_in=Always ;qualify=no qualify=150 port=5060 nat=yes ; for external extension only [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callgroup=1 pickupgroup=1 I added these lines to sip.conf: #added for external extensions externip=x.x.x.x localnet=192.168.1.0/255.255.255.0 Here is my rtp.conf: ; ; RTP Configuration ; [general] ; ; RTP start and RTP end configure start and end addresses ; rtpstart=1 rtpend=2 Why doesn't this work, and what can I do to fix it ? Should I post the logs of the X-Lite debug log and asterisk full log ? If I did a tcpdump on the NAT gateway while a call was attempted, would that help ? --Rob P.S. A copy of this post is at http://pastebin.ca/28236, from when I asked this on IRC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Simple Dial for If Busy Send to Voicemail
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail Is pretty simple. Replace the 102 priority with a call to voicemail and youre set. hth -Original Message- From: Pleasants Email Lists [mailto:[EMAIL PROTECTED] Sent: Thursday, November 10, 2005 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Simple Dial for If Busy Send to Voicemail I am looking for a simple dial plan for if my zap channel is busy/unavailable send to Voicemail. I couldnt find anything simple online. -chip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] R2-Digital (Q.421)
This seems to be what Cisco have implemented as r2-digital-dtmf-dnis. Cisco have quite a few other combinations of strange R2 related options. I can't imagine they are all really used. It seems this one is, though, in Venezuela Regards, Steve Julio Arruda wrote: Just to clarify this in my head :-).. So... They are using E1/R2 (the R2 Digital)in fact, for all the line signaling (nothing unusual) The register signaling, that I was under impression would be MF in each timeslot (MFC5C in .br, not sure if the same in others), is in fact DTMF in this trunk, and only to provide DNIS ? (in Brazil R2, the register signaling has some collect call information and etc). Steve Underwood wrote: Hi, I tried hunting for a little more info. I think all that happens with this is they use the Q.421 spec for handling the ABCD bits, and then simply send the DNIS through as DTMF after the seize if acknowledged. That means they loose some of the functionality of real R2 signalling - e.g. no busy, NU, or congestion detection. It wouldn't take a lot of work to implement that. Regards, Steve Steve Underwood wrote: Hi Jesus, The Cisco kit, and one or two other products, offer an R2 digital using DTMF mode, but this is the first time I have heard of it being used. The spec for this is definitely not Q.421. That spec does not mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU specs, as far as I can tell. Without a spec, or any equipment to play with, there isn't a lot I can do right now. Steve Jesus Mogollon wrote: Hi Steve: Thanks for your help. I really appreciate it.. My provider is CANTV in Venezuela. There's a venezuelan variant in the code and I'm using that. Incoming works perfectly, outgoing is not working. I'm being told that incoming is MFCR2 but outgoing is R2-Digital with DNIS DTMF. There is a Cisco router working and it's using the following: r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS What's the equivalent in libmfcr2 and Unicall? Again, thank you for your help and your code! Jesus Mogollon 2005/11/5, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi Jesus, FX is not a variant of R2. It is a completely different signalling protocol. This means your service provider is using R2 for some of your channels, and providing all your incoming calls on those channels. It is use FX signalling for other channels, and you must make your outgoing calls there. Someone else told be about a similar configuration. I think they were able to use chan_zap for the other channels, and make use of its FX signalling features. I am not sure how that works, as FX signalling over E1s is far from standardised. Regards, Steve Jesus Mogollon wrote: Steve: That's exactly what I'm using. Incoming calls work like a charm but when I try calling I get a protocol error. My provider says that for outgoing I need to use fx signalling. I see that in unicall.conf there's such a thing as protocolvariant=fx but if I uncomment that line, unicall gives me an error. Any ideas? Thanks for your help... 2005/11/4, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Jesus Mogollon wrote: Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are to use Q421 R2 Digital signalling. Any help is appreciated. Jesus Mogollon See http://www.soft-switch.org http://www.soft-switch.org Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound quality of the new BT 101 and 102 models
Hi. Im having sound quality problems using the new BT 101 and 102 models (the ones with solid colour bottoms like the gxp model). Im using firmware 1.0.6.7. Does anyone as the same problem with these new models? Sound quality has no cuts or noise. But the sound is much more lower and not clear and crystalline. Im using PCMA. Regards. André M. S. Rodrigues ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 revision of my MFC/R2 software available
Thx Steve! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Steve Underwood |Sent: Thursday, November 10, 2005 7:24 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] New revision of my MFC/R2 software available | |Hi, | |Users of my MFC/R2 software may be interested to know that new |versions are available. These fix a bug where a timer was not |always correctly cancelled. The result could be the locking up |of a channel. You can download the updates from |http://www.soft-switch.org | |Regards, |Steve | |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clarification on chan_modem.so module
Hi, Just so I am clear for version 1.2 has chan_modem.so been depreciated? That means I should also remove this module from loading in the modules.conf if I am using Asterisk 1.2 rc1. Do I have to do anything to replace this functionality (I do not really understand what chan_modem.so was used for other than it seemed to be linked to musiconhold...) Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Clarification on chan_modem.so module
On 11/10/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Just so I am clear for version 1.2 has chan_modem.so been depreciated? That means I should also remove this module from loading in the modules.conf if I am using Asterisk 1.2 rc1. Do I have to do anything to replace this functionality (I do not really understand what chan_modem.so was used for other than it seemed to be linked to musiconhold...) Yes. It has been deprecated. I believe it's original purpose was to be able to use the voice modems out there as FXO ports in Asterisk. You musiconhold will function without it. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P Zaptel config questions
I have a TE110P that I will be connecting to a T1 PRI. This seems pretty standard, but I am only using 7 channels for voice. Its a shared voice/data T1; 7 channels voice, 16 channels data and 1 D-chan, it comes into a telco router and is split into a voice PRI and an Ethernet connection. The 7 voice channels and one D chan are the only things on the backside PRI. Does zaptel need any special configuration for this sort of setup, or would the telco router handle the conversion and restriction? Would I still define the bchan as 1-23 and dchan as 24? Or would it be bchan=1-7 dchan=8? This seems like a pretty common product for most telco/ISPs to deliver to small businesses. I am a system/network admin by trade, not a telecom engineer, so please excuse my ignorance! I have not tried connecting it yet, as we need these phone lines during the course of the business day, but I will be testing it tonight so I am trying to ask any pertinent questions before Im up to my neck in it. :) -Ryan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nortel BCM 3.6 and Asterisk 1.0.9 via H.323
On voip-info.org there is a claim that asterisk and a BCM can interconnect via H.323. There is little on the page beyond setting the H.323 connection on the BCM to other. Hardware restrictions at the moment make the H.323 solution preferable to ISDN or SIP. I am using oh323. Every time that I implement this, all the IP connectivity on the BCM hangs within about 10 minutes. The BCM is connected to another BCM via H.323 as well (which goes down), and we have about 6 IP phones on this system. Has anyone worked this out, really? Can you provide me with a configuration/instructions? Regards, Mark McQuiggan _ This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. application/ms-tnef___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Intel Desktop MotherBoards *NOT* Unsuitable for Digium Boards
Colin Anderson wrote: Forrest: Any secondary effects you can see from running SP on an SMP kernel, any bitching from dmesg at boot? Cool hack. Nope... no other side effects I can tell. Of course, it boots like a SMP kernel (looking at the processor table and all). -forrest ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Clarification on chan_modem.so module
Hello BJ all , On Thu, 10 Nov 2005, BJ Weschke wrote: On 11/10/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Just so I am clear for version 1.2 has chan_modem.so been depreciated? That means I should also remove this module from loading in the modules.conf if I am using Asterisk 1.2 rc1. Do I have to do anything to replace this functionality (I do not really understand what chan_modem.so was used for other than it seemed to be linked to musiconhold...) Yes. It has been deprecated. I believe it's original purpose was to be able to use the voice modems out there as FXO ports in Asterisk. You musiconhold will function without it. Can you speak to , what other functionality has taken it place ? Some people out here use the chan_modem.so functionality . Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 Card
Is there some kind of limit to the number of TDM04B cards you can use in your Asterisk system (Red Hat 9, kernel 2.4, Asterisk CVS-v1-0-11-11/16/04-13:41:01))? I have 2 cards right now(rev B) with 8 analog lines connected to 8 FXO modules. I wanted to add 2 more analog lines but the third card (rev I) refuses to recognize the two new FXO modules. Digium have said their newer version TDM cards are backward-compatible. There is no problem with the PCI slot or IRQ. I'm using the motherboard (Asus P4P800-E) as recommended by Digium. Any ideas? Shaun Singh, Manager Travelwave 1655 Dufferin Street, Suite 201 Toronto, ON M6H 3L9 Tel: (416) 652-1212 Ext 101 Fax: (416) 652-7073 Website: www.travelwave.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP and VPN
Lists Pleasants wrote: ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have listed below. set policcy id 1001 from Trust to Trust Local Remote SIP permit log count set policy id 1001 application IGNORE set policy id 1002 from Trust to Trust Remote Local SIP permit log count set policy id 1002 application IGNORE I am running 5.2r1 without any issues but I have turned off any application deep scanning. unset alg sql unset alg q931 unset alg h245 unset alg ras unset alg sip -Chip I tried adding the above and it made no difference. My unset alg lines look a little different. They end in enable, but that could be the software version. I'm still getting stumped as to how it can register correctly and not have audio on outbound calls. I double checked and if I call from the phone system to the home phone, audio is fine! Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Wits end with echo
1.2-beta2 is more efficient against echo issues with ECHO_CAN_MG2 :-) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jon Reynolds Envoyé : jeudi 10 novembre 2005 08:58 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Wits end with echo Richard Scobie wrote: Jon Reynolds wrote: I have updated the phones to 1.0.12 firmware, I have echotraining=800, echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using Mark2 as the echo suppresion and still I have echo. Is this correct? I do not believe having these echo parameters in sip.conf will achieve anything. They should be at the top of zapata.conf. Regards, Richard That is incorrect, I wasn't thinking clearly, it is zapata.conf that these settings are in. Thanks for the correction Richard, Jon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TDM400 Card
Shaun Singh wrote: Is there some kind of limit to the number of TDM04B cards you can use in your Asterisk system (Red Hat 9, kernel 2.4, Asterisk CVS-v1-0-11-11/16/04-13:41:01))? I have 2 cards right now(rev B) with 8 analog lines connected to 8 FXO modules. I wanted to add 2 more analog lines but the third card (rev I) refuses to recognize the two new FXO modules. Digium have said their newer version TDM cards are backward-compatible. There is no problem with the PCI slot or IRQ. I'm using the motherboard (Asus P4P800-E) as recommended by Digium. Any ideas? Digium's TDM2400P is better suited to your configuration. Maybe ask Digium if they have some kind of trade in program? Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] looking for keypad free sip phones
I am looking for sip phones which do not have keypads but only a ringer/light for use in factories, outdoors, etc. -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Partner Sr. Manager 7 West 24th Street #100 - - +1 (443) 269-1555 Baltimore, Maryland 21218 - - - -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, purge the message from your system and notify the sender immediately. Any other use of the email by you is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP and VPN
Lists Pleasants wrote: ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have listed below. set policcy id 1001 from Trust to Trust Local Remote SIP permit log count set policy id 1001 application IGNORE set policy id 1002 from Trust to Trust Remote Local SIP permit log count set policy id 1002 application IGNORE I am running 5.2r1 without any issues but I have turned off any application deep scanning. unset alg sql unset alg q931 unset alg h245 unset alg ras unset alg sip -Chip Why do you go from Trust to Trust in your policies? I tried that and the phone won't work at all. The only way to get it to register is for me to put Remote as an Untrust zone. Thanks! Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sipphone for freebsd
On Thu, Nov 10, 2005 at 12:57:45PM +0800, Dinesh Nair wrote: On 11/10/05 08:52 Pablo Allietti said the following: yes but both of them have problem with voice. some skype too anybody can have this problems in freebsd? i hear cutted conversations`: perhaps there's contention for your sound/mic devices. what does the hw.snd.pcm0.vchans say, also what's the output of cat /dev/sndstat ? yesss i solve the problem with that. and you know in linux how to setup for 1 channel only? with multiple virtual sound channels, you can have different apps sharing the sound devices cleanly. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug in 1.2rc1
Guys.
I just discovered a bug in rc1, whenever We try to do an addqueuemember,
asterisk core dumps.
Here is the dialplan:
exten = 766,1,AddQueueMember(Ventas)
exten = 766,2,AddQueueMember(Soporte-Tecnico)
exten = 766,3,AddQueueMember(Soporte-Contrato)
exten = 766,4,UserEvent(Agentlogin|Agent: ${CALLERIDNUM})
exten = 766,5,Playback(agent-loginok)
exten = 766,6,Playback(vm-goodbye)
[Nov 10 11:15:41] -- Executing AddQueueMember(SIP/201-5a35, Ventas)
in new stack
voip*CLI
Disconnected from Asterisk server
[Nov 10 11:15:41] Executing last minute cleanups
[Nov 10 11:15:41] Asterisk cleanly ending (0).
Any more info I can provide to help debug this?
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[Asterisk-Users] voicemail to two emails?
Can this be done? I have a customer service que that if full go to v-mail. I would like to know how I can put two e-mail address for it to go to. Is that possible? Thanks! -J ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bug in 1.2rc1
Anton -
Thanks for the report. I've just posted a bug for you on the bug tracker at
http://bugs.digium.com/view.php?id=5705
Please refer to that URL for further information/resolution.
On 11/10/05, Anton Krall [EMAIL PROTECTED] wrote:
Guys.
I just discovered a bug in rc1, whenever We try to do an addqueuemember,
asterisk core dumps.
Here is the dialplan:
exten = 766,1,AddQueueMember(Ventas)
exten = 766,2,AddQueueMember(Soporte-Tecnico)
exten = 766,3,AddQueueMember(Soporte-Contrato)
exten = 766,4,UserEvent(Agentlogin|Agent: ${CALLERIDNUM})
exten = 766,5,Playback(agent-loginok)
exten = 766,6,Playback(vm-goodbye)
[Nov 10 11:15:41] -- Executing AddQueueMember(SIP/201-5a35, Ventas)
in new stack
voip*CLI
Disconnected from Asterisk server
[Nov 10 11:15:41] Executing last minute cleanups
[Nov 10 11:15:41] Asterisk cleanly ending (0).
Any more info I can provide to help debug this?
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http://www.btwtech.com/
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[Asterisk-Users] (Some problems sending this menssage) Sound quality of the new BT 101 and 102 models
De: André Rodrigues ( Cheyenne) [mailto:[EMAIL PROTECTED] Enviada: quinta-feira, 10 de Novembro de 2005 16:18Para: 'asterisk-users@lists.digium.com'Assunto: Sound quality of the new BT 101 and 102 models Hi. Im having sound quality problems using the new BT 101 and 102 models (the ones with solid colour bottoms like the gxp model). Im using firmware 1.0.6.7. Does anyone as the same problem with these new models? Sound quality has no cuts or noise. But the sound is much more lower and not clear and crystalline. Im using PCMA. Regards. André M. S. Rodrigues ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ring silent
I have a request to have an extension to ring silently or different When a call comes into a queue. This extension is a manager that is monitoring the queue that the customer server is taking calls in. Is this Possible? -J ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I factory reset a Grandstream BT-102
Hi, Just pulled out the BT-102 because I need to use it again, entered in the TFTP server to get the latest firmware so its now in 1.0.6.7 and i now was to factory default the phone and set it up from scratch.. I tried the instructions (copied below this message) from the latest available version of the user guide on the Grandstream site but it didn't appear to work.. Anyone got any idea how to factory reset these phones? Thanks 8 Restore Factory Default Setting Warning: Restore the Factory Default Setting will delete all configuration information of the device. Step one: Find the Mac Address of the device. The Mac address of the device is located on the bottom of the device. It is a 12 digit number. Step two: Encode the Mac address. The encode rule is: 2 is the first letter on the button 2 so its encoding is 2 . A is the second letter on button 2 so its encoding is 22 . B is the third letter on button 2 and its encoding is 222 . C is the fourth letter on button 2 and its encoding is . 3 is the first letter on the button 2 so its encoding is 3 . D is the second letter on button 2 so its encoding is 33 . E is the third letter on button 2 and its encoding is 333 . F is the fourth letter on button 2 and its encoding is . For example, the Mac address is 000b8200e395, User should encode it as 000222820095 . Step three: Access the phone screen menu, then select the -- reset -- with the up or down arrows keys. Step four: Dial in the encode of the Mac address. Once the correct encode Mac address dial in, the device will reboot automatically and restore the factory default setting. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 1.2-rc1 and sip show inuse
I apologize if this question has been asked before. Did something change with the behaviour of the 'sip show inuse' command between 1.0.9 and 1.2-rc1? I used to be able to see a list of extensions and the number of in/out calls. Now it just reports: asterisk*CLI sip show inuse * User name In use Limit * Peer name In use Limit no matter how many calls are being used. asterisk*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.70.128 1234339ad96826e 00102/0 ulaw No Tx: ACK 192.168.70.116 1235723e1612-52 00101/2 ulaw No Rx: ACK 2 active SIP channels Any info about getting the previous functionality back would be greatly appreciated. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] voicemail to two emails?
If you are using Sendmail you can alias a single email address to multiple email addresses: http://www.uwsg.iu.edu/usail/mail/aliasing/ If you are using Exchange you can create a distribution list with a single email address that expands to multiple recipients: http://imanami.com/support/viewer.aspx?ID=10013 In both cases, you would enter the alias email address or the distribution list email address into voicemail.conf hth -Original Message- From: Jason Brashear [mailto:[EMAIL PROTECTED] Sent: Thursday, November 10, 2005 10:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] voicemail to two emails? Can this be done? I have a customer service que that if full go to v-mail. I would like to know how I can put two e-mail address for it to go to. Is that possible? Thanks! -J ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] voicemail to two emails?
At 11:27 11/10/2005, Jason Brashear, wrote: Can this be done? I have a customer service que that if full go to v-mail. I would like to know how I can put two e-mail address for it to go to. Is that possible? You can type in the emails and see if it works. I think I tried, but didn't have success. Set it up in the email client (Eudora) to forward. Also, possible to set it up in email server. Needed feature, for sure. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP and VPN
The example I gave was going over a VPN with tunnel terminating in the trusted zone. Put the polices how our traffic traverse through the netscreen. I would config a policy for trust to untrust traffic and for untrust to trust or untrust to global if you have MIPing going on. -chip -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Johnson Sent: Thursday, November 10, 2005 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP and VPN Lists Pleasants wrote: ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have listed below. set policcy id 1001 from Trust to Trust Local Remote SIP permit log count set policy id 1001 application IGNORE set policy id 1002 from Trust to Trust Remote Local SIP permit log count set policy id 1002 application IGNORE I am running 5.2r1 without any issues but I have turned off any application deep scanning. unset alg sql unset alg q931 unset alg h245 unset alg ras unset alg sip -Chip Why do you go from Trust to Trust in your policies? I tried that and the phone won't work at all. The only way to get it to register is for me to put Remote as an Untrust zone. Thanks! Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] receive fax with asterisk
Receiving faxes with Asterisk. Is there a good resource for learning how to set this up? -J ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sched.c: Attempted to delete nonexistent schedule entry
Is anyone else having all IAX peers die right after receiving this in the log? I have CVSHEAD from about 2 weeks ago. Packet capture shows Asterisk stops transmitting all IAX packets after this messages appears. - Dustin - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bug in 1.2rc1
Thx BJ, Ill monitor the bug there in case more info is needed.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|BJ Weschke
|Sent: Thursday, November 10, 2005 11:30 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Bug in 1.2rc1
|
| Anton -
|
| Thanks for the report. I've just posted a bug for you on the
|bug tracker at
|
| http://bugs.digium.com/view.php?id=5705
|
| Please refer to that URL for further information/resolution.
|
|On 11/10/05, Anton Krall [EMAIL PROTECTED] wrote:
| Guys.
| I just discovered a bug in rc1, whenever We try to do an
| addqueuemember, asterisk core dumps.
|
| Here is the dialplan:
|
| exten = 766,1,AddQueueMember(Ventas)
| exten = 766,2,AddQueueMember(Soporte-Tecnico)
| exten = 766,3,AddQueueMember(Soporte-Contrato)
| exten = 766,4,UserEvent(Agentlogin|Agent: ${CALLERIDNUM}) exten =
| 766,5,Playback(agent-loginok) exten = 766,6,Playback(vm-goodbye)
|
| [Nov 10 11:15:41] -- Executing
|AddQueueMember(SIP/201-5a35, Ventas)
| in new stack
| voip*CLI
| Disconnected from Asterisk server
| [Nov 10 11:15:41] Executing last minute cleanups [Nov 10 11:15:41]
| Asterisk cleanly ending (0).
|
| Any more info I can provide to help debug this?
|
| ___
| --Bandwidth and Colocation sponsored by Easynews.com --
|
| Asterisk-Users mailing list
| Asterisk-Users@lists.digium.com
| http://lists.digium.com/mailman/listinfo/asterisk-users
| To UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
|
|
|
|--
|Bird's The Word Technologies, Inc.
|http://www.btwtech.com/
|___
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|
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|Asterisk-Users@lists.digium.com
|http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Ex-girlfriend mode on invalid/no CID?
Does anyone know how to ignore/send straight to voicemail all calls with invalid or no CID? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simple Dial for If Busy Send to Voicemail
I still cant get it to work. My
traffic will be coming from the PSTN (Zap/1) into one context and will Dial a
SIP extension in another context. I have tried making the changes to both
without luck. In the example exten=s,3,Dial(${theChannel}/12345678)
confuses me. Why am I dialing the Zap channel and do I need to change the
12345678 to SIP/myextension? I actually tried replacing it but it also did not
work. Any suggestions?
-chip
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Thursday, November 10, 2005
11:00 AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Simple
Dial for If Busy Send to Voicemail
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
Is pretty simple. Replace the 102 priority with a call
to voicemail and youre set. hth
-Original
Message-
From: Pleasants Email Lists
[mailto:[EMAIL PROTECTED]
Sent: Thursday, November 10, 2005
8:22 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Simple
Dial for If Busy Send to Voicemail
I
am looking for a simple dial plan for if my zap channel is busy/unavailable
send to Voicemail. I couldnt find anything simple online.
-chip
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[Asterisk-Users] Need help can't figure out what wrong with zapata.conf
Hi, I get the following when I reload: -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found Nov 10 10:57:34 WARNING[4475]: chan_zap.c:10816 setup_zap: Ignoring signalling Nov 10 10:57:34 ERROR[4475]: chan_zap.c:10249 setup_zap: Unable to reconfigure channel '1' Nov 10 10:57:34 WARNING[4475]: chan_zap.c:11009 reload: Reload of chan_zap.so is unsuccessful! My zapata.conf look like the following: ;File Name - zapata.conf ;Revision Date 11-9-05 ;Version 1.3.0 [channels] language=en usecallerid=yes callerid=asreceived hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no faxdetect=both context=incoming-rest ;Incoming calls to to incoming-rest in extensions.conf signalling=fxs_ks ;Use FXS signalling for an FXO channel group=1 ;Group association for outbound trunk channel=1 ;PSTN attached to port 1 - Resturaunt context=incoming-home ;Incoming calls to incoming-home in extensions.conf signalling=fxs_ks ;Use FXS signalling for an FXO channel group=1 ;Group association for outbound trunk channel=2-3;PSTN attached to port 2,3 - Homecare context=trunkdial ;Uses the trunkdial context in extensions.conf signalling=fxo_ks ;Use FXO signalling for an FXS channel channel=5-8;Telephone attached to ports 5,6,7,8 I must be missing something any ideas. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot find where error message is comming from...
Hi, I am getting the following from the Asterisk console: Nov 11 10:33:50 NOTICE[3578]: pbx.c:1747 pbx_extension_helper: Cannot find extension context 'default' Nov 11 10:34:10 NOTICE[3578]: pbx.c:1747 pbx_extension_helper: Cannot find extension context 'default' I installed the Asterisk sample files for 1.2rc1 and then replaced the following files: agents.conf, queues.conf, sip.conf, zaptel.conf, meetme.conf, voicemail.conf, and extensions.conf with my own configuration. I have no 'default' context in these files. I guess the obvious question is a default context required? Also do the sample files contain a reference to a default context? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TDM400 Card
Is anyone using these high-density TDM2400P cards? I'm cautious about using anything that's brand new. Regards, Shaun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jason Becker Sent: November 10, 2005 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM400 Card Shaun Singh wrote: Is there some kind of limit to the number of TDM04B cards you can use in your Asterisk system (Red Hat 9, kernel 2.4, Asterisk CVS-v1-0-11-11/16/04-13:41:01))? I have 2 cards right now(rev B) with 8 analog lines connected to 8 FXO modules. I wanted to add 2 more analog lines but the third card (rev I) refuses to recognize the two new FXO modules. Digium have said their newer version TDM cards are backward-compatible. There is no problem with the PCI slot or IRQ. I'm using the motherboard (Asus P4P800-E) as recommended by Digium. Any ideas? Digium's TDM2400P is better suited to your configuration. Maybe ask Digium if they have some kind of trade in program? Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Speex codec problems
On Mon, Nov 07, 2005 at 01:23:26PM -0500, Branko Samardzic wrote: I am trying to tweak my Asterisk servers to talk to each other using Speex codec. I downloaded and installed speex and speex devel libraries, recompiled asterisk (including make clean), did set up speex codec as only one allowed on both sides. Sounds enough. However, conversations are not Speex encoded!!! It is codec 64 (16 bit Signed Linear PCM) all the time. Please provide more information. a trace from sip debug would help. Alternatively (if either side is asterisk): sip/iax config of that side. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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_modem_aopen.so loaded despite being told not too!
Hi, I am using 1.2rc1 and my modules.conf looks like this: [EMAIL PROTECTED] ~]# vi /etc/asterisk/modules.conf [modules] autoload=yes ; ; Any modules that need to be loaded before the Asterisk core has been ; initialized (just after the logger has been initialized) can be loaded ; using 'preload'. This will frequently be needed if you wish to map all ; module configuration files into Realtime storage, since the Realtime ; driver will need to be loaded before the modules using those configuration ; files are initialized. ; ; An example of loading ODBC support would be: ;preload = res_odbc.so ;preload = res_config_odbc.so ; ; If you want, load the GTK console right away. ; Don't load the KDE console since ; it's not as sophisticated right now. ; noload = pbx_gtkconsole.so ;load = pbx_gtkconsole.so noload = pbx_kdeconsole.so ; ; Intercom application is obsoleted by ; chan_oss. Don't load it. ; noload = app_intercom.so ; ; Explicitly load the chan_modem.so early on to be sure ; it loads before any of the chan_modem_* 's afte rit ; noload = chan_modem.so noload = chan_modem_aopen.so load = res_musiconhold.so ; ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; noload = chan_alsa.so ;noload = chan_oss.so ; ; Module names listed in global section will have symbols globally ; exported to modules loaded after them. ; [global] I get the following during a reload: == Parsing '/etc/asterisk/modem.conf': Found Nov 10 11:07:31 WARNING[4568]: loader.c:305 __load_resource: Module 'chan_modem_aopen.so' already exists Nov 10 11:07:31 ERROR[4568]: chan_modem.c:1023 load_module: Failed to load driver chan_modem_aopen.so == Loading modem driver chan_modem_aopen.so-- Reloading module 'pbx_config.so' (Text Extension Configuration) Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Transfer Problem with IAX2
I'm using IAX2 with VP-320I hardphones for remote users. Everything seems to be working fine except for call transfer. Is this an issue with the IAX2 itself or the phone? If I flash the same phone with SIP, the problem disappears. Regards, Shaun Singh, Manager Travelwave 1655 Dufferin Street, Suite 201 Toronto, ON M6H 3L9 Tel: (416) 652-1212 Ext 101 Fax: (416) 652-7073 Website: www.travelwave.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_merge_contexts_and_delete: Requested contexts didn't get merged???
Hi, I get the following during a reload with Asterisk 1.2rc1 Nov 10 11:07:31 WARNING[4568]: pbx.c:3757 ast_merge_contexts_and_delete: Requested contexts didn't get merged -- Reloading module 'codec_gsm.so' (GSM/PCM16 (signed linear) Codec Translator) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_gsm: using generic PLC -- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)... What is this and where do I look to fix it??? What is the ast_merge_context_and_delete file Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] MAX TNT SIP / Asterisk
Hi,
Someone have running a MTNT,SIP and Asterisk please let me know really I
don't know which way to take.
Greetings,
JC.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Cesar Pinto
Sent: Wednesday, November 09, 2005 3:56 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] MAX TNT SIP / Asterisk
Hi,
I'm implementing MAX TNT in SIP mode with Asterisk, and I couldn't
establish the connection.
So, reviewing the messages posted in this list I found a message with
date Nov 10 2004, a year ago :)
Well, I have the same problem posted by James Taylor; my configuration
is the same that Darren Bentley propose.
I'd like to know if some have more information about that.
Thanks in advance,
JC.
---
PDTA: I page the history.
Using Software version 10.1.0
Here's what I did:
1. Create a Media Profile (called voip)
name* = voip
active = yes
protocol-type = sip
[in MEDIA-GATEWAY/voip:voip-options]
packet-audio-mode = g711-ulaw
frames-per-packet = 2
silence-det-cng = no
ena-adap-jitter-buffer = yes
max-jitter-buffer-size = 19
initial-jitter-buffer-size = 2
voice-ann-dir = /current
voice-ann-enc = g711-ulaw
call-inter-digit-timeout = 6000
silence-threshold = 0
dtmf-tone-passing = inband
maxcalls = 672
rfc2833-payload-type = 96
g711-transparent-data = no
rtp-problem-reporting = { no 30 60 }
[in MEDIA-GATEWAY/voip:sip-options]
t1-timer = 500
t2-timer = 4000
invite-retries = 6
non-invite-retries = 10
primary-proxy = { x.x.x.x 5060 compact } (IP ADDRESS OF ASTERISK)
secondary-proxy = { 0.0.0.0 5060 compact }
registration-proxy = { x.x.x.x 5060 compact 1 } (IP ADDRESS OF
ASTERISK)
proxy-heartbeat = 0
proxy-failover-window = 60
reroute-on-proxy-failure = no
trusted-proxy =
unknown-ani =
blocked-ani =
privacy-proxy-require = disabled
cause-code-map = s
start-call-method = invite
trunk-group-options =
onhold-minutes = 0
support-100rel = disabled
internationalize = no
international-prefix = no
country-code =
national-destination-code =
local-number-ton = unknown-ton
call-transfer-method = ip-transfer
notify-timer = 0
invite-with-multiple-codecs = disabled
2. Configure Call Route for Digitam Modem card
admin get call-route {{{1 3 0}0}0}
[in CALL-ROUTE/{ { { shelf-1 slot-3 0 } 0 } 0 }]
index* = { { { shelf-1 slot-3 0 } 0 } 0 }
active = yes
trunk-group = 0
phone-number = 7299 (last 4 digits of your DID)
preferred-source = { { any-shelf any-slot 0 } 0 }
call-route-type = voice-call-type
cost = 0
3. Configure the T1 ports
default-call-type = dnis-or-voip
media-gateway = voip
I did this about 8 months ago and don't have my notes with me so I hope
I remembered everything. Give it a shot. Good luck
- Darren
On Tue, 2004-11-09 at 09:49, Tim Connolly wrote:
Do you have the TNT's config available? I'd love to see this work!
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Darren Bentley
Sent: Monday, November 08, 2004 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MAX TNT SIP / Asterisk
Have you attempted to use SIP? It's working quite well for me.
sip.conf
[maxtnt]
type=friend
host=xxx.xxx.xxx.xxx
dtmfmode=inband
callerid=MaxTNT maxtnt
context=toll-access
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=g729
allow=ulaw
extensions.conf
(xxx.xxx.xxx.xxx would be the address of your MaxTNT)
[toll-trunks]
;
; Outbound 1-nxx-nxx- goes via: PSTN
;
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],60)
exten = _1NXXNXX,2,Hangup
[local-trunks]
;
; Outbound to nxx- goes via: PSTN
;
exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED],60)
exten = _NXX,2,Hangup
;
[local-access]
;
; Extensions that are this context are allowed to only call local PSTN
numbers and other extensions
;
include = extensions
include = local-trunks ; Access to Local numbers
[toll-access]
;
; Extensions that are this context are allowed to call local and long
distance PSTN numbers and other extensions
;
include = local-access ; Everything local-access has
include = toll-trunks ; Access to toll numbers
- Darren
On Mon, 2004-11-08 at 10:36, James Taylor wrote:
Your question indicates that there may be a better way...
???
I want to use the voice mail and extension features of Asterisk, and
sometimes there is this NAT problem that Asterisk seems to handle
very
well.
I've been using H.323 with the TNT.
Do you have an alternate solution?
On Mon, 8 Nov 2004 10:41:31 -0500 (EST), alex at pilosoft.com
wrote:
On Tue, 2 Nov 2004, James Taylor wrote:
I can't get my MAX TNT to register with Asterisk.
TAOS 11.0.
SIP phone registeration show up in Asterisk like this:
sip:user_name at ip_address and works.
The TNT shows up as:
sip:@ip_address.
Does anyone have
Re: [Asterisk-Users] looking for keypad free sip phones
You are probably not going to find a ip phone that does that. I recommend taking a look at http://www.vikingelectronics.com, they have a number of emergency/hot phone type devices. Then you would simply plug it into a Sipura SPA FXS configured to dial a number when it senses off hook. We are looking to put one of these at one of our vehicle gates. The driver just lifts the handset and the combo dials the receptionist who can remotely unlock the gate.On 11/10/05, Jason Pyeron [EMAIL PROTECTED] wrote: I am looking for sip phones which do not have keypads but only aringer/light for use in factories, outdoors, etc. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bug in 1.2rc1
On 11/10/05, Anton Krall [EMAIL PROTECTED] wrote: Thx BJ, Ill monitor the bug there in case more info is needed. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |BJ Weschke |Sent: Thursday, November 10, 2005 11:30 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Bug in 1.2rc1 | | Anton - | | Thanks for the report. I've just posted a bug for you on the |bug tracker at | | http://bugs.digium.com/view.php?id=5705 | | Please refer to that URL for further information/resolution. | There's a patch up now. It was a legitimate bug with memory handling brought out when you didn't specify an interface. Thanks for testing it before the release! -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] How do I factory reset a Grandstream BT-102
On Thu, 2005-11-10 at 17:36 +, WipeOut wrote: For example, the Mac address is 000b8200e395, User should encode it as 000222820095 . Step three: Access the phone screen menu, then select the -- reset -- with the up or down arrows keys. Step four: Dial in the encode of the Mac address. Once the correct encode Mac address dial in, the device will reboot automatically and restore the factory default setting. The part about encoding the MAC is a little unclear. Just remember that you should see the MAC on screen as it is written on the back of the phone, including letters. I have used this procedure many times and it works as advertised. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Little OT.. SER Question
Anyone with SER knowledge could you point me in a direction to setup SER to rewrite the SIP URI? Currently I have the following [EMAIL PROTECTED] I am setting it so it does the change but its still showing up with the prefix. I need it to look like this: [EMAIL PROTECTED] I got xxx.xxx.xxx.xxx to change to yyy.yyy.yyy.yyy I just need the prefix to go away now.. ☺ ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
