[asterisk-users] Need help connecting Alcatel 4400 PBX to Asterisk
Title: Need help connecting Alcatel 4400 PBX to Asterisk Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp ([EMAIL PROTECTED]) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps on digium website...no errors reported. The modules seem to have loaded...here's what lsmod shows: Module Size Used by wcte11xp 30496 31 zaptel 196740 67 wcte11xp still the light on my card is offdoes that mean the card has not initialised properly? On loading Asterisk, I do not get any errors, but I do see these warnings: Parsing '/etc/asterisk/zapata.conf': Found Nov 1 11:57:21 WARNING[3454]: chan_zap.c:10874 setup_zap: Ignoring switchtype Nov 1 11:57:21 WARNING[3454]: chan_zap.c:10874 setup_zap: Ignoring signalling on running asterisk -cvvv . I do see Aterisk Ready at the end ...then What do these warnings mean? Also, I DO NOT get these lines on asterisk startup:- channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 does that mean my channels are not available? *CLI zap show status Description Alarms IRQ bpviol CRC4 Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 *CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 --- here's my extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes [sip] exten = 9820,1,Dial(SIP/iyer) exten = 9821,1,Dial(SIP/shweta) exten = 9810,1,Dial(SIP/shashi) exten = 9851,1,Dial(Zap/g1/851,20) [incoming] exten = s,1,Answer() exten = s,2,Playback(hello-world) exten = s,3,Hangup() exten = 9821,1,Dial(SIP/shashi) exten = 9851,n,Dial(Zap/g1/851) --- here's zapata.conf [trunkgroups] trunkgroup = 1,16 spanmap =1,1,1 [channels] switchtype=euroisdn signalling=pri_cpe context=incoming language=uk group=1 callgroup=1 pickupgroup=1 echocancel=yes immediate=no channel = 1-15,17-31 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancelwhenbridged=yes musiconhold=default --- here's zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us --- Now the problem I can call and talk SIP to SIP...here's what I see on asterisk CLI -- Executing Dial(SIP/iyer-09326480, SIP/shweta) in new stack -- Called shweta -- SIP/shweta-0932b9c0 is ringing But when I call zap extension, here's what I get: Executing Dial(SIP/iyer-09326480, Zap/g1/851|20) in new stack Nov 1 12:07:55 NOTICE[3513]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/iyer-09326480' status is 'CONGESTION' I have connected the PBX to digium card with the specified cable and done the settings in PBX specified at: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI What am I doing wrong? I'd like to mention that on the Alcatel PX rack on the PRA2 card, the NO-SIGNAL (NOS) light comes on when I shut down my linux box but it's off when I load zapteldoesn't that mean that PBX is able to sync to my asterisk server? Any help would be greatly appreciated. Thanks in advance Kind Regards Shweta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards vs. Channel bank with T1
Dovid B wrote: Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? Thanks. channel bank is more friendly to faxes and modems (v90 can work too) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Example Polycom function key config
Hi Jamie - Hi Noah, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example Here's the keys line that I use for one of my clients: keys key.scrolling.timeout=1 key.IP_500.37.function.prim=DialpadPound key.IP_500.31.function.prim=DialpadStar key.IP_600.37.function.prim=DialpadPound key.IP_600.30.function.prim=DialpadStar/ Thanks for that, I have something similar but what I can't work out is how to send multiple digits. For example 2x 'DialpadPound'. I have tried putting it in twice etc. to no avail. Anyone know how to get this to work? I'm trying to get our transfer key (##) programmed to one of the function keys basically. Thanks, Jamie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP realtime issues
Does anyone see anything wrong here? CLI realtime load sipusers name 1000 Column Name Column Value id 1 name 1000 callerid "Don" 1000 host dynamic nat yes disallow all allow gsm type friend context inbound secret 41674 extconfig.conf sipusers = mysql,my_sip_table,sipbuddies Now if the same info is just put right into sip.conf xlite will register fine... I tried 1.2.13 asterisk...now I am using 1.2.12.1 because I thought maybe it was broken in 1.2.13 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager API - Originate Call - Need Help
Hi all,How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number?I can originate a call from my SIP-network using this parameters in Originate call command :Channel = SIP/0041435215301Context = defaultExten = 00982166501553Priority = 1CallerID = 0041435215301this works with out any problems I initiate a call from one of my network sip clients (0041435215301) and call someone at anyside of the world, but Can I initiate a call from (00982166501553) to one of my sip users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API - Originate Call - Need Help
On Wed, 2006-11-01 at 03:23 -0800, Ehsan Khosrowshahi wrote: Hi all, How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command : Channel = SIP/0041435215301 Context = default Exten = 00982166501553 Priority = 1 CallerID = 0041435215301 this works with out any problems I initiate a call from one of my network sip clients (0041435215301) and call someone at anyside of the world, but Can I initiate a call from (00982166501553) to one of my sip users? Why not do this: Channel = ZAP/g1/00982166501553 Context = default Exten = whateveryoursipphoneis priority =1 CallerID = whateveryouwant If you don't have an extension for your sip phone, add this in context default: exten = whateveryoursipphoneis,1,SIP/SIP/0041435215301 Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API - Originate Call - Need Help
If I understood your question correctly, you just need to reverse everything. Channel = OUTGOING TRUNK i.e. ZAP/00982166501553 Context = default Exten = internal extension that points to - 0041435215301 Priority = 1 CallerID = 0041435215301 This will first initiate the call to the number 0041435215301 and then connect it to the internal extension you specify in Exten that points to SIP/0041435215301. Cheers On 11/1/06, Ehsan Khosrowshahi [EMAIL PROTECTED] wrote: Hi all, How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command : Channel = SIP/0041435215301 Context = default Exten = 00982166501553 Priority = 1 CallerID = 0041435215301 this works with out any problems I initiate a call from one of my network sip clients (0041435215301) and call someone at anyside of the world, but Can I initiate a call from (00982166501553) to one of my sip users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help me on Call parking
Hello Users...I'm Strucked in Call parking...I'm Using the Asterisk-1.1.11 version in My FC5 box,In That there is feature.confI'm Using SIP channel By using Asterisk + OpenSER [general]parkext = 9006 ; What extension to dial to parkparkpos = 9007-9009 ; What extensions to park calls on. These needs to be ; numeric, as Asterisk starts from the start position ; and increments with one for the next parked call.context = parkedcalls ; Which context parked calls are inparkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds)IIn Extension.conf .. I'm confused to give the Dial planning..Can Help -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ${CALLERIDNUM}
Hi
Does anyone know how I can check if a callerID is more than
2 digits.
I am setting up my phones so that if the callerID is 3
digits the phones ring one way if it is more than 3 digits it rings another
i.e. internal calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM} =
]?5)
This will tell it to jump to 5 if callerID if but how
do i tell it do jump based on length of callerID?
Many thanks
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[asterisk-users] Re: Manager API - Originate Call - Need Help
In article [EMAIL PROTECTED], Ehsan Khosrowshahi [EMAIL PROTECTED] wrote: How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command : Channel = SIP/0041435215301 Context = default Exten = 00982166501553 Priority = 1 CallerID = 0041435215301 this works with out any problems I initiate a call from one of my network sip clients (0041435215301) and call someone at anyside of the world, but Can I initiate a call from (00982166501553) to one of my sip users? Try using the Local channel type: Channel = Local/[EMAIL PROTECTED] Context = sip-extensions Exten = 0041435215301 Priority = 1 CallerID = 00982166501553 (use whatever context contains your SIP extensions) This will place the call to your PSTN number first, and when that is answered, it will call the SIP extension and connect the call to it. You ought to think about how the person answering the PSTN extension will know what is happening. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Realtime MD5 authentication
Hi, Is there any possibility to have md5 encoded passwords in the IAX users database? I notice the secret AND/OR md5secret columns always have to contain the password in plain text even when you set the auth column value to md5?!? Am I missing out something? Any ideas on how to correct this? Having plain text passwords in the realtime database is not very suitable for me and poses a security vulnerability. Thanks, Pedros ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${CALLERIDNUM}
The following will:
exten = s,1,GotoIf($[${LEN(${CALLERID(num)})}=2]?50)
On 11/1/06, Scott Pinhorne [EMAIL PROTECTED] wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the phones
ring one way if it is more than 3 digits it rings another i.e. internal
calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
This will tell it to jump to 5 if callerID if but how do i tell it do
jump based on length of callerID?
Many thanks
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Re: [asterisk-users] Help me on Call parking
raviprakash sunkara wrote: In Extension.conf .. I'm confused to give the Dial planning.. You don't need to do anything in the dial plan for parking. Just transfer the call to your parking extension and Asterisk will take it from there. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Newbie Questions
Ken If these are older comdials then they are just analog phones with extra signaling. The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do. Always remember that it would cost the companies a ton to make every system totally closed That being said, the entry price for IP phones or ADSI phones can be much lower than you think. Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection. That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option was available. Thanks for any help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${CALLERIDNUM}
On 11:53, Wed 01 Nov 06, Scott Pinhorne wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the phones
ring one way if it is more than 3 digits it rings another i.e. internal
calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
This will tell it to jump to 5 if callerID if but how do i tell it do
jump based on length of callerID?
check the LEN() dialplan function
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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Re: [asterisk-users] ${CALLERIDNUM}
On Wed, 2006-11-01 at 11:53 +, Scott Pinhorne wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the
phones ring one way if it is more than 3 digits it rings another i.e.
internal calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
I'm sure on the wiki (http://voip-info.org) is a list of functions, including
one to
determine length of strings, but you could also do something like:
exten = ,1,Goto(${CALLERIDNUM},1)
exten = _XXX,1,dostuffwith3digits
exten = _.,1,dostuffwithmorethan3digits
Conrad
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[asterisk-users] AEL2 - CUT function usage
Hi,
In Asterisk 1.2.7, my AEL code looks like this:
macro callForwardHunt(numargs,numlist,typelist,ttr)
{
for(x=1;${x}${numargs}+1;x=${x}+1)
{
CUT(number=numlist,-,${x});
CUT(type=typelist,-,${x});
NoOp(${number});
NoOp(${type});
Dial(${type}${number},${ttr});
};
};
In Asterisk 1.4.0beta3, the CUT function looks like this:
NoOp(${range});
Set(time_range=${CUT(range|/|1)});
NoOp(${time_range});
No I understand that the CUT application has been removed in 1.4, so now I am usung the CUT
function, but where is it explained that you have to have to use SET and the commas ',' has to be
replaced with '|'.
Or have I done something stupidly wrong :)
--
thanks,
yusuf
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Re: [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - cardrecommendation
Hi Wendy, I got this info from digium developers, that caller id name transfer/display (asterisk/iphone - pbx/clasic phone)) using ISDN/Q.SIG should work, so, do you have possibility to confirm this, if it realy working in practice (with siemens hipath idealy)? thanks PJ Original Message Subject:Re: [asterisk-dev] Zaptel/Asterisk - Q.SIG status Date: Tue, 31 Oct 2006 09:54:05 -0600 From: Matthew Fredrickson [EMAIL PROTECTED] Reply-To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com References: [EMAIL PROTECTED] On Oct 31, 2006, at 6:40 AM, Pavel Jezek wrote: Hello developers, because too litle info about what features are currently supported with Q.SIG, I would like to ask if caller id name supplementary service is currently available, i.e. if caller id name will be displayed on ip phone when calling from pbx to asterisk (through PRI and Digium card) and vice versa thank you Yeah, I haven't tested it in a while, but it should work. Just make sure you have in zapata.conf facilityenable=yes and switchtype=qsig. Matthew Fredrickson ___ [EMAIL PROTECTED] wrote: Hi, we have tested the Digium-Cards, they work fine, but don't expect to much! Only segmentation 1 in Ecma (it is not a digium-problem) The Name ist displayed, but only in Hex-Code (this is due to the Libpri/Zaptel Drivers but I didn't fint a way to display it in *) There is also very less documentation, on Asterisk.org (Features) there is non Q.Sig Support offered. Also very less documentation through google available. ;-( If you find some hints, i'm also interested! Regards wendy - Original Message - From: Pavel Jezek [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, October 19, 2006 5:22 PM Subject: [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - cardrecommendation Hello, if somebody using this scenario in production successfully, please send me info, which ISDN card for asterisk server is usefull for me (Digium, Sangoma)? my crucial requirement is caller id name transfer/display between ISDN (Siemens PBX) and IP phone connected to asterisk I'm using PRI interface and Q.SIG signaling. thank you PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${CALLERIDNUM}
Scott Pinhorne wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the phones
ring one way if it is more than 3 digits it rings another i.e. internal
calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
This will tell it to jump to 5 if callerID if but how do i tell it
do jump based on length of callerID?
Hi,
would this work:
exten = _X.,4,GotoIf($[${LEN(${CALLERIDNUM})} != 3 ] ? 40)
--
thanks,
yusuf
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Re: [asterisk-users] ${CALLERIDNUM}
That worked great
Many Thanks
-Original message-
From: C F [EMAIL PROTECTED]
Date: Wed, 1 Nov 2006 06:57:28 -0600
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ${CALLERIDNUM}
The following will:
exten = s,1,GotoIf($[${LEN(${CALLERID(num)})}=2]?50)
On 11/1/06, Scott Pinhorne [EMAIL PROTECTED] wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the phones
ring one way if it is more than 3 digits it rings another i.e. internal
calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
This will tell it to jump to 5 if callerID if but how do i tell it do
jump based on length of callerID?
Many thanks
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[asterisk-users] wav format isn't compatible with Windows Media Player
Hi, When playing a wav-format (( low compression),(wav49-format?)) file with Windows Media Player, it plays the file and then sometimes bombs out with an error about how the file is corrupt or unsupported. If you listen to the file in wavepad you will hear the whole file, in Media Player the last 2-3 sec. is missing (not played). The file is recorded with monitor function. The problem is reproduceable in Asterisk version 1.2.4 til 1.2.13. Error dialog: Windows Media Player cannot play the file. The file is either corrupt or the Player does not support the format you are trying to play. Clicking on More Information gives 0xC00D1199: Cannot play the file and a fair bit of troubleshooting stuff. _ Del dine store filer uden problemer med MSN Messenger: http://messenger.msn.dk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk and Panasonic KX Model
Thanks for help me with this issue. I've this scenario, a PANASONIC KX domain and an ASTERISK domain, each one with their own pool of extensions, incoming calls are recived by the PANASONIC KX as a gateway from PSTN to the office. Once a call is recived by the PANASONIC,it bridge the call to ASTERISK, asterisk then check if the extension called belongs to panasonic domain or to asterisk domain. If the called number belongs to panasonic, asterisk dial ( i don't know how to do) the internal number through the panasonic, else Asterisk dial a sip phone. Well, this is what i've to do, then my doubt falls in how to connect the devices(asterisk and panasonic kx), wich is the interface of the PANASONIC KX to send incoming calls to asterisk, and how to configure extensions.conf to match calls that send the PANASONIC KX System to the asterisk box. I´ve read that the interface may be the VM of the PANASONIC to send calls to asterisk, if it is right how i configure the systems to talk as i need?. Thanks. G. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a2billing
Dear How can I customize a2billing to have two groups One have service to play its balance and the second group do not play the balance. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom or Cisco Phones?
I love the Snom phones as well. The function keys are great and easy to use.On 10/31/06, mitcheloc [EMAIL PROTECTED] wrote:My vote is definitely for Snom, I've worked with Cisco phones foryears, but the Snom is much better integrated, and the feature buttons can be retooled for any environment, making custom installs very easy.On 10/31/06, Conrad Wood [EMAIL PROTECTED] wrote: On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote: Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia cars Not really. Both are very good phones. * My Clients prefer cisco because it looks more business-like. - The new snom phones do look better though and the side car rules. * The Cisco phone 'feels' very good in your hand, and the voicequality is superb. (I'd say slightly better than that of the snom 360) * Technically, I find the snom phone more advanced and I can do more cool stuff with it - Cisco doesn't seem to like giving features away in SIP. * Snom phones, for example, have freely programmable buttons that can park/retrieve/transfer calls, show line status etc. I can't get that to work with Cisco phones at all. * Putting custom ringtones (and choosing which ones to use) is a no-brainer with snoms and real trouble with ciscos. * On ciscos, I find the upgrade path from sccp to sip a totally unnecessary annoyance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Mitchel ConstantinSnap - A desktop user interface for Asteriskwww.snapanumber.com___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a2billing
Dear How can I customize a2billing to have two groups One have service to play its balance and the second group do not play the balance. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] Fxo box for asterisk ?
Hello, All the biggest gateways manufacturers do that. Search for Aliwei, Audiocodes, Patton, etc... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Noc Phibee Envoyé : lundi 30 octobre 2006 20:51 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Fxo box for asterisk ? Hi do you know if they have external Box (not internal card) for connect Analog Line and Pri/Isdn to asterisk for incomming and outgoing calls ... Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 faxing with spandsp and Grandstream HT.486
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Johann Steinwendtner schrieb: I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? I've successfully faxed via T.38 using these combinations: fax machine - HT486 - SIP server - HT486 - fax machine fax machine - HT486 - SIP server - Thomson ST620 - fax machine fax machine - Thomson ST620 - SIP server - HT486 - fax machine fax machine - AVM Fritzbox 7050 - SIP server - HT486 - fax machine fax machine - AVM Fritzbox 7050 - SIP server - Thomson ST620 - fax machine (note: T.38 is not officially supported on the AVM Fritzbox. It only works with older firmware versions and only when sending. When receiving a fax tone from the remote end, the Fritzbox does not issue a T.38 re-invite). I also just recently tested the following cases: fax machine - HT486 - OpenPBX with spandsp T.38 - RxFAX fax machine - Thomson ST620 - OpenPBX - RxFAX TxFAX - OpenPBX - HT486 - fax machine TxFAX - OpenPBX - Thomson ST620 - fax machine ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator inititate a re-invite ? When I tested T.38 with OpenPBX, it never detected the fax tone from the fax machine and thus did not issue a T.38 re-invite. This might be due to a configuration problem because I'm not overly familiar with OpenPBX and there are not many user reports available on the net. If you point me to the necessary patches for Asterisk, I will be glad to repeat my tests ;-) The fax tone detection and T.38 re-invite is always performed by the receiving terminator or gateway. I haven't read the corresponding RFC thoroughly enough but this is the way it works on all ATAs and gateways I've worked with so far. I also noticed that many modems which are connected to or built into PCs do not send a proper fax tone (CNG) so make sure you are using a good old fax machine. Thirdly, make sure that your fax machine does not support V.34 (class 2.1 aka Super G3) because T.38 is limited to 14400 bps on the modem side. I only tested one V.34 fax machine but its modem would not negotiate with the HT486 properly. txfax as originator: T.38 fax exchange takes place but the transmission is not successful, txfax reports errorcode 60 (Disconnected after permitted retry). I haven't been able to send more than a simple page using TxFAX with T.38 support yet. And even that did not work reliably. Either, the single page wasn't transmitted at all or fax machine on the HT486/ST620 recognized the end of the page but failed to negotiate the end of the transmission correctly. Once again, I only tried OpenPBX with T.38 termination support, yet, so your problem may be different. Can someone recommend a T.38 able ATA which is working with spandsp ? T.38 support in spandsp is still work-in-progress so I think it's a litte early to make any recommendations. All I can say is that the T.38 support on the HT486 works reliably if the remote end does not stretch the specifications too much. Cheers, Henning Holtschneider - -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-25, fax +49 231 91596-55 sip [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) iD8DBQFFSLloP9goCV2uudcRAg6dAJ9QQrTBvyCt7vEPO4YV+kXvfHsn6wCgv/5I EKunq1uf2+mI0+pjm0+yeAw= =TFn1 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom or Cisco Phones?
http://www.aztech.com/prod_iptelephony_ip150.htmlaztech rawks... the lcd has backlighting and methinks is snom inside ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Neat Application for Text to Speech
I was reading Octobers online edition of Wireless Asia and came across a company called CDyne (www.cdyne.com) They build a number of web services applications but among other things they have an application which you can fill out your details on a web page will some time in the future call the number and run a text to speech file http://ws.cdyne.com/NotifyWS/phonenotify.asmx?op=NotifyPhoneBasic There is also a wiki with some other apps here as well http://wiki.cdyne.com/index.php/Main_Page Thought it might be of interest to some people to have something similar for Asterisk; eg get yourself out of a meeting by receiving an urgent phone call, or to remind yourself of something you shouldnt forget by a certain time. Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FXO Cards vs. Channel bank with T1
This is incorrect. The data is still packetized and passed through IP which provides the same echo cancellation and distortion issues as a call that passed through an FXO/FXS card. Ejay Hire -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Sent: Wednesday, November 01, 2006 3:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FXO Cards vs. Channel bank with T1 Dovid B wrote: Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? Thanks. channel bank is more friendly to faxes and modems (v90 can work too) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject
That looks like a rebranded Snom 300 to me. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rosli SukriSent: Wednesday, November 01, 2006 10:16 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject http://www.aztech.com/prod_iptelephony_ip150.htmlaztech rawks... the lcd has backlighting and methinks is snom inside ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help me on Call parking
Hello. In extensions.conf; in the context that is dialed by your internal extensions, add this line. include=parkedcalls This will include the extensions created by the extensions module, and create your extensions 9006-9009. Good luck, Ejay Hire From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of raviprakash sunkara Sent: Wednesday, November 01, 2006 5:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Help me on Call parking Hello Users... I'm Strucked in Call parking... I'm Using the Asterisk-1.1.11 version in My FC5 box, In That there is feature.conf I'm Using SIP channel By using Asterisk + OpenSER snip In Extension.conf .. I'm confused to give the Dial planning.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject
it is, the navigation button is exactly the same, also notice the extreamly short handset cord On 11/1/06, Cory Andrews [EMAIL PROTECTED] wrote: That looks like a rebranded Snom 300 to me. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rosli Sukri Sent: Wednesday, November 01, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject http://www.aztech.com/prod_iptelephony_ip150.html aztech rawks... the lcd has backlighting and methinks is snom inside ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All that being said, any comments on the Grandstorm phones? I've ordered the GS-101 for my house, and I'm seeing the GXP-2000 is VERY inexpensive for a business solution. I see it has room for 4 lines with 7 programmable buttons. I assume I can put a few more lines on the programmable buttons (we have 6 lines at our main location). One last newbie question, I assume if I have an Asterisk PBX at 2 locations in different states, I'll be able to transfer a call that comes into location1 to a user at location2. Thanks again for the quick responses help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Wednesday, November 01, 2006 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Newbie Questions Ken If these are older comdials then they are just analog phones with extra signaling. The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do. Always remember that it would cost the companies a ton to make every system totally closed That being said, the entry price for IP phones or ADSI phones can be much lower than you think. Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection. That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option was available. Thanks for any help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compilation problem with asterisk-addons
Erick Perez wrote: Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this: Note: MySQL libraries are installed and the structure is as follows: /usr/src/astsources/asterisk-1.2.13 /usr/src/astsources/asterisk-addons-1.2.5 in /usr/src/astsources/asterisk-addons-1.2.5 I do: make clean make and the output is: ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:25:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:26:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:27:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:28:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:29:29: asterisk/module.h: No such file or directory You need to install Asterisk before trying to compile and install Asterisk-addons. -- Russell Bryant Software Engineer Digium, Inc. begin:vcard fn:Russell Bryant n:Bryant;Russell org:Digium, Inc. adr:;;150 West Park Loop;Huntsville;AL;35806;USA email;internet:[EMAIL PROTECTED] title:Software Engineer tel;work:+1-256-428-6000 x-mozilla-html:FALSE url:http://www.digium.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which IP phones have best voice quality, preferably under $150
Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 andLinksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200.-- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk manager
I am trying to send commands to Asterisk manager via a telnet session. I am able to lo in and receive event logs from AMI, but when I try to issue commands I get an invalid/unknown command error. Here are some of the commands I am trying to send. Asterisk Call Manager/1.0 Action: login Username: xxx Secret: x Response: Success Message: Authentication accepted ACTION: Originate Channel: Local/1656 Exten: 1710 Priority: 1 Context: it Response: Error Message: Invalid/unknown command ACTION: Command command: show dialplan Response: Error Message: Invalid/unknown command Action: Originate Channel: Zap/g1/17329250730 Context: default Exten: 1656 Priority: 1 Callerid: 3125551212 Response: Error Message: Invalid/unknown command Here is how my manager.conf file looks [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 displayconnects = yes [ami] secret = XXX permit=0.0.0.0/0.0.0.0 ;deny=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Any help would be greatly appresiated Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [SPAM HEADER] - RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones? - Email found in subject
Ken - take a look at using IAX protocol to route calls between your Asterisk boxes. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams Sent: Wednesday, November 01, 2006 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM HEADER] - RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones? - Email found in subject Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All that being said, any comments on the Grandstorm phones? I've ordered the GS-101 for my house, and I'm seeing the GXP-2000 is VERY inexpensive for a business solution. I see it has room for 4 lines with 7 programmable buttons. I assume I can put a few more lines on the programmable buttons (we have 6 lines at our main location). One last newbie question, I assume if I have an Asterisk PBX at 2 locations in different states, I'll be able to transfer a call that comes into location1 to a user at location2. Thanks again for the quick responses help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Wednesday, November 01, 2006 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Newbie Questions Ken If these are older comdials then they are just analog phones with extra signaling. The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do. Always remember that it would cost the companies a ton to make every system totally closed That being said, the entry price for IP phones or ADSI phones can be much lower than you think. Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection. That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option was available. Thanks for any help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. -- Zeeshan A Zakaria Zeeshan, Anything from Polycom - IP 301, IP 430. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrading from 1.0.9 to 1.2.6
Hi, I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk version. What do I need to be aware of? I AM aware 1.2.6 is not the newest version, but anything above .6, at this time, seems to have stability issues (I've tried them on multiple machines) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF over IAX
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the next dialplan fall through. It happens 60-70% of the time.
extentions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
;OEM
exten = _12125551212,1,Goto(OEM,s,1)
[OEM]
exten = s,1,Answer()
exten = s,n,Set(CALLERID(name)=OEM - ${CALLERID(number)})
exten = s,n,Background(Outsource)
exten = s,n,WaitExten(10)
exten = s,n,Goto(inside,133,1)
exten = 9,1,Background(OEM_Menu)
exten = 9,n,WaitExten(10)
exten = 9,n,Goto(0,1)
exten = 0,1,Goto(inside,133,1)
IAX.conf
[general]
jitterbuffer=yes
forcejitterbuffer=no
maxjitterbuffer=500
autokill=yes
; -
; IAX INCOMING USER
;
; This is the user for incoming calls from:
; connect02.voicepulse.com
; -
[voicepulse] ; -- Name must be [voicepulse]
context=voicepulse-in ; -- Should match the context you
; are using in extensions.conf
; to handle incoming calls
type=user
host=connect02.voicepulse.com
qualify=yes
notransfer=yes
disallow=all
allow=g729 ; -- List supported codecs
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=g726
allow=adpcm
allow=lpc10
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RE: [SPAM HEADER] - [asterisk-users] Which IP phones have best voice quality, preferably under $150 - Email found in subject
I'd recommend any of the following, which are all in your price range Snom 300 Polycom IP430 Polycom IP501 Aastra 9112i Linksys SPA-922 Grandstream GXP-2000 Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan ZakariaSent: Wednesday, November 01, 2006 11:17 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [SPAM HEADER] - [asterisk-users] Which IP phones have best voice quality,preferably under $150 - Email found in subject Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 andLinksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200.-- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
The quality issues you describe may not be the fault of the individual phones! The quality of the PSTN connection, and the hardware through which it's connected can play as big a part in this scenario. The quality of the internal network with 50 IP Phones could also be part of the problem. It sounds like cost is a leading factor in purchasing decisions, so if the same factors applied to building wiring and network equipment, you could have much bigger problems than just the quality of the individual phone sets. Has there been a network study using tools such as Qcheck to determine whether the LAN is capable of handling 50 IP phones? What about the server on which Asterisk is running? Again, with cost being a factor, is this server capable of supporting the load? Does it have sufficient memory and capacity to run efficiently? I would evaluate my infrastructure before I spent more money replacing 50 phones. I certainly wouldn't want to be in front of the boss's desk trying to explain why we spent all this money on new phones and the voice quality still sucks! Good luck! Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${CALLERIDNUM}
At 03:53 AM 11/1/2006, you wrote:
exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
This will tell it to jump to 5 if callerID if but how do i tell
it do jump based on length of callerID?
exten = ,1,GotoIf($[1${CALLERIDNUM} = 1999]?5)
Ira
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Re: [asterisk-users] Snom or Cisco Phones?
Writing this as a user of VoIP and not a reseller, (meaning off the record), we really love the Snom phones here as well, I wish the Snom 300's had a bit more functionality (like the Grandstream), and the Snom 320's and 360's were a little less confusing with their buttons (aka too many buttons on the keypad for some non-tehcy end users).But they have terrific functionality and great audio quality in most office environments, and are very easy to set up and install. Everyone seems to really love them.The Cisco phones are nice as well, but IF you decide to go with Cisco, READ what you are buying and what you are getting before just blindly buying it (in fact, do this anyways, it's common sense to do this before buying any product, anywhere). Cisco products normally don't come with half of the items you need, and unfortunately most resellers (and Cisco) don't make this too easy to read and understand. DO NOT buy refurbished Cisco if you want support, especially since there has been some bogus Cisco voice equipment shipping lately from some of the certified Cisco resellers/distributors. Network World had an article on this recently: http://www.networkworld.com/news/2006/102306counterfeit.htmlCisco may have a great look to their phones and have the design very well thought out (not to mention the big Cisco name - which is good enough for some), but they are normally harder to install and configure and are VERY proprietary. If you buy Cisco, Cisco wants you to ONLY buy Cisco (for support and marketing reasons).Snom 320's are a great choice just because these phones mainly support everything the Snom 360's support (i.e. sidecars) Only main differences between these two models is that the Snom 360 has the larger LCD screen as well as newly added XML support.We have about 50 stations here, some management, some support, some sales and have pretty much decided as a company to completely use Snom phones for all of our employees.Keep in mind, each phone out there will have their specific pro's and con's, as well as quarks ... seems there is no real "perfect phone" out there yet. But Snom in my mind, is pretty dang close.This all of course is just personal opinion from past experience.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 8:35 AM, Tom Vile wrote:I love the Snom phones as well. The function keys are great and easy to use.On 10/31/06, mitcheloc [EMAIL PROTECTED] wrote:My vote is definitely for Snom, I've worked with Cisco phones foryears, but the Snom is much better integrated, and the feature buttons can be retooled for any environment, making custom installs very easy.On 10/31/06, Conrad Wood [EMAIL PROTECTED] wrote: On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote: Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia cars Not really. Both are very good phones. * My Clients prefer cisco because it looks more business-like. - The new snom phones do look better though and the side car rules. * The Cisco phone 'feels' very good in your hand, and the voicequality is superb. (I'd say slightly better than that of the snom 360) * Technically, I find the snom phone more advanced and I can do more cool stuff with it - Cisco doesn't seem to like giving features away in SIP. * Snom phones, for example, have freely programmable buttons that can park/retrieve/transfer calls, show line status etc. I can't get that to work with Cisco phones at all. * Putting custom ringtones (and choosing which ones to use) is a no-brainer with snoms and real trouble with ciscos. * On ciscos, I find the "upgrade" path from sccp to sip a totally unnecessary annoyance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Mitchel ConstantinSnap - A desktop user interface for Asteriskwww.snapanumber.com___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
I tend to stay away from the Grandstream phones for business use because they simply break to easily. I would suggest using Snom phones like the Snom 300 for around $99.2 Asterisk boxes in different locations? Sure, you can do that and its quite easily. On 11/1/06, Ken Williams [EMAIL PROTECTED] wrote: Thanks everyone for the input.After pricing everything we need out,it's not worth trying to get our old system to work, so I've pitchedditching everything and starting over.I'm very excited and hopingthey'll go for it. Regardless, I'm going to throw a box together for my house, we have nohome phone (just cell phones) so this'll be a great way of testing.All that being said, any comments on the Grandstorm phones?I've ordered the GS-101 for my house, and I'm seeing the GXP-2000 is VERYinexpensive for a business solution.I see it has room for 4 lines with7 programmable buttons.I assume I can put a few more lines on the programmable buttons (we have 6 lines at our main location).One last newbie question, I assume if I have an Asterisk PBX at 2locations in different states, I'll be able to transfer a call thatcomes into location1 to a user at location2. Thanks again for the quick responses help.-Original Message-From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]] On Behalf Of AndrewLathamSent: Wednesday, November 01, 2006 5:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Re: Newbie Questions KenIf these are older comdials then they are just analog phones with extrasignaling.The extra signaling could be on the main twisted pair(likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do.Always remember that it would cost the companies aton to make every system totally closedThat being said, the entry price for IP phones or ADSI phones can bemuch lower than you think.Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with.You can order the Aastra phones from your local electrical supplycompany (the place with a long counter and lots of electricians drinking coffee ordering their parts.).AndrewOn 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection.That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option wasavailable. Thanks for any help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problemsthan my email!Hind sight is most always 20/20 or better.---___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk manager
Sorry, but I failed to mention that I am running Asterisk BE B 1-1 I am trying to send commands to Asterisk manager via a telnet session. I am able to lo in and receive event logs from AMI, but when I try to issue commands I get an invalid/unknown command error. Here are some of the commands I am trying to send. Asterisk Call Manager/1.0 Action: login Username: xxx Secret: x Response: Success Message: Authentication accepted ACTION: Originate Channel: Local/1656 Exten: 1710 Priority: 1 Context: it Response: Error Message: Invalid/unknown command ACTION: Command command: show dialplan Response: Error Message: Invalid/unknown command Action: Originate Channel: Zap/g1/17329250730 Context: default Exten: 1656 Priority: 1 Callerid: 3125551212 Response: Error Message: Invalid/unknown command Here is how my manager.conf file looks [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 displayconnects = yes [ami] secret = XXX permit=0.0.0.0/0.0.0.0 ;deny=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Any help would be greatly appresiated Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-114 noise
Jason,There are a couple things we can try to fix your problem.Your firmware shouldn't be an issue, but latest I've got now is: MP118_SIP_F4.80A.034.004.cmpLet's try some quick things first though:In your web interface, go to advanced config - channel settings / voice settingsThere are some options here you can play with:"Voice Volume" (IP side of this thing) - by default this should be set at '1'. Try bringing this down slowly, I'd say in increments of 5 (-4, then -9, and so on).Range on this option can be anywhere from -32 to +32, you really shouldn't need to go beyond -15; but you're actual volume on the calls should still stay reasonable."Input Gain" (telco side) is another option you can slowly change as well (set to 0 by default).There should also be spot where you can specify the "codername", you could possibly try changing this to another codec such as G.729 or G.711u-law (should be the same codec being used on your Asterisk system) try changing packet size from 20 to 40 or 60. This may also help.If none of this stuff helps, let me know. We can then start getting really technical.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 1:45 AM, Jason Kim wrote:Thank you Jessee,Firmware seems to be recent(4.80A.025.004).For 'noisy', I mean IP Phone -- * -- MP-114 side.Audio quality of MP-114 -- PSTN -- Analog phone isgood.I think it can be power ground or gain problem.Any experience?Thanks,Jason--- Jessee J Holmes [EMAIL PROTECTED] wrote: Dear Jason,Please define better noisy? You talking echo issues?Is it on just your side or on the called party's side as well?This start happening immediately, or was the boxworking before and the problem just started?Also, a quick heads up, make sure before evenbeginning to troubleshoot an issue like this you do a factoryreset to the unit and get the latest available firmware on it. Usuallythat fixes annoying issues like this.Thanks,Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIPstore at http:// voipstore.atacomm.com/On Oct 30, 2006, at 10:36 PM, Jason Kim wrote: It's noisy while talking.Any idea?Thanks in advance.Jason __ __Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com)___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com--asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Low, Low, Low Rates! Check out Yahoo! Messenger's cheap PC-to-Phone call rates (http://voice.yahoo.com)___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. The total cost for Cisco phones is fairly high. The phones can be found for about the same price as Polycoms, but if you want to be LEGAL, the SIP firmware can cost over $100. If you do not want to be legal, then you can get the firmware for under $20. Cisco phones generally support PoE. I recommend Polycom phones. You can find them for well under $200, they include SIP firmware. PoE adapters can be purchased for $20 - $40 each if you need PoE. Polycom does not provide firmware to end users, you need to get it from the place you bought your Polycom from. There is at least one web site that you can get greymarket firmware for Polycom phones. Polycom SoundPoint 301 has a low quality display, no handsfree microphone (for speakerphone). But it works just fine as a basic phone. People have said that the 301's sound quality is less than the other Polycom phones. You can also get the 430, which is supposed to have a handsfree microphone and a better display. I've not used them. Also look at the 501s for a higher end phone. You should be able to get the 301s for well under $150 (I've heard people say they have gotten the 301s for $120) and the 501s for well under $200. I assume the 430s are somewhere in between. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6
Matt wrote: Hi, I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk version. What do I need to be aware of? I AM aware 1.2.6 is not the newest version, but anything above .6, at this time, seems to have stability issues (I've tried them on multiple machines) /path/to/src/asterisk/docs/UPGRADE.txt or similar file name. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [asterisk-users] Which IP phones have best voice quality, preferably under $150
snom 300 : CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kristian Kielhofner Gesendet: Mittwoch, 1. November 2006 12:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Which IP phones have best voice quality,preferably under $150 Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. -- Zeeshan A Zakaria Zeeshan, Anything from Polycom - IP 301, IP 430. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Which IP phones have best voice quality, preferably under $150
This e-mail, including attachments, contains privileged and confidential information intended only for the use of the addressee(s) name above. If you are not the intended recipient of this e-mail, or an authorized employee or agent responsible for delivering it to the intended recipient, please be aware that the unauthorized use, dissemination, distribution or reproduction of this e-mail, including attachments, is strictly prohibited and may be unlawful. The Aastra 9133i is close to that price. They work great. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Wednesday, November 01, 2006 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Which IP phones have best voice quality,preferably under $150 Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. -- Zeeshan A Zakaria Zeeshan, Anything from Polycom - IP 301, IP 430. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF over IAX
The problem is voicepulse, but they refuse to accept responsibility. From What phone are you pressing the DTMF?On 11/1/06, Jason Walker
[EMAIL PROTECTED] wrote:Ok sorry for not being specific.I am having a problem when people
outside call in to my number which terminates at VoicePluse then Thesend IAX to me and I do not get any tones. People press buttons but itjust goes to the next dialplan fall through.It happens 60-70% of the time.
extentions.conf[general]static=yeswriteprotect=noautofallthrough=yesclearglobalvars=nopriorityjumping=no;OEMexten = _12125551212,1,Goto(OEM,s,1)[OEM]exten = s,1,Answer()
exten = s,n,Set(CALLERID(name)=OEM - ${CALLERID(number)})exten = s,n,Background(Outsource)exten = s,n,WaitExten(10)exten = s,n,Goto(inside,133,1)exten = 9,1,Background(OEM_Menu)
exten = 9,n,WaitExten(10)exten = 9,n,Goto(0,1)exten = 0,1,Goto(inside,133,1)IAX.conf[general]jitterbuffer=yesforcejitterbuffer=nomaxjitterbuffer=500autokill=yes; -
; IAX INCOMING USER;; This is the user for incoming calls from:; connect02.voicepulse.com; -
[voicepulse] ; -- Name must be [voicepulse]context=voicepulse-in; -- Should match the context you ; are using in extensions.conf ; to handle incoming calls
type=userhost=connect02.voicepulse.comqualify=yesnotransfer=yesdisallow=allallow=g729 ; -- List supported codecsallow=ulawallow=alaw
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Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
I have the Budgetone 101 and GXP2000 and thought the sound quality was excellent. Even over the internet... I agree with Joe that something else may be the factor... Todd Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DTMF over IAX
I had the same problem trying to use an iaxy for an overhead paging system.
SIP has an option to set DTMF to inline, but iax does not.
There was nothing I could do to get the iaxy to play audible DTMF tones.
I had to use a SIP ATA for my paging system with the inline DTMF option.
Note: The DTMF was to preselect the zone.
--
--
Steven
http://www.glimasoutheast.org
Jason Walker [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Ok sorry for not being specific. I am having a problem when people outside
call in to my number which terminates at VoicePluse
then The send IAX to me and I do not get any tones. People press buttons but
it just goes to the next dialplan fall through. It
happens 60-70% of the time.
extentions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
;OEM
exten = _12125551212,1,Goto(OEM,s,1)
[OEM]
exten = s,1,Answer()
exten = s,n,Set(CALLERID(name)=OEM - ${CALLERID(number)})
exten = s,n,Background(Outsource)
exten = s,n,WaitExten(10)
exten = s,n,Goto(inside,133,1)
exten = 9,1,Background(OEM_Menu)
exten = 9,n,WaitExten(10)
exten = 9,n,Goto(0,1)
exten = 0,1,Goto(inside,133,1)
IAX.conf
[general]
jitterbuffer=yes
forcejitterbuffer=no
maxjitterbuffer=500
autokill=yes
; -
; IAX INCOMING USER
;
; This is the user for incoming calls from:
; connect02.voicepulse.com
; -
[voicepulse] ; -- Name must be [voicepulse]
context=voicepulse-in ; -- Should match the context you
; are using in extensions.conf
; to handle incoming calls
type=user
host=connect02.voicepulse.com
qualify=yes
notransfer=yes
disallow=all
allow=g729 ; -- List supported codecs
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=g726
allow=adpcm
allow=lpc10
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Ken, Also stay away from Swissvoice phones I have found several ways to do the second thing. http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers It works great. Jason Tom Vile wrote: I tend to stay away from the Grandstream phones for business use because they simply break to easily. I would suggest using Snom phones like the Snom 300 for around $99. 2 Asterisk boxes in different locations? Sure, you can do that and its quite easily. On 11/1/06, Ken Williams [EMAIL PROTECTED] wrote: Thanks everyone for the input.After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over.I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All that being said, any comments on the Grandstorm phones?I've ordered the GS-101 for my house, and I'm seeing the GXP-2000 is VERY inexpensive for a business solution.I see it has room for 4 lines with 7 programmable buttons.I assume I can put a few more lines on the programmable buttons (we have 6 lines at our main location). One last newbie question, I assume if I have an Asterisk PBX at 2 locations in different states, I'll be able to transfer a call that comes into location1 to a user at location2. Thanks again for the quick responses help. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Andrew Latham Sent: Wednesday, November 01, 2006 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Newbie Questions Ken If these are older comdials then they are just analog phones with "extra signaling".The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do.Always remember that it would cost the companies a ton to make every system totally closed That being said, the entry price for IP phones or ADSI phones can be much lower than you think.Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection.That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option was available. Thanks for any help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Managment tools
Does anyone have a management tool for Polycom phones? For instance something to view software and boot versions of all the phones? I am looking for a product to remotely mange all phones in the environment without having to connect to each phones web config individually. Thanks Clint Neider Email Administrator [EMAIL PROTECTED] Alta Resources | IT Application Services | 120 N Commercial St | Neenah, WI 54956 | Office (920) 751-5800 x 7472 | This email message is intended only for the addressee(s) and contains information that may be confidential and/or copyright. If you are not the intended recipient please notify the sender by reply email and immediately delete this email. Use, disclosure or reproduction of this email by anyone other than the intended recipient(s) is strictly prohibited. No representation is made that this email or any attachments are free of viruses. Virus scanning is recommended and is the responsibility of the recipient. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960 password/shared secret problem --- Related to OS X ?
Hello, Whenever I put in a password/Shared Secret in my 7960 and try and get it to register with asterisk on OS X setup, the phone fails to register. Oct 31 20:03:46 NOTICE[989]: chan_sip.c:11045 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '67.121.71.120' - Wrong password When I make the password blank in both sip.conf and SIP.cnf then it registers and works perfectly. The same phone connects with a linux based [EMAIL PROTECTED] installation without any problems. 3 Questions: 1) How can I tell the actual password/shared secret the phone is transmitting to Asterisk. 2) Has anyone had any trouble configuring 7960s with passwords via the SIP.cnf file? 3) Do you think there is a bug in Asterisk that comes out when its compiled for PowerPC OS X ? Please let me know if you have any suggestions too. Mark Engelhardt Here is my setup: I have a Cisco 7960s running sip image : P003-08-4-00 I am setting the passwords via tftp server and the SIPmacaddress.cnf file. Line 4 on the phone: Asterisk on OS X : Asterisk 1.2.10 running on Mac OS X Server 10.4 (tiger) The portion of the .cnf file: line4_name: 575 line4_authname: 575 line4_password: line4_displayname: 575 line4_shortname: 575 Line 1 on the phone: Asterisk on Linx Asterisk 1.2.5 running on Linux line1_name: 701 line1_authname: 701 line1_password: password line1_displayname: 701 line1_shortname: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Still no CLI in 1.4 branch (OSX)
I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?). I am currently running SVN-branch-1.4-r46716. Any ideas on why this might be, or how to figure out how to fix it? Thanks, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simultaneous ring - call groups or queues orsomething else?
Dovid B wrote: Read the book Asterisk: The future of Telephony http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 It will teach you a lot. The trouble with this (I have it) is that it's dated. I do wish we had a more structured and maintained documentation project. voip-info.org is okay, but there's lots of dated and contradictory information there, too. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk both behind a NAT and outside at the same time
BT == Brad Templeton [EMAIL PROTECTED] writes: BT The correct behaviour, as I see it is: BT a) Native bridge when connecting two external channels -- BT everybody is on the real internet b) Native bridge when connecting BT two internal channels -- everybody is on the 192.168.* network c) BT Route RTP through Asterisk when connecting internal and external BT d) When a channel is to a device behind a remote NAT, the usual BT rules apply (either use STUN or other smart NAT, or route RTP BT through Asterisk) You won't get asterisk to do what you want. That kind of logic simply isn't implemented, and no amount of fiddling with configuration files will make it happen. I'm sure patches are welcome. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DTMF over IAX
On 2006-11-01 08:28:28 -0800, Jason Walker [EMAIL PROTECTED] said: Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. Sounds like VoicePulse should be supporting this issue, as it seems like their problem... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards vs. Channel bank with T1
On Wed, Nov 01, 2006 at 09:08:43AM -0600, Ejay Hire wrote: This is incorrect. The data is still packetized and passed through IP which provides the same echo cancellation and distortion issues as a call that passed through an FXO/FXS card. The issue here is an implementation bug of Zaptel rather than a fundemental archtectual flaw. For fax or modem to work well you need a good line. One of the problems that may cause line quality problems is different clock speeds of different components of the system. They may cause an occasional click every number of seconds. The problem I referred to is that different Zaptel cards may have a different clock. Asterisk uses the clock of the master zaptel device, but it is not exactly clear who that master device is (basically: the first Zaptel device). No other device tries to get clocking from it. If you use an external channel bank you work around the problem by connecting all the external connections (both PRI lines and channel bank FXO/FXS lines) through the same PRI card. That card will not have a problem being in sync with itself. As for our device: our short-term solution is to sync the PC clock from Zaptel as we can already sync our device from the PC. But the long term solution is to sync our device (and other zaptel devices) from the master zaptel device. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: 1.4 branch on OSX?
Martin Joseph wrote: Good news! I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems to have resolved the asterisk hogging the whole CPU issue. I still can't use the regular console though (asterisk -r) as that is unresponsive. Using asterisk -c to start it , works and gives me a color CLI too. At least now it's working well enough to test a bit for real... Awesome to see all those changes and fixes flowing in. This project is really pretty incredible. Thanks to all who contribute and make this possible! Marty This issue has been resolved on the latest 1.4 branch and trunk. Turns out that poll() is broken enough that it goes funky when used on the console stuff. -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6
Thanks for the suggestions.. there is no such document in 1.2.6 in docs. On 11/1/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Matt wrote: Hi, I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk version. What do I need to be aware of? I AM aware 1.2.6 is not the newest version, but anything above .6, at this time, seems to have stability issues (I've tried them on multiple machines) /path/to/src/asterisk/docs/UPGRADE.txt or similar file name. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Opinions on the best wholesale origination/term providers
I wanted to buy service from SellVoip, however I have NEVER been able to reach anyone via phone, and I never really got email responses from them either.I have recommened a few times ISPhone ( www.isphone.net) however they don't have nationwide DIDs.On 11/1/06, Brad Templeton [EMAIL PROTECTED] wrote:On Tue, Oct 31, 2006 at 08:03:56PM -0800, Martin Joseph wrote: On 2006-10-31 17:29:47 -0800, Brad Templeton [EMAIL PROTECTED] said: I've been losing patience with my current provider, a small company called Sellvoip.Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service.So I'm shopping. I also use Sellvoip and I am close to them (Seattle).They by FAR produce the best call quality for me, when compared to nufone and Teliax, although both of those companies do ok, my routes to them aren't nearly as clean. I recommend Teliax for good support.Their DIDs ($5/month plus 2 cents/minute) are much too high, their termination is 2 cents which is tolerable but in generaltoo high for a wholesale service.But thanks for the comment.The sellvoip guys (guy?) are indeed producing good quality.Anotherthing they are doing, which I really like, is processing termination quickly, in that when I do the invite it's ringingwithin a fraction of a second. A few other termination providersI have tried are taking 3-4 seconds to ring after invite.You thought I wrote a lot and I didn't even put that on my list.We just have to convince Jed at Sellvoip to hire somesome support techs, even if he has to add a couple of tenthsper minute.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Java WEB Phone
Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance! Visita www.tutopia.com y comienza a navegar ms rpido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
Hello, The problem was wrong contexts defined like Marco said, and is solved. Now, i have another problem...of course :) On incoming calls, i only can receive calls if i define a line like the following, in extensions.conf: exten = _.,n,Dial(SIP/500,30,tr) (all incoming calls are redirected to extension 500). The problem is that i have some DDI's assigned by my telco (xxx302500 to xxx302509) and i need to route each DDI to diferent internal extension. If i define someting like exten = _0,n,Dial... (for DDI xxx302500) the call is not answered by asterisk. I think that asterisk cannot identify the destination DDI of the incoming call...is this normal? This is the capi debug of one incoming call: asterisk1*CLI CONNECT_IND ID=001 #0x1975 LEN=0045 Controller/PLCI/NCCI= 0x401 CIPValue= 0x10 CalledPartyNumber = 810 CallingPartyNumber = 00 83X CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default -- CONNECT_IND (PLCI=0x401,DID=0,CID=X,CIP=0x10,CONTROLLER=0x1) ISDN1#02: msn='*' DNID='0' MSN == ISDN1#02: setting format alaw - 0x8 (alaw) == ISDN1#02: Incoming call 'X' - '0' INFO_IND ID=001 #0x1976 LEN=0017 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x70 InfoElement = 810 INFO_RESP ID=001 #0x1976 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element CALLED PARTY NUMBER ISDN1#02: INFO_IND DID digits not used in this state. INFO_IND ID=001 #0x1977 LEN=0015 Controller/PLCI/NCCI= 0x401 InfoNumber = 0xa1 InfoElement = default INFO_RESP ID=001 #0x1977 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element Sending Complete CONNECT_RESP ID=001 #0x1977 LEN=0032 Controller/PLCI/NCCI= 0x401 Reject = 0x1 BProtocol B1protocol = 0x0 B2protocol = 0x0 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default ConnectedNumber = default ConnectedSubaddress = default LLC = default AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default INFO_IND ID=001 #0x1978 LEN=0016 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x18 InfoElement = 81 INFO_RESP ID=001 #0x1978 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element CHANNEL IDENTIFICATION 81 INFO_IND ID=001 #0x1979 LEN=0015 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x8005 InfoElement = default INFO_RESP ID=001 #0x1979 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element SETUP ISDN1#02: IE SETUP / SENDING-COMPLETE already received. DISCONNECT_IND ID=001 #0x197b LEN=0014 Controller/PLCI/NCCI= 0x401 Reason = 0x0 DISCONNECT_RESP ID=001 #0x197b LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: DISCONNECT_IND on incoming without pbx, doing hangup. CAPI/ISDN1/0-15: set channel task to 1 == ISDN1#02: CAPI Hangingup for PLCI=0x401 in state 4 == ISDN1#02: Interface cleanup PLCI=0x401 CAPI devicestate requested for ISDN1/0 Anyone can give me ideas about this problem? Thanks in advance! Best regards, PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote-Party-Id and Attended Transfers
Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and the From: should be the party doing the attended transfer. This seems like a bug. - Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Hi Andrew, I can highly recommend using the Granstream GXP 2000. Upgrade the firmware to ver. 1.1.1.14 and you won't have any problems. The 4 line buttons are not actual lines they are calls queued up on an extension so you can have as many incoming lines as you want. The first call comes in on line 1 second simulatanoius call on line 2 etc. The main features that make this a great deal is POE if you want it and dual ports (so you can plug a computer into the back of the phone, plug the phone into the LAN and away you go!) The 7 buttons down the side can be programmed as DSS/BLF, Speed dial buttons or just to show if an extension is registered which is very useful if you use softphones. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All that being said, any comments on the Grandstorm phones? I've ordered the GS-101 for my house, and I'm seeing the GXP-2000 is VERY inexpensive for a business solution. I see it has room for 4 lines with 7 programmable buttons. I assume I can put a few more lines on the programmable buttons (we have 6 lines at our main location). One last newbie question, I assume if I have an Asterisk PBX at 2 locations in different states, I'll be able to transfer a call that comes into location1 to a user at location2. Thanks again for the quick responses help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Wednesday, November 01, 2006 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Newbie Questions Ken If these are older comdials then they are just analog phones with extra signaling. The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do. Always remember that it would cost the companies a ton to make every system totally closed That being said, the entry price for IP phones or ADSI phones can be much lower than you think. Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection. That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option was available. Thanks for any help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
I strongly recommend you upgarde to the latest firmware for the GXP 2000. I have been using them for almost a year now and while the early firmware was poor they are now very stable and working fine (from 1.1.1.9) onwards. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing
Khaled wrote: Dear How can I customize a2billing to have two groups One have service to play its balance and the second group do not play the balance. This is not the a2billing support forum. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] imap on debian
Any potential testers eager to build imap storage support using proper Debian packages: Resonably up-to-date packages of c-client (uw-imap) 2004/2006 are by now only availble from experimental: http://packages.debian.org/experimental/mail/uw-imapd On my test Etch system I simply downloaded the sources of those packages nd rebuilt them (debuild). IIRC with that installed the configure script had enabled imap support in the voicemail. I may be remembering incorrectly. So if after installing this you still have problems, please let me know. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Still no CLI in 1.4 branch (OSX)
Martin Joseph wrote: I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?). I am currently running SVN-branch-1.4-r46716. Any ideas on why this might be, or how to figure out how to fix it? Thanks, Marty I fixed this as of revision 46780 in the 1.4 branch. Give it a go. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simultaneous ring - call groups or queues orsomething else?
On Wed, Nov 01, 2006 at 11:15:23AM -0700, Stephen Bosch wrote: Dovid B wrote: Read the book Asterisk: The future of Telephony http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 It will teach you a lot. The trouble with this (I have it) is that it's dated. I do wish we had a more structured and maintained documentation project. voip-info.org is okay, but there's lots of dated and contradictory information there, too. Could you please take one page there an update it? And please keep that page in your watch list. One page. Nothing more than that. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards vs. Channel bank with T1
On Wed, Nov 01, 2006 at 08:29:32PM +0200, Tzafrir Cohen wrote: On Wed, Nov 01, 2006 at 09:08:43AM -0600, Ejay Hire wrote: This is incorrect. The data is still packetized and passed through IP which provides the same echo cancellation and distortion issues as a call that passed through an FXO/FXS card. The issue here is an implementation bug of Zaptel rather than a fundemental archtectual flaw. For fax or modem to work well you need a good line. One of the problems that may cause line quality problems is different clock speeds of different components of the system. They may cause an occasional click every number of seconds. The jargon is clock slip, and it happens when you don't have your T-1 clocking master/slave hierarchy set up correctly -- or when you have drops from two different switches from two different carriers (local and LD spans, for example). The problem I referred to is that different Zaptel cards may have a different clock. Asterisk uses the clock of the master zaptel device, but it is not exactly clear who that master device is (basically: the first Zaptel device). No other device tries to get clocking from it. If you use an external channel bank you work around the problem by connecting all the external connections (both PRI lines and channel bank FXO/FXS lines) through the same PRI card. That card will not have a problem being in sync with itself. As for our device: our short-term solution is to sync the PC clock from Zaptel as we can already sync our device from the PC. But the long term solution is to sync our device (and other zaptel devices) from the master zaptel device. Well, optimally, every T-1 card should be slaved to it's span, and buffering should take care of keeping various spans in sync with each other. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Managment tools
Neider, Clint wrote: Does anyone have a management tool for Polycom phones? For instance something to view software and boot versions of all the phones? I am looking for a product to remotely mange all phones in the environment without having to connect to each phones web config individually. Thanks Clint Neider Email Administrator [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Alta Resources | IT Application Services | 120 N Commercial St | Neenah, WI 54956 | Office (920) 751-5800 x 7472 | Clint, That is what the Polycom config files are for. You can configure a DHCP server, and TFTP/FTP/HTTP server so that the phones can download their configuration files when they boot. You can also reboot them using SIP NOTIFY with Asterisk. These config files should work for you: http://misc.krisk.org/pcom/ P.S. - Say hi to Neenah for me - my mom is from Little Chute, and my sister was named after Kimberly. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk both behind a NAT and outside at the same time
On Wed, Nov 01, 2006 at 06:16:25PM +0100, Benny Amorsen wrote: BT == Brad Templeton [EMAIL PROTECTED] writes: BT The correct behaviour, as I see it is: BT a) Native bridge when connecting two external channels -- BT everybody is on the real internet b) Native bridge when connecting BT two internal channels -- everybody is on the 192.168.* network c) BT Route RTP through Asterisk when connecting internal and external BT d) When a channel is to a device behind a remote NAT, the usual BT rules apply (either use STUN or other smart NAT, or route RTP BT through Asterisk) You won't get asterisk to do what you want. That kind of logic simply isn't implemented, and no amount of fiddling with configuration files will make it happen. I'm sure patches are welcome. Thanks. Will look into it. Probably need to switch to 1.4 before I start writing more patches though. Though to my surprise I am now discovering something worse. It doesn't seem to work in the lastest 1.2 even with canreinvite=no and nat=yes on the natted (internal) phone with a connection coming in from outside.The outsider has to presume it's calling a natted phone rather than a non-natted asterisk, the invalid SDP is leaking out. I'll see if I can pin that down a bit better. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Managment tools
Neider, Clint wrote: Does anyone have a management tool for Polycom phones? For instance something to view software and boot versions of all the phones? I am looking for a product to remotely mange all phones in the environment without having to connect to each phones web config individually. Thanks Clint Neider Email Administrator [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Alta Resources | IT Application Services | 120 N Commercial St | Neenah, WI 54956 | Office (920) 751-5800 x 7472 | Clint, Oh yeah, I almost forgot - you can see what version of the SIP application the phone is running by (once the phone is registered) executing sip show peer [peer name]. You will see the Polycom phone model and SIP version number in the User Agent: field. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
I think I will agree with folks here, it must be something else on the network, not the phones themselves. I am not going to replace all of the phones, its too expensive, but for trial, want to try something better. PoE is also important to me at this point. I am thinking of trying Linksys 942. I was thinking of Polycom, but there its LCD is not backlit. I keep all LCDs backlit so that is important for me. As for good Aastra phones, there in no external power adapter. Snoms are expensive. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on the best wholesale origination/term providers
Hi Brad,I can confirm the service quality of unlimitel.Have you look at www.les.net they provide both US and Canada DID. I heard good feedback about them On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote:I've been losing patience with my current provider, a small company called Sellvoip.Their termination is good, and they areasterisk based, but they are understaffed and have no conceptof customer service.So I'm shopping.I am interested in the opinions of others on the providers they work with.Here are my criteria, roughly in ordera) Decent quality, low latency.In particular, this means they probably tie into the PSTN atmultiple points, definitely east and west coast and also in Europe.I don't want a California caller calling Californiato have to send their packets to the east coast and back.(This made me discard RNKVoIP, which was high on my list)b) Fair pricing.I've seen blended rates down to a penny, and non-blended down to half/cent in the big city Tier-1s.Idon't expect the lowest possible price but I don't want tosee 100% markup either. For a blended rate, let's seeunder 1.5 cents to the USA and Canada.(Canada is actually down to .8 cents at some providers now, others charge morefor it.)c) Origination, also at a fair priceWhich seems to be about $1/month for DID in USA, $2 inCanada, and close to 1 cent/minute.But I can pay more to get other factors.I guess I can go to another firmfor origination outside the USA in a pinch.d) Reliability very high.Duh.e) Decent customer service.If things go down you fix them and I can reach you to fix them.I don't need handholding, I know my tools, but I do need you to fix problems. If you know your Asterisk, linux and SIP even better.f) Decent automated interface. So I can get DIDs, configure IPs, billing etc.g) Static IP authentication It's faster.Though dynamic IP registration as a backup is handy.h) Global termination I don't want to have to manage and support too many different providers.That's work for me.So give me good global termination prices too. That knocked out termination.com/icall Though if I can't get all I want, I guess I'll buy global from one company and domestic from another.i) No high minimums I am just testing my software apps right now so I'm not going to bill minutes until much later when they ship. So I can't give you tons of minutes per month.I don't mind prepaying.j) SIP, and decently implemented.Asterisk/SER is fine.Now we get to my nice to have listo) IAX as well as SIP. Makes testing stuff easier.o) DTMF via SIP-INFO. This lets me have native bridge for the voice but stillhear the DTMFs at my server, which would be handy.o) Origination worldwideo) Toll free originationo) Cheap toll free termination.(Why does this cost money anyway?) o) Don't want E911 service now.Might want it in future. Don't want to pay now.So here's what I have found that come closesellvoip -- good quality, low latency, good price.Online tools suck, customer service nonexistentrnkvoip -- most of what I want but east coast gateways only.Goodcustomer service but som unreliability in equipment telcommone.net -- Looks fairy good so far.$2/DID in smallquantities, but comes down eventually.Very good term prices.Claims to enforce instate calling prices.(Old world thinking) termination.com -- very good prices but USA onlyterravon -- 1.7 / minute.trxtelecom -- offers free 800 termination, they claim, and pay-you origination in rural latas if that's your style.(Great if you expect most calls to come from cell phones or other peoplewith bundled long distance blended rates.)unlimitel -- for canada netiqsys.net -- no origination but good pricesAny views on these or other providers?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marcel Ericmailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF over IAX
Jason Walker wrote: Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. DTMF issues like this are caused by a problem where the PSTN call is converted to IAX. There is nothing you can do about it unless you manage the box that converts PSTN to IAX. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote-Party-Id and Attended Transfers
Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and the From: should be the party doing the attended transfer. This seems like a bug. - Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6
Sorry, the file is located here: [EMAIL PROTECTED] ~]# ls -l asterisk-1.2.6/UPGRADE.txt -rw-r--r-- 1 1000 1000 8739 Dec 1 2005 asterisk-1.2.6/UPGRADE.txt Matt wrote: Thanks for the suggestions.. there is no such document in 1.2.6 in docs. On 11/1/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Matt wrote: Hi, I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk version. What do I need to be aware of? I AM aware 1.2.6 is not the newest version, but anything above .6, at this time, seems to have stability issues (I've tried them on multiple machines) /path/to/src/asterisk/docs/UPGRADE.txt or similar file name. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration problem
Sergio R. D'Ippolito wrote: Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: */SIP/2.0 401 Unauthorized/* /Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x/ /From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0/ /To: SPA922 sip:[EMAIL PROTECTED];tag=as4da6f6ce/ /Call-ID: [EMAIL PROTECTED]/ /CSeq: 5503 REGISTER/ /User-Agent: incore-PBX/ /Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY/ /WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=372b2479/ Asterisk is asking the phone to resend the registration with WWW-Authenticate using MD5 hash. Make sure the phone supports this and retry. Or you could turn this option off in the sip.conf. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-Id and Attended Transfers
Douglas Garstang wrote: Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and the From: should be the party doing the attended transfer. This seems like a bug. pbx-1*CLI show application dial [snip] o- Specify that the CallerID that was present on the *calling* channel be set as the CallerID on the *called* channel. This was the behavior of Asterisk 1.0 and earlier. [snip] Also from UPGRADE.txt: Dialing: * The Caller*ID of the outbound leg is now the extension that was called, rather than the Caller*ID of the inbound leg of the call. The o flag for Dial can be used to restore the original behavior if desired. Note that if you are looking for the originating callerid from the manager event, there is a new manager event Dial which provides the source and destination channels and callerid. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 1.4 branch on OSX?
On 2006-11-01 10:42:12 -0800, Joshua Colp [EMAIL PROTECTED] said: Martin Joseph wrote: Good news! I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems to have resolved the asterisk hogging the whole CPU issue. I still can't use the regular console though (asterisk -r) as that is unresponsive. Using asterisk -c to start it , works and gives me a color CLI too. At least now it's working well enough to test a bit for real... Awesome to see all those changes and fixes flowing in. This project is really pretty incredible. Thanks to all who contribute and make this possible! Marty This issue has been resolved on the latest 1.4 branch and trunk. Turns out that poll() is broken enough that it goes funky when used on the console stuff. Ok, Thanks. My CLI is working again now! Too bad I just started another thread about this, moments ago... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Still no CLI in 1.4 branch (OSX)
On 2006-11-01 09:09:26 -0800, Martin Joseph [EMAIL PROTECTED] said: I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?). I am currently running SVN-branch-1.4-r46716. Any ideas on why this might be, or how to figure out how to fix it? Ok, This was fixed in 1.4-r46802. Nothing to see here... ;~) Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Java Web Phone
Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance! Visita www.tutopia.com y comienza a navegar ms rpido en Internet.Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
On Wed, 1 Nov 2006, Pedro Silva wrote: Hello, The problem was wrong contexts defined like Marco said, and is solved. Now, i have another problem...of course :) On incoming calls, i only can receive calls if i define a line like the following, in extensions.conf: exten = _.,n,Dial(SIP/500,30,tr) (all incoming calls are redirected to extension 500). The problem is that i have some DDI's assigned by my telco (xxx302500 to xxx302509) and i need to route each DDI to diferent internal extension. If i define someting like exten = _0,n,Dial... (for DDI xxx302500) the call is not answered by asterisk. I think that asterisk cannot identify the destination DDI of the incoming call...is this normal? As you can see in the log below, the called number is just '0': CalledPartyNumber = 810 It seems DDI 0 of your line was called. So just do exten = 0,n,Dial... Armin This is the capi debug of one incoming call: asterisk1*CLI CONNECT_IND ID=001 #0x1975 LEN=0045 Controller/PLCI/NCCI= 0x401 CIPValue= 0x10 CalledPartyNumber = 810 CallingPartyNumber = 00 83X CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default -- CONNECT_IND (PLCI=0x401,DID=0,CID=X,CIP=0x10,CONTROLLER=0x1) ISDN1#02: msn='*' DNID='0' MSN == ISDN1#02: setting format alaw - 0x8 (alaw) == ISDN1#02: Incoming call 'X' - '0' INFO_IND ID=001 #0x1976 LEN=0017 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x70 InfoElement = 810 INFO_RESP ID=001 #0x1976 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element CALLED PARTY NUMBER ISDN1#02: INFO_IND DID digits not used in this state. INFO_IND ID=001 #0x1977 LEN=0015 Controller/PLCI/NCCI= 0x401 InfoNumber = 0xa1 InfoElement = default INFO_RESP ID=001 #0x1977 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element Sending Complete CONNECT_RESP ID=001 #0x1977 LEN=0032 Controller/PLCI/NCCI= 0x401 Reject = 0x1 BProtocol B1protocol = 0x0 B2protocol = 0x0 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default ConnectedNumber = default ConnectedSubaddress = default LLC = default AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default INFO_IND ID=001 #0x1978 LEN=0016 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x18 InfoElement = 81 INFO_RESP ID=001 #0x1978 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element CHANNEL IDENTIFICATION 81 INFO_IND ID=001 #0x1979 LEN=0015 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x8005 InfoElement = default INFO_RESP ID=001 #0x1979 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element SETUP ISDN1#02: IE SETUP / SENDING-COMPLETE already received. DISCONNECT_IND ID=001 #0x197b LEN=0014 Controller/PLCI/NCCI= 0x401 Reason = 0x0 DISCONNECT_RESP ID=001 #0x197b LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: DISCONNECT_IND on incoming without pbx, doing hangup. CAPI/ISDN1/0-15: set channel task to 1 == ISDN1#02: CAPI Hangingup for PLCI=0x401 in state 4 == ISDN1#02: Interface cleanup PLCI=0x401 CAPI devicestate requested for ISDN1/0 Anyone can give me ideas about this problem? Thanks in advance! Best regards, PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150
All the phones already have the latest firmware. They keep updating themselves automatically. In my setup of Grandstream phones, all the computers of the network go through the phones, i.e. I am using the builtin phones as swithces. They all have 2 ethernet ports. Does this has to do anything with the voice quality, or do I need to change something in the phones' setup, like switching it from switch to router in basic settings? What is this NAT/Router setting anyways and how should it be setup? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I use Realtime entries to do multiple registers to same trunk/peer
I have a config where I define a single peer and have possibly hundreds of register commands for that single peer.I'm not clear if I can do the register part via Asterisk Realtime (right now I updated a file and force a reload which re-registers all the users defined in the register directives). I want to avoid reload/restart everytime I add a register user to the list:ie:[EMAIL PROTECTED]:[EMAIL PROTECTED][EMAIL PROTECTED]:[EMAIL PROTECTED] [EMAIL PROTECTED]:[EMAIL PROTECTED]replaced with realtime interface in MySQL table.Thanks in advanceTom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TE110P Card Little help
Hi, I really need some assistance in installing and configuring this card. I have already physically installed it into the computer which is running Mandriva 2006. I have compiled and installed asterisk 1.2.13 along with zaptel-1.2.10 and libpri-1.2.4. However I do not know what the next step is. Better yet several sites that list some kind of walk through are completely different from each other. Any help would be greatly appreciated. Thanks Julian From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Mon, 30 Oct 2006 19:45:50 +Subject: [asterisk-users] TE110P Card Hi Groupies,I am sort of new to the whole asterisk thing, especially when it comes to the Digium TE110P card. Does anyone have experience setting this up? If so can you help me out? The provider for the PRI is going to be ATT/SBC.ThanksJulian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] imap on debian
On 21:47, Wed 01 Nov 06, Tzafrir Cohen wrote: Any potential testers eager to build imap storage support using proper Debian packages: Resonably up-to-date packages of c-client (uw-imap) 2004/2006 are by now only availble from experimental: http://packages.debian.org/experimental/mail/uw-imapd On my test Etch system I simply downloaded the sources of those packages nd rebuilt them (debuild). IIRC with that installed the configure script had enabled imap support in the voicemail. I may be remembering incorrectly. So if after installing this you still have problems, please let me know. I'm running etch as well. Tell me how I can help -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Opinions on the best wholesale origination/termproviders
I am testing toll free and US DID inbound as well as A-Z outbound with les.net at the moment. Both the quality and support are quite good. Ping time to Vancouver is around 80ms. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcel Eric Loiselle Sent: Wednesday, November 01, 2006 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Opinions on the best wholesale origination/termproviders Hi Brad, I can confirm the service quality of unlimitel. Have you look at www.les.net they provide both US and Canada DID. I heard good feedback about them On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote: I've been losing patience with my current provider, a small company called Sellvoip.Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service.So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In particular, this means they probably tie into the PSTN at multiple points, definitely east and west coast and also in Europe.I don't want a California caller calling California to have to send their packets to the east coast and back. (This made me discard RNKVoIP, which was high on my list) b) Fair pricing.I've seen blended rates down to a penny, and non-blended down to half/cent in the big city Tier-1s.I don't expect the lowest possible price but I don't want to see 100% markup either. For a blended rate, let's see under 1.5 cents to the USA and Canada.(Canada is actually down to .8 cents at some providers now, others charge more for it.) c) Origination, also at a fair price Which seems to be about $1/month for DID in USA, $2 in Canada, and close to 1 cent/minute.But I can pay more to get other factors.I guess I can go to another firm for origination outside the USA in a pinch. d) Reliability very high.Duh. e) Decent customer service.If things go down you fix them and I can reach you to fix them.I don't need handholding, I know my tools, but I do need you to fix problems. If you know your Asterisk, linux and SIP even better. f) Decent automated interface. So I can get DIDs, configure IPs, billing etc. g) Static IP authentication It's faster.Though dynamic IP registration as a backup is handy. h) Global termination I don't want to have to manage and support too many different providers.That's work for me.So give me good global termination prices too. That knocked out termination.com/icall Though if I can't get all I want, I guess I'll buy global from one company and domestic from another. i) No high minimums I am just testing my software apps right now so I'm not going to bill minutes until much later when they ship. So I can't give you tons of minutes per month.I don't mind prepaying. j) SIP, and decently implemented.Asterisk/SER is fine. Now we get to my nice to have list o) IAX as well as SIP. Makes testing stuff easier. o) DTMF via SIP-INFO. This lets me have native bridge for the voice but still hear the DTMFs at my server, which would be handy. o) Origination worldwide o) Toll free origination o) Cheap toll free termination.(Why does this cost money anyway?) o) Don't want E911 service now.Might want it in future. Don't want to pay now. So here's what I have found that come close sellvoip -- good quality, low latency, good price.Online tools suck, customer service nonexistent rnkvoip -- most of what I want but east coast gateways only.Good customer service but som unreliability in equipment telcommone.net -- Looks fairy good so far.$2/DID in small quantities, but comes down eventually.Very good term prices. Claims to enforce instate calling prices.(Old world thinking) termination.com -- very good prices but USA only terravon -- 1.7 / minute. trxtelecom -- offers free 800 termination, they claim, and pay-you origination in rural latas if that's your style.(Great if you expect most calls to come from cell phones or other people with bundled long distance blended rates.) unlimitel -- for canada netiqsys.net -- no origination but good prices Any views on these or other providers? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marcel Eric mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS and 1.2.12
Can anyone confirm that SMS() works correctly under asterisk 1.2.12? It used to work around version 1.2.7, but a few people have reported that 1.2.8 and 1.2.9 were a bit dodgy and that all their problems went away when they used app_sms from 1.2.7 in the later versions of asterisk. When a fixed line sms comes in, I get: -- Executing SMS(SIP/pstn1-0819a388, 0198339100|a) in new stack -- SMS TX 93 00 6D -- SMS RX 93 00 6D -- SMS TX 94 00 6C -- SMS RX 94 00 6C But nothing gets queued or anything. It almost seems like the transmitted message is just getting echoed back again... Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] +Ura +md3200 nao encaminha ligacao
Salve Salve Galera. Tenho a seguinte situacao: Uma placa MD3200 ligada em uma linha telefonica comum(PTSN) e funcionando belezinha... Tenho configurado um URA, onde ele atende a ligacao que chegou no canal e solicita o numero do ramal de destino da ligação: Acontece que ao discar o ramal de destino, ele nao encaminha a ligacao, ficando mudo e posteriormente caindo a ligaçao. Fiz um teste, criando um extension _199 e encaminhando para o mesmo URA... Ligo de um ramal do asterisk para o 199 e o URA atende, disco o ramal de destino desejado e o mesmo encaminha a ligacao corretamente. ou seja, vindo do zapata nao funciona, direto de um ramal sip funciona.. abaixo esta o meu zapata.conf e o extension.conf. zapata.conf [channels] language=en context=Globalnova signalling=fxs_ks rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 relaxdtmf=no cidsignalling=DTMF cidstart=polarity rxgain=1.0 txgain=1.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=4 musiconhold=default channel = 1 callerid=Linha externa extension.conf exten = s,1,Goto,atendimento|s|1 exten = s,2,Hangup [atendimento] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,10 exten = s,4,Set(TIMEOUT(digit)=10) exten = s,5,ResponseTimeout,10 exten = s,6,Background(vm-enter-num-to-call) exten = i,1,Goto,atendimento|s|1 exten = t,1,Goto,atendimento|s|1 exten = 2010,1,Dial(SIP/2010,50,Trt) exten = 2012,1,Dial(SIP/2012,50,Trt) exten = 2004,1,Dial(SIP/2004,50,Trt) exten = 2021,1,Dial(SIP/2021,50,Trt) Valeu!!! -- []´s .'. Wederson R. .'. [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time
Seems to me that you have a routing problem, asterisk should not know how to send packets to an outside IP using the NATed network. Make sure that the internal (NAT) interface doesn't have a gateway to it. On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote: I've read a lot of the descriptions of handling NAT with Asterisk, and the use of both the nat and canreinvite flags. I am very familiar with Sip and NAT but have not seen an answer to the following question. My Asterisk server runs on a machine with two ethernets. One is an external net, with exposed IP addresses. The other is an internal net with natted IP addresses. Thus the server has two addresses. The server is _not_ the NAT gateway. That's a linksys box which has its own external IP to gateway traffic from the internal natwork. The phones are on the internal NATwork. Asterisk talks to them over it. Outside peers, such as SIP termination providers etc. talk to the Asterisk server via its outside address, which is as you would expect. However, from time to time I get the famous one-way audio because Asterisk has decided to do a native bridge between a natted SIP phone and an external SIP peer. It sends the internal IP of the SIP phone in the SDP and of course the outside service can't send packets to that. I could just turn off reinvites on the internal phones, but this would cause them to route all traffic through the asterisk box, even on internal calls between phones on the same ethernet, which seems foolish to me. I don't want to turn off reinvites to the external peers -- if a call comes in from a SIP originator for example, and is send back out to a SIP terminator (call forwarding) I want a native bridge for sure.(Handling the internal traffic is not so much of a burden though sometimes I hear latency because of it, but routing external traffic through the asterisk box is a bad thing.) So what I want is for Asterisk to use native bridges when connecting two channels behind the NAT, or two channels on the real internet, but not to do so when connecting an internal and external channel. It should be able to see the IP addresses, and know the difference between natted and external ones and know they can't talk to one another. (The ICE protocol would handle this someday.) Is IAX smarter about this? Of course I might even want to get smarter about this. Is it possible, typically by configuring stun in the phones, to have them be aware of their external IP and tell Asterisk about it? With a full cone NAT, it would work to do a native bridge between the internal and external devices so long as the external device is given the right address and port of the NAT box, not the internal address of the phone. However, we don't want to do this on internal to internal calls -- many NATs can't hairpin. I would think this would be a common situation (though perhaps more commonly the asterisk server IS the firewall/NAT.) Is there a solution that does the right thing most of the time? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time
Sorry for my previous post I misunderstood the problem. You should set canreinvite=no to all sip peers that connect from outside. On 10/31/06, C F [EMAIL PROTECTED] wrote: Seems to me that you have a routing problem, asterisk should not know how to send packets to an outside IP using the NATed network. Make sure that the internal (NAT) interface doesn't have a gateway to it. On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote: I've read a lot of the descriptions of handling NAT with Asterisk, and the use of both the nat and canreinvite flags. I am very familiar with Sip and NAT but have not seen an answer to the following question. My Asterisk server runs on a machine with two ethernets. One is an external net, with exposed IP addresses. The other is an internal net with natted IP addresses. Thus the server has two addresses. The server is _not_ the NAT gateway. That's a linksys box which has its own external IP to gateway traffic from the internal natwork. The phones are on the internal NATwork. Asterisk talks to them over it. Outside peers, such as SIP termination providers etc. talk to the Asterisk server via its outside address, which is as you would expect. However, from time to time I get the famous one-way audio because Asterisk has decided to do a native bridge between a natted SIP phone and an external SIP peer. It sends the internal IP of the SIP phone in the SDP and of course the outside service can't send packets to that. I could just turn off reinvites on the internal phones, but this would cause them to route all traffic through the asterisk box, even on internal calls between phones on the same ethernet, which seems foolish to me. I don't want to turn off reinvites to the external peers -- if a call comes in from a SIP originator for example, and is send back out to a SIP terminator (call forwarding) I want a native bridge for sure.(Handling the internal traffic is not so much of a burden though sometimes I hear latency because of it, but routing external traffic through the asterisk box is a bad thing.) So what I want is for Asterisk to use native bridges when connecting two channels behind the NAT, or two channels on the real internet, but not to do so when connecting an internal and external channel. It should be able to see the IP addresses, and know the difference between natted and external ones and know they can't talk to one another. (The ICE protocol would handle this someday.) Is IAX smarter about this? Of course I might even want to get smarter about this. Is it possible, typically by configuring stun in the phones, to have them be aware of their external IP and tell Asterisk about it? With a full cone NAT, it would work to do a native bridge between the internal and external devices so long as the external device is given the right address and port of the NAT box, not the internal address of the phone. However, we don't want to do this on internal to internal calls -- many NATs can't hairpin. I would think this would be a common situation (though perhaps more commonly the asterisk server IS the firewall/NAT.) Is there a solution that does the right thing most of the time? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] connecting internal line with external line
Hi, I'm new to asterisk. I want asterisk to connect a external line with an internal line: the PC dials a number and connects this call to a internal telephone (telephone switchboard, based on ISDN, 4 analogue telephones) of my office. Can somebody here give me keyword how to search (e.g. with google) to realise it? tia Ekkard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
