Re: [Asterisk-Users] Install Xc-Ast $$$
Hello, we make XC-AST and can install it for you, or we can help you installing it. How big is your call center? Under which environment did you try to install it? Thanks l. In data Fri, 10 Dec 2004 14:58:09 -0500, John Bittner [EMAIL PROTECTED] ha scritto: I have spent the last 3 days trying to get this software working. I am now at the point I am willing to pay to get this installed. Anyone that has installed this before and is looking for some cash please email me with price. I need this installed asap. Thanks John Bittner Simlab.net 9734333009 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] long list of prefixes
if a phone number starts with one of 50+ prefixes, i want to send the sip call to gateway X. if it is in any other prefix, i want to send it to gate Y. i am not excited about a long list of extens, but will do it if i have to. i suspect there is a database hack, but i lose all database contents if i reinstall the port (this may be a feature of the freebsd port), and i have not figured out a script that will let me load it. surely there is a well-known and reasonable way out of this corner. but i can not seem to find the right wiki incantation. thanks for clue. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to test enum?
If somebody has done it before and has the time, please contact me off list.
The list is worthless if answers are sent by private mail.
ENUMLOOK=123
; Test ENUM lookup watching the CLI
; a file that says no enulm listing found
; was recorded
exten = _${ENUMLOOK}.,1,EnumLookup(${EXTEN:3})
exten = _${ENUMLOOK}.,2,NoOp(ENUM result: ${ENUM})
exten = _${ENUMLOOK}.,3,Hangup
exten = _${ENUMLOOK}.,102,Playback(noenumfound)
exten = _${ENUMLOOK}.,103,Hangup
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Re: [Asterisk-Users] IAXPeers for Windows Beta released
On Sat, 2004-12-11 at 11:53 +1300, Matt Riddell wrote: Hi, I've just done up a quick proggy to show me the status of my IAX peers from my windows box. It plugs into the simple manager proxy. You can see more information (including a screenshot) at: http://www.sineapps.com/news.php?rssid=384 You can download it directly from: http://www.sineapps.com/down/IAXPeers.zip Could you please have a look and let me know your thoughts. First I like it. I can use it straight away. Only comment at the moment is would it be possible to save the configuration of the Host and the order of the peers for the next start up? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
On Fri, 10 Dec 2004 21:53:53 -0600, nik martin wrote: news.gmane.org wrote: nik martin wrote: Anyone ever thought about an Ethernet based channel bank? Basically a rack mount set of 24 IAXys? That would be cool, IMO. No wrangling with zaptel, etc. IAX as the * - Channel bank protocol. Just an idea... Allied Telesyn VoIP Access Device http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf This is a 24-port FXS 1u device, conveniently presented as a single RJ-21 TELCO connector. yeah, but those are expensive as crap. i was thinking about something more competetive with a channel bank Compare it to the price of 24 x IAXys ?? . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hfc card and isdn error E001B
Dnia pitek, 10 grudnia 2004 20:24, Peer Oliver Schmidt napisa:
Marco Parmeggiani schrieb:
I'm trying to use an hfc based pci card with asterisk but every call
fails falling in the congestion extension.
exten = _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr)
exten = _0.,2,Congestion
Looking in the syslog i can see:
isdn: HiSax,ch0 cause: E001B
isdn card: HFC based, type 35
See error codes, this probably problem with cables or
phisical connection, or you did not load proper drivers.
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Re: [Asterisk-Users] IAXPeers for Windows Beta released
On Sun, 2004-12-12 at 00:00 +1300, Matt Riddell wrote: Dave Cotton wrote: http://www.sineapps.com/down/IAXPeers.zip Could you please have a look and let me know your thoughts. First I like it. I can use it straight away. Cool, that's good to hear! :-) Only comment at the moment is would it be possible to save the configuration of the Host and the order of the peers for the next start up? Ok, done (you will need to download it from the above URL). The logic works as such: 1. When you click the connect button, it will save your hostname to the registry (and will load it when the app starts). 2. If you change one of the items in the dropdown box, it will now start up next time with that same entry allocated (assuming the number of IAX peers you have does not change) 3. I have fixed a little bug with regard to the last bar not going green once the connection returned from above 333ms. Let me know if this is ok for you. Certainly easier for a normal(tm) user. But it's opened up another problem I can't have another instance of the program monitoring another server through the VPN. Message is Run-time error 380 Invalid property value if I change the ip address. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RealTime and Macro question?
Is it possible to call a macro, which is defined in extensions.conf from a realtime extension configured in Mysql. Beacuse when i try i receive an error - no such context. -- Executing Macro(SIP/1007-2165, dialnumber_wvm,1004,SIP/1004) Dec 11 12:51:04 WARNING[22551]: app_macro.c:100 macro_exec: No such context 'macro-dialnumber_wvm,1004,SIP/1004' for macro 'dialnumber_wvm,1004,SIP/1004' Here is what i have in extensions table : id context extenpriority app appdata _ 1sip-internal 10041Macro dialnumber_wvm,1004,SIP/1004 -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to test enum?
Wilson Pickett schrieb:
ENUMLOOK=123
; Test ENUM lookup watching the CLI
; a file that says no enulm listing found
; was recorded
exten = _${ENUMLOOK}.,1,EnumLookup(${EXTEN:3})
exten = _${ENUMLOOK}.,2,NoOp(ENUM result: ${ENUM})
exten = _${ENUMLOOK}.,3,Hangup
exten = _${ENUMLOOK}.,102,Playback(noenumfound)
exten = _${ENUMLOOK}.,103,Hangup
Are there any test numbers where I can see if ENUM lookup is working?
And is it possible as well to test if a number of a SIP or IAX provider
exists?
Bye!
Michael
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[Asterisk-Users] Asterisk 1.0.3 and chan_capi ?
did asterisk 1.0.3 and chan_capi runs together ? thx nico ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXPeers for Windows Beta released
Dave Cotton wrote: But it's opened up another problem I can't have another instance of the program monitoring another server through the VPN. Message is Run-time error 380 Invalid property value if I change the ip address. If you change the ip to what? On my copy here, I can change the IP address of the running copy with no issue. I suspect there's something in the IP address it doesn't like - is it just a normal xxx.xxx.xxx.xxx addy? I'm not quite sure how to get around the problem of the two copies. Even though I can change the IP on a running instance (the hack for this is that the IP address is actually not saved until you click connect - it will not connect if it is already connected, but will save the variable to the registry). Maybe it would be best if you reply to this off-list so that we can get this fixed. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.1 Too many open files
On 09/12/2004 at 09:22 Eric wrote: Hi Sean, Thanks for your reply, but that wasn't exactly what I was getting at. I don't need to increase the system's imposed limit on the number of open files. I'm more concerned to see if anyone has run across a memory or fd leak in asterisk that sucks them all up. There should be no reason that I hit my limit of open files on this machine. Restarting asterisk immediately solved the problem, so I'm leaning towards a leak, however, I didn't have the opportunity, in the moment, to check and see how many files and what type were open. - Eric I'm pretty sure that it's a leak, if I recount a problem I have (had) when trying to register with FWD is should make it obvious. About a week or two ago I started having problems with registering with FWD using SIP, the request was sent but there was never a reply. Indeed a traceroute showed a problem at peer1.net (this is still the case). I noticed that after a few hours I was getting the same errors as you. A restart of asterisk cured the problem temporarily until a few hours later, when it reappeared. incidentally I 'fixed' the issue by using an iax2 connection to fwd instead... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] long list of prefixes
Randy Bush wrote: if a phone number starts with one of 50+ prefixes, i want to send the sip call to gateway X. if it is in any other prefix, i want to send it to gate Y. Take a look at http://www.voip-info.org/wiki-Asterisk+app_dbodbc I run a home server so I have never had the need to do stuff like that but it looks like the thing you want. I'm sure there are other alternatives out there that will do the same only differently... /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Handling raw audio (8000 signed 16bit big-endian)
Title: Message Does anyone know if there is a "format-raw.c" routine available for Asterisk-0.9.0? Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail
Around 1 customers. On Fri, 10 Dec 2004 17:24:56 +0100, Wilson Pickett [EMAIL PROTECTED] wrote: How many customers, Sharon? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy: no dial tone
I have this good looking IAXy device... I have managed to provision it, i can see it registering to my asterisk box, however when I pick up the phone which is plugged in the IAXy I have no dialtone, nothing. What leds are lit? What kind of phone is connected to it? Can you call it? (watch the IAXy, even if phone doesn't ring you can see the led react to ring) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy: no dial tone
What leds are lit? Looking with the orange bit facing you, the network led on the left (network) is permanently lit. The led on the right blinks once every 7 seconds or so. There is also the network plug's led which is lit. That's all. What kind of phone is connected to it? France Télécom Amarys 1400. It's supposed to work either with an AC adapter or even without (lifeline). I've tried both scenarios - no change. It's a pretty standard phone and works fine on a normal landline. Can you call it? (watch the IAXy, even if phone doesn't ring you can see the led react to ring) When I ring it, the led that blinks every 7 seconds or so then starts to blink much faster, by bursts - so it looks like it's working. Except that the phone which is plugged onto it doesn't ring - and if I pick it up nothing happens... I'm thinking that the device came with a 9V 800 milliamp AC adapter, and I have read somewhere on the site that the IAXy needs 1200 milliamp and that it might be the problem. Can anybody confirm? Cheers, Jean-Michel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie MusicOnHold issues
Hi Everyone, Merry Christmas :-)
My Asterisk Box doesn't have a sound card, it is running
Asterisk 1.02
Zaptel 1.02
Libpri 1.02
Mpg123 0.59r
All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2
Any help would be very much appreciated.
The error I am getting is
-- Executing WaitMusicOnHold(SIP/snom-james-849d, 30) in new
stack
Dec 12 00:27:29 WARNING[409616]: res_musiconhold.c:366 moh1_exec: Unable
to start music on hold (class '30') on channel SIP/snom-james-849d
== Spawn extension (sip, 098, 1) exited non-zero on
'SIP/snom-james-849d'
/etc/asterisk/musiconhold.conf
;
; Music on hold class definitions
;
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3
;loud = mp3:/var/lib/asterisk/mohmp3
random = quietmp3:/var/lib/asterisk/mohmp3,-z
I also tried doing a
default = custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -z -q -r 8000
-f 8192 -b 2048 --mono -s
/etc/asterisk/extensions.conf
[pstn]
exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,SetMusicOnHold(random)
exten = s,3,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
exten = s,4,Hangup
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 098,1,WaitMusicOnHold(5)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
include = outgoing
include = sip
[outgoing]
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup
[sip]
exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,tr)
exten = 690,3,voicemail2,u690
exten = 690,102,voicemail2,b690
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,tr)
exten = 691,3,voicemail2,u691
exten = 691,102,voicemail,b691
include = internal
include = outgoing
[from-sip]
include = internal
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip
[snom-james]
type=friend
secret=apassword
host=dynamic
callerid=James Bean 690
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=690
[bt-karen]
type=friend
secret=apassword
host=dynamic
callerid=Karen Colomb 691
defaultip=192.168.69.251
dtmfmode=info
mailbox=691
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Re: [Asterisk-Users] IAXy: no dial tone
Not a European phone expert, but would that phone work on a US POTS telephone network? Is the signalling and ringer voltage the same as US? You're right to put that in question. I've had issues with older Siemens phones (purchased in France) on both IAXy and Digium cards. They don't ring at all without a patch in wcfxs.c to change ring frequency for one thing. At the office our (horribly crappy Alcatel $60 ) phone wokrs ok though, on the Digium TDM400P. I currently have an old phone from the US connected to the IAXy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy: no dial tone
Jean-Michel Hiver wrote: What leds are lit? Looking with the orange bit facing you, the network led on the left (network) is permanently lit. The led on the right blinks once every 7 seconds or so. There is also the network plug's led which is lit. That's all. What kind of phone is connected to it? France Télécom Amarys 1400. It's supposed to work either with an AC adapter or even without (lifeline). I've tried both scenarios - no change. It's a pretty standard phone and works fine on a normal landline. Can you call it? (watch the IAXy, even if phone doesn't ring you can see the led react to ring) Not a European phone expert, but would that phone work on a US POTS telephone network? Is the signalling and ringer voltage the same as US? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor, append audio?
Hello, I have a situation where I need to first check if a previous clip was recorded, and if so, append to it.. Otherwise create a new file.. I'm using Monitor. Monitor automatically calls sox after the call ends.. Is there a way to manually control this process, and instruct sox to append to the destination file, rather than overwrite? Thanks in advance! = Johnathan Proffer Viable Technologies, Inc. = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux basics and Asterisk basics
On Fri, 2004-12-10 at 16:29, Jim Guy wrote: Hello, I am just starting to research Asterisk and I would like to install it on a PC to try out. I have looked around quite a bit but I haven't found much information on the Linux part. I know you need to put Linux on the PC first but what version or flavor of Linux do you recommend? I contacted Red Hat and they had not heard of Asterisk and they said Asterisk is not certified for Red Hat. Are there any Linux installation instructions that you would recommend? If there are any other getting started suggestions, I sure would appreciate it. Tim, Check out http://www.xorcom.com/rapid/ These guys came up with a really neat solution for beginners and have a cool interface utility for logs, stats and other stuff. Just grab an old PC and load up this CD and away you go. Also check out http://www.automated.it/asterisk/ This distro is a live CD with Asterisk installed, no need for a hard drive install, just boot the CD and your up and running with Linux and Asterisk, it works on the Knoppix principles. And definitely check out Knoppix http://www.knoppix.net/ , it's the bomb for newbie's. Good luck and welcome to Asterisk. JR ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 5, Issue 158
I am sorry that I was not more clear. I am only looking for departments that will fit into the string: press 1 for the DEPT department or press 1 for DEPT the 'into' is what I should have been clearer about. I am only looking for words that will fit into the DEPT portion of the above strings. As you mention we already have press, the numbers, for, the, and department (? not sure about this one).The reason I gave those examples was to clarify the context of what I was thinking of. I am struggling right now to think of examples that do not fit but I assure you that I had some when I first wrote that message ;-) Thanks; James Subject: Re: [Asterisk-Users] Voice Prompt Info From: Christopher Dobbs [EMAIL PROTECTED] Date: Fri, 10 Dec 2004 16:24:00 -0800 You should not put the press or the number in the prompt. Have them as separate sounds, that way, they are more generic. EG: Background(press-1-for) Background(sales) Background(and) Background(service) Background(department) Background(press-2-for) Background(Tech) Background(support) -- Christopher Dobbs [EMAIL PROTECTED] wrote: I am trying to put together a list of 'departments' to request as voice prompts. I have the biggies (sales, accounting, shipping, etc...) but I want to make sure I do not miss any. If anyone anyone has some suggestions (Ha... that is like going to an NRA meeting ans asking if anybody has a gun :-) ) please forward them to me (and / or post here although, with the volume of this list I do not always have time to read every digest so the 'and' option may be best.) so that I can compile a single list, verify that they are not already available, group them, and send them on. Please put 'voice prompt' in the subject line of anything you forward me so that I am less likely to miss it. I am looking for titles that fit into the string: press 1 for the DEPT department or press 1 for DEPT but if you have other suggestions, let me know. I will be collecting these for about a week so please try to get them to me in that time frame. I am hopeful that, with these prompts, it will be possible to make a complete (albeit fairly generic) tree, all with the same voice. Thanks; James alspachfam at charter dot net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cant set H323 up
Hi. I need to set up H323 on an Asterisk box. I've succesfuly compiled the asterisk oh323 (including of course all the dependencies: PWlib and OpenH323), and then compiled asterisk. However, asterisk doesn't report a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP channels, however, the 323 word doesn't appear in the whole output). Is there anything I'm missing? I've read the documentation on the wiki, and none said nothing about editing a config file. I did noticed that they talked about the oh323.conf file, which I dont have. Any help will be great. Thanks in advance, RODOLFO ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: canterburyfortmyers.org returned mail
Wilson Pickett wrote: Why do I get a MAILER DAEMON return for every message I post? Is there something I need to change in my replies? You'd probably be referring to Aster Risk. Mr Risk has been returning messages for quite some time now. Maybe it's been long enough for someone to remove him? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does aanyone have an example of how to dial out with a sip phone on a pstn line?
I am using a card that has an fxo and fxs module. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: canterburyfortmyers.org returned mail
On Sun, 12 Dec 2004 04:36:56 +1300, Matt Riddell [EMAIL PROTECTED] wrote: Wilson Pickett wrote: Mr Risk has been returning messages for quite some time now. Maybe it's been long enough for someone to remove him? Maybe its time for mailing list moderators? (just throwing that out there :)) Leif Madsen. http://www.leifmadsen.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Prompt Info
I have sent this twice now but, I think, for some reason, it has been sent as HTML which is causing it to be drooped (and rightly so). I apologize in advance if, suddenly, those two make it though along with this one. Anyway, I should have been more clear in my original message. I am looking for departments that fit - into - those strings. Pretty much, if a person could replace DEPT with what they are thinking, they are on track. I mention the strings them selves only as a way to show context. When I first posted that message I had a handful of examples that did not fit into that 'mold' but, for the life of me, I can not think of one now. Thanks; James Date: Fri, 10 Dec 2004 16:24:00 -0800 You should not put the press or the number in the prompt. Have them as separate sounds, that way, they are more generic. [EMAIL PROTECTED] wrote: I am looking for titles that fit into the string: press 1 for the DEPT department or press 1 for DEPT but if you have other suggestions, let me know. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail from mysql / change password
Im having a problem where I've just switched from static configs to "realtime" configs stored in mysql It's all working fine (in terms of it reading the configs and loading them as it should), except my problem is that if a user changes there voicemail password via the "Advanced Options (0)" in the Voicemail menu via there SIP phone, the password doesn't get updated in the mysql database (like it used to in the static voicemail.conf file) - and consequently the next time I reload asterisk, there voicemail password gets reset back to whatever it was/is in the mysql database. Am I overlooking something, or is there an easy solution? If I could just disable the change password option in Voicemail, that'd be enough for me (and force them to change it via a web interface). Is that do-able? Here's the line from my extconfig.conf: voicemail = mysql,asterisk,users And the mysql users table schema: CREATETABLEusers( context char(79)DEFAULT''NOTNULL, mailbox char(79)DEFAULT''NOTNULL, passwordchar(79)DEFAULT''NOTNULL, fullnamechar(79)DEFAULT''NOTNULL, emailchar(79) DEFAULT''NOTNULL, pagerchar(79) DEFAULT''NOTNULL, options char(159)DEFAULT''NOTNULL, stamptimestamp, PRIMARYKEY(context,mailbox) ); Thanks Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to setup private enum server ?
Hi, I'd like to setup little private enum server. Any more info on how to do that ? Regards, Rob. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What might be blocking RTP
When I make a call from a SIP phone to a speaking extension on *, such as one that speaks digits or similar, when I monitor * in very verbose mode I can see it running through the routine associated with the extension, but I am getting no RTP data stream back to the phone. Does the machine housing * need a sound card? Does it need OSS or ALSA modules installed? What actually generates the RTP data stream? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Prompt Info
We developed IVR machines for a long time (using Dialogic and our own code) In order to be able to get the most of prerecorded prompts, you need to have a folder general sounds (numbers - 1-20, 30-90, 100-900, 1000-9000 and so on, month names, dept. names, etc.) Then, complete sentences can be built. You'll get the best sounds if you record the whole sentence, but if you need to change anything, you'll need to record again. -Original Message- From: Christopher Dobbs [mailto:[EMAIL PROTECTED] Sent: Saturday, December 11, 2004 2:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voice Prompt Info You should not put the press or the number in the prompt. Have them as separate sounds, that way, they are more generic. EG: Background(press-1-for) Background(sales) Background(and) Background(service) Background(department) Background(press-2-for) Background(Tech) Background(support) -- Christopher Dobbs [EMAIL PROTECTED] wrote: I am trying to put together a list of 'departments' to request as voice prompts. I have the biggies (sales, accounting, shipping, etc...) but I want to make sure I do not miss any. If anyone anyone has some suggestions (Ha... that is like going to an NRA meeting ans asking if anybody has a gun :-) ) please forward them to me (and / or post here although, with the volume of this list I do not always have time to read every digest so the 'and' option may be best.) so that I can compile a single list, verify that they are not already available, group them, and send them on. Please put 'voice prompt' in the subject line of anything you forward me so that I am less likely to miss it. I am looking for titles that fit into the string: press 1 for the DEPT department or press 1 for DEPT but if you have other suggestions, let me know. I will be collecting these for about a week so please try to get them to me in that time frame. I am hopeful that, with these prompts, it will be possible to make a complete (albeit fairly generic) tree, all with the same voice. Thanks; James alspachfam at charter dot net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What might be blocking RTP
On Sun, 2004-12-12 at 03:46, Eric Wieling aka ManxPower wrote: Howard Lowndes wrote: When I make a call from a SIP phone to a speaking extension on *, such as one that speaks digits or similar, when I monitor * in very verbose mode I can see it running through the routine associated with the extension, but I am getting no RTP data stream back to the phone. Does the machine housing * need a sound card? Does it need OSS or ALSA modules installed? What actually generates the RTP data stream? You don't need a soundcard. That's what I thought. Is Asterisk behind NAT? No, this is a local network. If so look at localnet= and externip= in sip.conf and look into portforwarding and rtp.conf. It won't need portforwarding being a local network. I might just check out rtp.conf. Remember AUDIO on SIP/H323/MGCP/SCCP are sent using the RTP protocol. Yes, I am aware of that, and that is what I am not getting back from *. SIP is just a signaling protocol. ...aware of that too. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Will Adtran TSU 600 work with *?
Hi, I am looking at getting adtran tsu 600 p/n 1200.076L2 for my small office It comes with 6 FXS ports and I would use 2 X100Ps for FXO ports. Would that work ? Is there anything I would have to be aware of in such configuration? What would be a better solution? robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Apply Patch for Broadvoice.
On Fri, 2004-12-10 at 20:02, Dealer Backup Admin wrote: Received errors as follows. snip Are you using version 1.0 or CVS HEAD? The patch will probably only apply cleanly on the 1.0 branch. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: canterburyfortmyers.org returned mail
[EMAIL PROTECTED] wrote: Its not a moderator issue, it is a bounce issue, Mailman can be setup to deal with this. However, if this guy bounces messages, just remove him from the list. He's not bouncing them to the list. He (well, his MTA) is bouncing them to the original sender, so the mailing list software never sees the bounce. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 says Protocol Application Invalid?
I have been able to upgrade my Cisco 7905G phones to the SIP Image without any problems, but I just got a 7960, and I can't seem to get to the settings so I can upgrade to a SIP Image. When the phone boots up, it says Configuring VLAN, Configuring IP, TFTP ..., then Protocol Application Invalid. I noticed on the wiki page Firmware issues on 7940 - 7960 http://www.voip-info.org/tiki-index.php?page=Firmware%20issues%20on%207940%2 0-%207960#comments it has some xml script. Can I use that script to fix my phone? If so, how do I go about it? I can't seem to get to the settings to direct it to my TFTP. Any help would be appreciated. Randy MacKay --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 11/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 says Protocol Application Invalid?
Randy MacKay wrote: I have been able to upgrade my Cisco 7905G phones to the SIP Image without any problems, but I just got a 7960, and I can't seem to get to the settings so I can upgrade to a SIP Image. When the phone boots up, it says Configuring VLAN, Configuring IP, TFTP ..., then Protocol Application Invalid. You are using the 7960/7940 SIP image? The 7905 and 7940/7960 firmware are different. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: long list of prefixes
if a phone number starts with one of 50+ prefixes,
i want to send the sip call to gateway X. if it
is in any other prefix, i want to send it to gate
Y.
Take a look at http://www.voip-info.org/wiki-Asterisk+app_dbodbc
too big a hammer. i finally did the agi hack. for the archive
[dial-hawi]
exten = s,1,NoOp(dial-hawi)
exten = _.,1,SetVar(PREFIX=)
exten = _.,2,AGI(agi-prefix|${EXTEN:4:3})
exten = _.,3,NoOp(agi-prefix returns ${PREFIX})
exten = _.,4,Dial(SIP/${PREFIX}${EXTEN:[EMAIL PROTECTED],60,Ttr)
exten = h,1,Hangup()
exten = i,1,GoTo(s,1)
exten = t,1,GoTo(s,1)
with the script being a brutal
#!/usr/local/bin/bash
if ! grep $1 /usr/local/etc/hawi-prefixes /dev/null; then
echo SET VARIABLE PREFIX 1808
fi
randy
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RE: [Asterisk-Users] udev or not?]
Thanks, but there is no zaptel file in /etc/init.d/ I'm using White Box Linux, which is derived from RHEL 3. Kernel is 2.4.x Did you run make config for zaptel? If not do the following; cd /usr/src/zaptel make config - Jose ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 says Protocol Application Invalid?
I had this a couple of days ago .. Randy MacKay wrote: I have been able to upgrade my Cisco 7905G phones to the SIP Image without any problems, but I just got a 7960, and I can't seem to get to the settings so I can upgrade to a SIP Image. When the phone boots up, it says Configuring VLAN, Configuring IP, TFTP ..., then Protocol Application Invalid. When I got this, it was due to the fact that I had incorrect network settings in the SIPDefault.cnf file, and was pointing to the wrong * server, the wrong tftp server etc etc. Also check that the image file specified in the SIPDefault SIPmac address.cnf file is the correct image as well. HTH Julian I noticed on the wiki page Firmware issues on 7940 - 7960 http://www.voip-info.org/tiki-index.php?page=Firmware%20issues%20on%207940%2 0-%207960#comments it has some xml script. Can I use that script to fix my phone? If so, how do I go about it? I can't seem to get to the settings to direct it to my TFTP. Any help would be appreciated. Randy MacKay --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 11/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tormenta PCI - tor2 module not loading
Hello : Have a Tormenta 2 PCI card - Quad E1. When I try to modprobe tor2, the following errors are displayed : /lib/modules/2.4.20-8smp/misc/tor2.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8smp/misc/tor2.o: insmod /lib/modules/2.4.20-8smp/misc/tor2.o failed /lib/modules/2.4.20-8smp/misc/tor2.o: insmod tor2 failed I´d already tried without success : - recompile zaptel - changed PCI slot - changed machine Another facts : - zaptel compiles with no errors - cat /var/log/messages : kernel: Registered Tormenta2 PCI - lspci -bv : Bridge: PLX Technology, Inc.: Unknown device d00d (rev 01) Subsystem: PLX Technology, Inc.: Unknown device 9030 Flags: medium devsel, IRQ 7 Memory at feadfc00 (32-bit, non-prefetchable) I/O ports at d880 Memory at feadf000 (32-bit, non-prefetchable) Memory at feadd800 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 1 Capabilities: [48] #06 [] Capabilities: [4c] Vital Product Data Any clues ? Regards Gustavo Russo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Migrating from CVS HEAD to Stable 1.0.3?
I am sorry to ask such a simple questions. I have been using Asterisk successfully for well over a year now on three servers. I was using CVS HEAD, and the last time I updated was sometime back in July. I decided to switch to the recent stable 1.0.3. I built zaptel, libpri and asterisk, and installed them in that order. All installations reported success. (I stopped asterisk before installing any of them...) When I started up safe_asterisk (and connected to the console), the first error I got was that iaxprov.conf wasn't found. I copied the sample from there to /etc/asterisk and it then got a little further. The last message I see is that it found phone.conf, and then it dies with an Error 1. I am sorry but I don't have the exact error message in front of me, and I had to revert quickly so that my phone would work. When migrating (I don't know if it's downgrading or upgrading) from a July CVS HEAD to 1.0.3, do I need to do anything special, like: 1) Add, change or delete any of my existing conf files in /etc/asterisk, or should they just work? 2) Remove the modules from the old build before doing the install (I assumed that they would just be overwritten, but perhaps that isn't the case...)? 3) Anything else?!? Again, if this appears in any docs, I really apologize, but a quick skim of the doc directory didn't seem to contain a file that seemed to cover this situation... Thanks in advance for any guidance! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] udev or not?]
On Sat, 11 Dec 2004 10:56:53 -0800, Jose Hernandez [EMAIL PROTECTED] wrote: Thanks, but there is no zaptel file in /etc/init.d/ I'm using White Box Linux, which is derived from RHEL 3. Kernel is 2.4.x Did you run make config for zaptel? If not do the following; cd /usr/src/zaptel make config No, I didn't do the make config. Of the many sets of instructions I waded through during the installation, none includes the make config step. Only make clean ; make install. Why do this step? I mean, is it important just for this version of Linux, or only if things don't work as they did in this case, or should it always be part of a compile? OK, I removed the modprobe wcfxo entry from rc.local and did the make config and it seems to work! Thank you. -- Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 and Asterisk...not working....
Sorry if this comes in twice. Wasn't subscribed
first time :-(
Anyone help me here..
It worked once :-(
I have a static IP address which is on my private
network.. Phone is 192.192.192.220 and asterisk server is
192.192.192.22
I have the 7690 with a SIP iamge (Whatever latest
is )
I have 3 lines setup with Free World Dial up and
have the 4th setup to connect to my asterisk server. Here are my config
files..It worked once but now the phone sits there with a 'x' next to it
:-(
;; SIP Configuration for Asterisk;;
Syntax for specifying a SIP device in extensions.conf is; SIP/devicename
where devicename is defined in a section below.;; You may also use ;
SIP/[EMAIL PROTECTED] to call any SIP
user on the Internet; (Don't forget to enable DNS SRV records if you want to
use this); ; If you define a SIP proxy as a peer below, you may
call; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED] ; where the
proxyhostname is defined in a section below ; ; Useful CLI commands to
check peers/users:; sip show peersShow all SIP peers
(including friends); sip show usersShow all SIP
users (including friends); sip show registryShow
status of hosts we register with;; sip
debugShow all SIP messages;
[general]context=home;
Default context for incoming calls
port=5060; UDP Port to bind to
(SIP standard port is 5060)bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)srvlookup=yes; Enable DNS SRV
lookups on outbound calls
;[sip_proxy]; For incoming calls only. Example:
FWD (Free World Dialup);type=user;context=from-fwd
;[sip_proxy-out];type=peer
; we only want to call out, not be
called;secret=guessit;username=yourusername; Authentication
user for outbound proxies;fromuser=yourusername; Many SIP
providers require
this!;host=box.provider.com;;
Test Ext 2201
; extension use - users name -
extension
number;
[2201]type=friendhost=192.192.192.220context=homesecret=xxcallerid="Paul"
2201mailbox=2201dtmfmode=rfc2833nat=no
EXTENSIONS.CONF
writeprotect=no
[globals]PHONES1=SIP/2201PHONES1VM=2201PHONES2=SIP/2202PHONES2VM=2202CONSOLE=Console/dsp;
Console interface for
demo;CONSOLE=Zap/1;CONSOLE=Phone/phone0IAXINFO=guest;
IAXtel
username/password;IAXINFO=myuser:mypassTRUNK=Zap/g2;
Trunk interfaceTRUNKMSD=1; MSD digits to strip
(usually 1 or 0)
[iaxtel700]exten = _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider];switch =
IAX2/user:[EMAIL PROTECTED]/mycontext
[international]; Master context for
international long distanceignorepat = 9include =
longdistanceinclude = trunkint
[longdistance]; Master context for long
distanceignorepat = 9include = localinclude =
trunkld
[local];Master context for local,
toll-free, and iaxtel calls only;ignorepat = 9include =
defaultinclude = parkedcallsinclude = trunklocalinclude
= iaxtel700include = trunktollfreeinclude =
iaxprovider
;This will create a macro we will use in the
dialling plan[macro-vmessage]exten =
s,1,VoiceMail2(u${ARG1})exten =
s,2,Playback(groovy)exten = s,3,Playback(goodbye)exten
= s,4,Hangup
[macro-stdexten];;; Standard extension
macro:; ${ARG1} - Extension (we could have used
${MACRO_EXTEN} here as well; ${ARG2} - Device(s) to
ring;exten = s,1,Dial(${ARG2},20);
Ring the interface, 20 seconds maximumexten =
s,2,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten =
s-NOANSWER,1,Voicemail(u${ARG1}); If unavailable, send to voicemail
w/ unavail announceexten =
s-NOANSWER,2,Goto(default,s,1); If they press #, return to
start
exten =
s-BUSY,1,Voicemail(b${ARG1}); If busy, send to voicemail w/
busy announceexten = s-BUSY,2,Goto(default,s,1);
If they press #, return to start
exten =
_s-.,1,Goto(s-NOANSWER,1); Treat anything else as no
answer
exten =
a,1,VoicemailMain(${ARG1}); If they press *, send the
user into VoicemailMain
;
--; DEFINE EXTENSIONS;
--
[home]; Next, add an extension for
voicemail .; now if we dial 8, we can check
voicemail.;exten = 8,1,VoiceMailMain2exten
= 8,2,Hangup; Add some more extensions for the two lines . now
we'll be able to call one line from the other.; And if no one answers,
it will go to the mailbox for that line.;; Line
1;exten = 2201,1,Dial(${PHONES1},20,Ttm)exten
= 2201,2,Macro(vmessage,${PHONES1VM})exten =
2201,3,Hangup;; Line 2;exten =
2202,1,Dial(${PHONES2},20,Ttm)exten =
2202,2,Macro(vmessage,${PHONES2VM})exten =
2202,3,Hangup;; Line 3;exten =
2203,1,Dial(${PHONES3},20,Ttm)exten =
2203,2,Macro(vmessage,${PHONES3VM})exten =
2203,3,Hangup
;
--; END DEFINE EXTENSIONS;
--
[demo];; We start with what to do when
a call first comes in.;exten = s,1,Wait,1; Wait a
second, just for funexten
RE: [Asterisk-Users] Cisco 7960 says Protocol Application Invalid?
My problem with this phone is I cannot get to the settings to change anything. This is a used phone, but new to me. I have not had it in service yet. None of the buttons on the phone seem to do anything. I assume I have to configure the phone TFTP settings so I can upgrade to the SIP Image and the new configuration settings. When I upgraded my 7905 phones, I entered the TFTP settings from the phone's display, then upgraded thru the TFTP. I assume I do the samething with the 7960, but with the 7960 SIP Image (Yes I do have the 7960 SIP Image, also I did not try and use the 7905 SIP Image). I do have the 7960 SIP Image, but I can't get into the phone to change the TFTP. Is the phone locked? Did the previous owner mess up an upgrade and now the phone is a paper weight? This is my first attempt working with a 7960 phone. Thanks, Randy -Original Message- From: Asterisk [mailto:[EMAIL PROTECTED] Sent: Saturday, December 11, 2004 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 says Protocol Application Invalid? I had this a couple of days ago .. Randy MacKay wrote: I have been able to upgrade my Cisco 7905G phones to the SIP Image without any problems, but I just got a 7960, and I can't seem to get to the settings so I can upgrade to a SIP Image. When the phone boots up, it says Configuring VLAN, Configuring IP, TFTP ..., then Protocol Application Invalid. When I got this, it was due to the fact that I had incorrect network settings in the SIPDefault.cnf file, and was pointing to the wrong * server, the wrong tftp server etc etc. Also check that the image file specified in the SIPDefault SIPmac address.cnf file is the correct image as well. HTH Julian I noticed on the wiki page Firmware issues on 7940 - 7960 http://www.voip-info.org/tiki-index.php?page=Firmware%20issues%20on%207940%2 0-%207960#comments it has some xml script. Can I use that script to fix my phone? If so, how do I go about it? I can't seem to get to the settings to direct it to my TFTP. Any help would be appreciated. Randy MacKay --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 11/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 11/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 11/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to setup private enum server ?
Robert Rozman wrote: I'd like to setup little private enum server. Any more info on how to do that ? You just need bind or any other name server that supports NAPTR records and to setup a zone with NAPTR records... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than myself. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Prompt Info
Your previous messages came through, but had [Asterisk-Users] Re: Asterisk-Users Digest, Vol 5, Issue 158 as the subject. I for one usually skip messages where the person did not think to change the digest subject to something more meaningfully. To help others help you could those of you who get digest form please fix the subject before replying? Thank you in advance. -- Christopher Dobbs [EMAIL PROTECTED] wrote: I have sent this twice now but, I think, for some reason, it has been sent as HTML which is causing it to be drooped (and rightly so). I apologize in advance if, suddenly, those two make it though along with this one. Anyway, I should have been more clear in my original message. I am looking for departments that fit - into - those strings. Pretty much, if a person could replace DEPT with what they are thinking, they are on track. I mention the strings them selves only as a way to show context. When I first posted that message I had a handful of examples that did not fit into that 'mold' but, for the life of me, I can not think of one now. Thanks; James Date: Fri, 10 Dec 2004 16:24:00 -0800 You should not put the press or the number in the prompt. Have them as separate sounds, that way, they are more generic. [EMAIL PROTECTED] wrote: I am looking for titles that fit into the string: press 1 for the DEPT department or press 1 for DEPT but if you have other suggestions, let me know. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] looking for input on broadband router with QoS and VPN support
Hi, We're installing an * box next week (pbxtra from fonality) and I'm trying to come up with a solution for remote users that want a phone in their home. I need VPN and QoS capability, wireless support would be a nice to have. Ethernet handoff is fine, i don't need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces (some using vpn, many using qos), but not a whole lot of anything on folks that are using voip over qos over vpn So far i've come up with this list: Linksys WRT54G(S) running Sveasoft Linksys WRV54G (stock) Draytek 2900 m0n0wall on a Soekris 4801 I do want a small fanless box b/c this will be the primary access router for the remote user (including when they are just at home surfing). I like the soekris because i can have a 'work' interface and a 'home' interface. The draytek seems to have everything i want out of the box, but there are some concerns over the quality of the product and the support level. The linksys wrt+sveasoft is attractive from a cost perspective, but i'm not sure of the ability for the sveasoft firmware to handle everything i'm after...(and i need to cough up $20 just to ask the question on their board). The wrv54g seems to get a lot of complaints, and it's not clear if there is any traffic shaping support. Am i missing anything? I'm sending this through m0n0wall on a PC now, i've got a wrt54g i'm going to test, and will probably place an order for the Soekris board next week. I can't get any response from the US reseller of Draytek, so i'm not sure i'm willing to put any money on that one yet. Thanks so much for any input!!! Bob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Voice Prompt Info
One more thing about prompts, it's better to say for sales press 5 than press 5 for sales, because by the time you hear sales you've already forgotten what number it was. So record for sales press and the digits (you could use the digits that come with *, but a sentence in two voices sounds very funny, I know, the user directory on an old IVR of ours works that way). That way when you need to change the numbers the menu you can do it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background Music via telephone speaker.
To everybody on this wonderful group. I am considering to use an asterisk PBX with 70 telephones and I would like to play background music trough the speakers of the telephones. When the hook is lifted, the background music stops till the phone is on hook again. Is this possible? Or can it easily be programmed. Is it not similar with the ring tone? Thank you all, Willy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Voice Prompt Info
Warren Burstein wrote: One more thing about prompts, it's better to say for sales press 5 than press 5 for sales, because by the time you hear sales you've already forgotten what number it was. If you add the sounds all you need is For Sales recorded the new sounds have press # already. So you don't need to get any additional recorded items except the one that says For Sales by Allison. If you want have her record Press as an additional recorded item. So record for sales press and the digits (you could use the digits that come with *, but a sentence in two voices sounds very funny, I know, the user directory on an old IVR of ours works that way). That way when you need to change the numbers the menu you can do it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best option for FSX: IAXy or TDM400P or Voip phone?
Personally I find the ATA adapters to be the most versatile, your mileage may vary though. When you need more extensions you just buy more ATA's, no need to tear up the * box or take it down etc. Buying IP phones is OK but you are limited to IP Phones only. With the ATA's you can buy ANY phone at the local store etc.. Just my opinion of course :) -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Humberto Aicardi [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Saturday, December 11, 2004 4:50 PM Subject: [Asterisk-Users] Best option for FSX: IAXy or TDM400P or Voip phone? Hi, I currently have a * server with a IAXy adapter and a Voip phone. My doubt is: which is the best option? I personally find IAXy to be very effective, except from the fact that they don't support G729. The other option would be to use the TDM400P, which I have heard that it has some problems with echo, is this true? And finally to use a VOIP phone which look good and includes several extra features. Oops, I forgot there's still the gateway option, including ATA186, VoicePlanet, Mediatrix and so on. The problem is that they are expensive compared to prior options, except the VOIP phone. What I really need is a solution that works without the usual * echo problems. The major issue with IAXy is the price at US$99. I can buy for US$ 75 a Grandstream BT102. Can anyone share their experience with the above solutions? Thanks in advance, Humberto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ACK from asterisk not matched to transaction by SER / LCS2005
For reasons unknown to me, SER and subsequently a Microsoft Live Communcations Server 2005 seems to have problems, matching a SIP ACK request from asterisk to the ongoing SIP transaction, I have attached the complete log, but the essential lines are: 13(2894) DEBUG: RFC3261 transaction matching failed13(2894) DEBUG: t_lookup_request: no transaction found13(2894) SER: forwarding ACK statelessly The result is a "half duplex" connection that will break down, as soon as the timeout for the missing ACK package is reached. Has anybody an idea how to fix this problem ? The problem only occurs when if a call originates in asterisk. Calls from the LCS system to asterisk work just fine. chris. 9(2890) SIP Request:9(2890) method: INVITE9(2890) uri: sip:[EMAIL PROTECTED]9(2890) version: SIP/2.09(2890) parse_headers: flags=19(2890) Found param type 232, branch = z9hG4bK61c24316; state=169(2890) end of header reached, state=59(2890) parse_headers: Via found, flags=19(2890) parse_headers: this is the first via9(2890) After parse_msg...9(2890) preparing to run routing scripts...9(2890) DEBUG : is_maxfwd_present: searching for max_forwards header9(2890) parse_headers: flags=1289(2890) end of header reached, state=99(2890) DEBUG: get_hdr_field: To [28]; uri=[sip:[EMAIL PROTECTED]9(2890) DEBUG: to body [sip:[EMAIL PROTECTED]]9(2890) get_hdr_field: cseq CSeq: 102 INVITE9(2890) DEBUG: get_hdr_body : content_length=3649(2890) found end of header9(2890) DEBUG: is_maxfwd_present: max_forwards header not found!9(2890) DEBUG: add_param: tag=as47998c2b9(2890) end of header reached, state=299(2890) parse_headers: flags=2569(2890) find_first_route(): No Route headers found9(2890) loose_route(): There is no Route HF9(2890) parse_headers: flags=20489(2890) check_via_address(192.168.4.39, 192.168.4.39, 0)9(2890) Sending:INVITE sip:[EMAIL PROTECTED] SIP/2.0Max-Forwards: 10Record-Route: sip:[EMAIL PROTECTED];transport=tcp;r2=on;ftag=as47998c2b;lrRecord-Route: sip:[EMAIL PROTECTED];r2=on;ftag=as47998c2b;lrVia: SIP/2.0/TCP 192.168.4.39;branch=0Via: SIP/2.0/UDP 192.168.4.39:5082;branch=z9hG4bK61c24316From: "10" sip:[EMAIL PROTECTED];tag=as47998c2bTo: sip:[EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]:5082Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEUser-Agent: Babble/0.6.10Date: Fri, 10 Dec 2004 16:58:02 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContent-Type: application/sdpContent-Length: 364 v=0o=root 2442 2442 IN IP4 192.168.4.39s=sessionc=IN IP4 192.168.4.39t=0 0m=audio 30016 RTP/AVP 8 0 97 3 2 110a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:97 iLBC/8000a=rtpmap:3 GSM/8000a=rtpmap:2 G726-32/8000a=rtpmap:110 speex/8000a=silenceSupp:off - - - -m=video 3 RTP/AVP 34 31a=rtpmap:34 H263/9a=rtpmap:31 H261/9.9(2890) orig. len=841, new_len=1043, proto=29(2890) tcp_send: no open tcp connection found, opening new one9(2890) tcpconn_new: new tcp connection: 192.168.4.379(2890) tcpconn_new: on port 5060, type 29(2890) tcp_send: sending...9(2890) tcp_send: after write: c= 0xf51740e0 n=1043 fd=159(2890) tcp_send: buf=INVITE sip:[EMAIL PROTECTED] SIP/2.0Max-Forwards: 10Record-Route: sip:[EMAIL PROTECTED];transport=tcp;r2=on;ftag=as47998c2b;lrRecord-Route: sip:[EMAIL PROTECTED];r2=on;ftag=as47998c2b;lrVia: SIP/2.0/TCP 192.168.4.39;branch=0Via: SIP/2.0/UDP 192.168.4.39:5082;branch=z9hG4bK61c24316From: "10" sip:[EMAIL PROTECTED];tag=as47998c2bTo: sip:[EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]:5082Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEUser-Agent: Babble/0.6.10Date: Fri, 10 Dec 2004 16:58:02 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContent-Type: application/sdpContent-Length: 364 v=0o=root 2442 2442 IN IP4 192.168.4.39s=sessionc=IN IP4 192.168.4.39t=0 0m=audio 30016 RTP/AVP 8 0 97 3 2 110a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:97 iLBC/8000a=rtpmap:3 GSM/8000a=rtpmap:2 G726-32/8000a=rtpmap:110 speex/8000a=silenceSupp:off - - - -m=video 3 RTP/AVP 34 31a=rtpmap:34 H263/9a=rtpmap:31 H261/9 9(2890) DEBUG:destroy_avp_list: destroing list (nil)9(2890) receive_msg: cleaning up27(2908) tcp_main_loop: read response= f51740e0, 2 from 9 (2890)27(2908) tcpconn_add: hashes: 772, 127(2908) tcp_main_loop: data available on 0xf51740e0 [h:772] 4127(2908) send2child: to tcp child 0 19(2900), 0xf51740e019(2900) received n=4 con=0xf51740e0, fd=2619(2900) tcp_read_req: content-length= 019(2900) SIP Reply (status):19(2900) version: SIP/2.019(2900) status: 10019(2900) reason: Trying19(2900) parse_headers: flags=119(2900) Found param type 232, branch = 0; state=619(2900) Found param type 237, ms-received-port = 42320; state=619(2900) Found param type 237, ms-received-cid = 2200; state=1619(2900) end of header reached, state=519(2900) parse_headers: Via found, flags=119(2900) parse_headers: this is the first via19(2900) After parse_msg...19(2900) forward_reply: found module tm, passing reply to it19(2900) DEBUG: t_check: msg
[Asterisk-Users] RE: Voice Prompt Info
First of all, I generally skip messages that have the entire digest subject as well. I am always thinking that it was somebody who has left the entire digest in their reply. Sorry I missed the subject in my messages. Second, currently the plan is to have Allison (the same person who recorded the rest of the prompts) record just the title of the department. I think, we already have everything else needed to put together the prompts our selves. I have found that I get fairly decent results editing the files together to make my prompts. I know that I can just have them played one after another but editing them together gives me a bit more control of the timing. If anyone has any objections to recording the files this way, please let me know. It just seems that it is the most flexible. It allows you to say For accounting press one , The accounting department is closed today, You have reached the accounting department, etc... Thanks; James Subject: Re: [Asterisk-Users] Voice Prompt Info From: Christopher Dobbs [EMAIL PROTECTED] Date: Sat, 11 Dec 2004 13:07:07 -0800 Your previous messages came through, but had [Asterisk-Users] Re: Asterisk-Users Digest, Vol 5, Issue 158 as the subject. I for one usually skip messages where the person did not think to change the digest subject to something more meaningfully. To help others help you could those of you who get digest form please fix the subject before replying? Thank you in advance. -- Christopher Dobbs [EMAIL PROTECTED] wrote: Subject: [Asterisk-Users] RE: Voice Prompt Info From: Warren Burstein [EMAIL PROTECTED] Date: Sat, 11 Dec 2004 23:41:38 +0200 One more thing about prompts, it's better to say for sales press 5 than press 5 for sales, because by the time you hear sales you've already forgotten what number it was. So record for sales press and the digits (you could use the digits that come with *, but a sentence in two voices sounds very funny, I know, the user directory on an old IVR of ours works that way). That way when you need to change the numbers the menu you can do it. Subject: Re: [Asterisk-Users] RE: Voice Prompt Info Date: Sun, 12 Dec 2004 17:57:49 -0500 Warren Burstein wrote: One more thing about prompts, it's better to say for sales press 5 than press 5 for sales, because by the time you hear sales you've already forgotten what number it was. If you add the sounds all you need is For Sales recorded the new sounds have press # already. So you don't need to get any additional recorded items except the one that says For Sales by Allison. If you want have her record Press as an additional recorded item. So record for sales press and the digits (you could use the digits that come with *, but a sentence in two voices sounds very funny, I know, the user directory on an old IVR of ours works that way). That way when you need to change the numbers the menu you can do it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recording voicemail and messages
What is the secret to getting * to record messages or voicemail. It goes thru the process but the file created is zero length. I don't think it can be a perm problem as I am running * as root - maybe not a good idea but I am only testing at this stage. Do I need a sound card in the * box? - I wouldn't have thought so. What * modules are needed? chan_oss.so? chan_alsa.so? ?? What device - /dev/dsp? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: -lssl
Hi Clive, if you have openssl already installed, make sure that you also have libssl-dev installed (development package for ssl, I don't know it is is called in distributions other than Debian). You probably have already found your solution, but I am answering this so it appears in the list archive and others can find the answer later. See also: http://lists.digium.com/pipermail/asterisk-users/2002-April/002160.html This fixes it. On Tue, 2004-11-23 at 12:04 +, Clive Carter wrote: Hi Having my first go at compiling Asterisk from cvs source. Compiled and installed zaptel ok Running make asterisk returns the following error message /usr/bin/ld cannot find -lssl collect2: ld returned 1 exit status The last part of the compile messages on screen are- editline/libedit.a db1.ast/libdb1.a stdtime/libtime.a -ldl -lncurses -lm -lresolv -lssl There is obviously something I have not installed, but what ? Have searched archives and thro package descriptions and come up with nothing Any help appreciated ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-Re: [Asterisk-Users] help with detecting fax.---fixed.
Original Message
From: Ariel Batista
I have Spandsp working fine. Asterisk sees a fax on the zap port and
redirects the call to the fax-in area.
This works if I have a simple dialing rules that goes answers first
and waits 10 secs then goes to the next item. If it hears a fax it
goes to the right place. Here is a sample that works. Also you need
to have the fax lines in the context that all calls from the pstn
side come in. In my case it was inbound.
[incoming]
exten = 2019,1,Goto(test,s,1)
[test]
exten = s,1,answer
exten = s,2,wait(5)
exten = s,3,Macro(stdexten,Sip/101)
exten = fax,1,Goto(fax-in,s,1)
[fax-in]
I am adding the notes as what we did to fix this problem in case
someone else needs these fixes. Note this problem is due to what I
feel is a bug in asterisk. But there is a work around. Here you need
to add an extra line which is not really used. Asterisk will work if
you add this to it.
exten = s,1,NoOp
exten = s,2,Dial(zap/1)
Instead of
exten = s,1,Dial(Zap/1)
We now actually have it going to spandsp now and it's working from
the macro.
But if I use this following macro it just detects the fax then goes
to congestion instead of the fax extenstion. [macro-followme]
;
; Standard single line follow me then to voicemail
; $ARG1 first device to dial to. $ARG2 2nd device to dial.
;
exten = s,1,Answer
exten = s,2,Wait(5)
exten = s,3,Dial(${ARG1},20)
exten = s,4,Dial(Zap/g1/${ARG2})
exten = s,5,Voicemail(u${MACRO_EXTEN})
exten = s,104,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?105:107)
exten = s,105,Dial(Zap/g1/${ARG2})
exten = s,106,Voicemail(b${MACRO_EXTEN})
exten = s,107,Voicemail(u${MACRO_EXTEN})
;
I am calling it with this setup.
exten = 2019,1,Macro(followme,Sip/101,16502468900)
I have fixed the problem. See notes below.
___
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RE: [Asterisk-Users] Cisco 7960 says Protocol Application Invalid?
My problem with this phone is I cannot get to the settings to change anything. This is a used phone, but new to me. I have not had it in service yet. None of the buttons on the phone seem to do anything. I assume I have to configure the phone TFTP settings so I can upgrade to the SIP Image and the new configuration settings. When I upgraded my 7905 phones, I entered the TFTP settings from the phone's display, then upgraded thru the TFTP. I assume I do the samething with the 7960, but with the 7960 SIP Image (Yes I do have the 7960 SIP Image, also I did not try and use the 7905 SIP Image). I do have the 7960 SIP Image, but I can't get into the phone to change the TFTP. Is the phone locked? Did the previous owner mess up an upgrade and now the phone is a paper weight? This is my first attempt working with a 7960 phone. Might try setting up an dhcp server. If the 7960 happens to be configured to use dhcp (and if memory serves correctly, that is the default), then your dhcp server can dole out a tftp address (type 66). If you have a packet sniffer (eg, Ethereal), take a look to see what (if anything) the phone is doing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 20 BT-100 setup - what firmware is recomended ?
Hi, I'd like to setup 20 BT-100 with Asterisk. If I got all discussion on grandstreams right, I should put my own tftp server and point phones to it. On phones is 1.0.5.16 firmware. Is this one good or should I up(down) grade to certain version ? What functionality is possible with BT-100 ? Which are user's favourite ? Regards, Rob. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 says Protocol Application Invalid?
I do have the 7960 SIP Image, but I can't get into the phone to change the TFTP. Is the phone locked? Did the previous owner mess up an upgrade and now the phone is a paper weight? First yes you can unlock the settings on the phone, the default password is cisco or if you have older software it is **# settings. If you don't have a dhcp server available you can manually set the tftp server. I have this done on several customer sites that did not want a dhcp server. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modprobe wcfxo causes fc3 box to crash
Hello everybody, After going through multiple posts on Internet and trying different things I can't seems get zaptel wcfxo loaded on the server correctly. I run dual P3 500 Mhz box. After reading README.udev I put a file 60-zaptel.rules /etc/udev/rules.d # Section for zaptel device KERNEL=zapctl, NAME=zap/ctl KERNEL=zaptimer, NAME=zap/timer KERNEL=zapchannel, NAME=zap/channel KERNEL=zappseudo, NAME=zap/pseudo KERNEL=zap[0-9]*, NAME=zap/%n In /etc/udev/permissions.d/50-udev.permissions I added: # zaptel devices zap/ctl:root:root:0664 zap/timer:root:root:0664 zap/channel:root:root:0664 zap/pseudo:root:root:0664 zap/*:root:root:0644 I then ran make linux26, make install, make config. I replaced insmod and rmmod in /etc/init.d/zaptel with modprobe and modprobe -r according to suggestions made in groups. When I run modprobe wcfxo I get following: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for wcfxo I see zaptel and wcfxo is loaded if I run lsmod though. In 1 minute my box crashes Rebooting the box and running it again, same thing. /var/log/messages before crash Dec 10 17:24:30 conference kernel: Zapata Telephony Interface Registered on major 196 Dec 10 18:36:03 conference kernel: wcfxo: DAA mode is 'FCC' Dec 10 18:36:03 conference kernel: Uhhuh. NMI received for unknown reason 15 on CPU 0. Dec 10 18:36:03 conference kernel: Dazed and confused, but trying to continue Dec 10 18:36:03 conference kernel: Do you have a strange power saving mode enabled? Dec 10 18:36:03 conference kernel: Found a Wildcard FXO: Generic Clone Dec 10 17:24:33 conference udev[4636]: configured rule in '/etc/udev/rules.d/60-zaptel.rules' at line 4 applied, ' zapchannel' becomes 'zap/channel' Dec 10 17:24:33 conference udev[4635]: configured rule in '/etc/udev/rules.d/60-zaptel.rules' at line 3 applied, ' zaptimer' becomes 'zap/timer' Dec 10 17:24:33 conference udev[4636]: creating device node '/dev/zap/channel' Dec 10 17:24:33 conference udev[4635]: creating device node '/dev/zap/timer' Dec 10 17:24:33 conference udev[4649]: configured rule in '/etc/udev/rules.d/60-zaptel.rules' at line 5 applied, ' zappseudo' becomes 'zap/pseudo' Dec 10 17:24:33 conference udev[4649]: creating device node '/dev/zap/pseudo' Dec 10 17:24:33 conference udev[4650]: configured rule in '/etc/udev/rules.d/60-zaptel.rules' at line 2 applied, ' zapctl' becomes 'zap/ctl' Dec 10 17:24:33 conference udev[4650]: creating device node '/dev/zap/ctl' Dec 10 17:24:33 conference udev[4665]: configured rule in '/etc/udev/rules.d/60-zaptel.rules' at line 6 applied, ' zap1' becomes 'zap/%n' Dec 10 17:24:33 conference udev[4665]: creating device node '/dev/zap/1' After each reboot /dev/zap disappears due to udev dynamic nature I guess. I tried to build zaptel commenting out ifeq ($(DYNFS),) else @echo Dynamic filesystem detected -- not creating device nodes @echo If you are running udev, read README.udev endif I see the devices created and can load wcfxpro but server crushes within less than a minute nevertheless Has anyone have an idea? I am ready to give up on FC3 although I wonder of the hardware has anything to do with my problem. Thanks in Advance, Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy: no dial tone
Hi List, I have this good looking IAXy device... I have managed to provision it, i can see it registering to my asterisk box, however when I pick up the phone which is plugged in the IAXy I have no dialtone, nothing. Any ideas what might be going on? Cheers, Jean-Michel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXPeers for Windows Beta released
Dave Cotton wrote: http://www.sineapps.com/down/IAXPeers.zip Could you please have a look and let me know your thoughts. First I like it. I can use it straight away. Cool, that's good to hear! :-) Only comment at the moment is would it be possible to save the configuration of the Host and the order of the peers for the next start up? Ok, done (you will need to download it from the above URL). The logic works as such: 1. When you click the connect button, it will save your hostname to the registry (and will load it when the app starts). 2. If you change one of the items in the dropdown box, it will now start up next time with that same entry allocated (assuming the number of IAX peers you have does not change) 3. I have fixed a little bug with regard to the last bar not going green once the connection returned from above 333ms. Let me know if this is ok for you. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to test enum?
Michael Vogel wrote:
I dumped some sample enum configs to http://www.asterisk.net.au/tutorial/7/
and more on e164.org
Are there any test numbers where I can see if ENUM lookup is working?
18005558355 (1800 555 tell, news and weather service etc)
And is it possible as well to test if a number of a SIP or IAX provider
exists?
Just do a if goto call...
exten = s,2,GotoIf($[$[${ENUM:0:3} = SIP] | $[${ENUM:0:3} = IAX]] ? 3 : 52)
That traps both, but obviously can be altered to trap one or the other
and handle them seperately...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
I do not try to dance better than anyone else.
I only try to dance better than myself.
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Re: [Asterisk-Users] RealTime and Macro question?
I found the problem ! In the for appdata I have - dialnumber_wvm,1004,SIP/1004 it must be - dialnumber_wvm|1004|SIP/1004 Damian Minkov wrote: Is it possible to call a macro, which is defined in extensions.conf from a realtime extension configured in Mysql. Beacuse when i try i receive an error - no such context. -- Executing Macro(SIP/1007-2165, dialnumber_wvm,1004,SIP/1004) Dec 11 12:51:04 WARNING[22551]: app_macro.c:100 macro_exec: No such context 'macro-dialnumber_wvm,1004,SIP/1004' for macro 'dialnumber_wvm,1004,SIP/1004' Here is what i have in extensions table : id context extenpriority app appdata _ 1sip-internal 10041Macro dialnumber_wvm,1004,SIP/1004 -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New PRI with DID in US?
Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the incoming calls for the 20 undefined numbers to a common resource (ivr, operator, or canned message) without having to define each one? You should be able to us a pattern to match. By placing the catch-all pattern in an included context it will only match if none of the actual extensions match. http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting Excellent, thanks Peter. Exactly what I need. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: canterburyfortmyers.org returned mail
Why do I get a MAILER DAEMON return for every message I post? Is there something I need to change in my replies? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P / Brazilian Telco Problem. (Urgent)
This is the second time that i configure a * box using a E100P card. The only difference at this time is that i´m using another Telco and the box have more one card ( Wildcard - 2 FXS + 2 FXO ). Well , everything looks fine i don´t have any kind of alarms on my zttool , the board gives me a green signal. BUT the span is Down :( snip voip*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active The Down indicates the d-channel is not active. Likely because your service provider has it turned down due to alarms on their end, etc. Might also verify with your provider that channel 16 is in fact the d-channel. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New PRI with DID in US?
On December 10, 2004 06:26 pm, Rich Adamson wrote:
Assuming I have 100 DID numbers but only define 50 of those in
extensions.conf, is there an easy way to send the incoming calls
for the 20 undefined numbers to a common resource (ivr, operator,
or canned message) without having to define each one?
I have 30 (Bell sends us 7 digits) -- I define the ones I actually use first,
then add this below -- this is hte last entry in that context for me:
; example of a DID I'm actually using:
exten = 2922022,1,Dial(IAX2/[EMAIL PROTECTED],,g)
exten = 2922022,n,Macro(handle-hangup)
; catch-all that matches anything I have not explicitly matched above:
exten = _29220XX,1,Wait(1)
exten = _29220XX,n,Playback(vm-num-i-have)
exten = _29220XX,n,SayDigits(${EXTEN})
exten = _29220XX,n,Playback(vm-goodbye)
exten = _29220XX,n,Hangup
Piece of cake.
-A.
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Re: [Asterisk-Users] polycom phone IP 500/600 conference feature
I think he means you need multiple lines on the polycom phone in order to use it's conference. On Fri, 10 Dec 2004 15:38:35 -1000, Richard [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Friday, December 10, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] polycom phone IP 500/600 conference feature Richard wrote: The phone and * are all on public ip address. All other features and calls are fine except the conference feature. I am using g711 ulaw for all calls. The problem only applies to conference feature on the polycom phone. We could not get the conference feature to work if we had the Polycom lines set to the same username. Once we set each registration (for each line) to a different username it started working. I am not sure what you mean. We only have one line and one registration on the phone. Btw, just want to be clear, this is the conference feature on polycom phone, not on *. Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cant set H323 up
Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa: Hi. I need to set up H323 on an Asterisk box. I've succesfuly compiled the asterisk oh323 (including of course all the dependencies: PWlib and OpenH323), and then compiled asterisk. However, asterisk doesn't report a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP channels, however, the 323 word doesn't appear in the whole output). Is there anything I'm missing? I've read the documentation on the wiki, and none said nothing about editing a config file. I did noticed that they talked about the oh323.conf file, which I dont have. Which OS? :) How do you did it ?? I'm sitting on this problem second day and nothing. :(. Slackware Current and Mandrake 10.1 Offcial. Would you be so nice and explain how you did it? versions etc. :) Kind Regards, Corvin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cant set H323 up
hello are you sure that you have loaded the module in the modules.conf files? load = chan_oh323.so K. - Original Message - From: Rodolfo Grave [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, December 11, 2004 3:32 PM Subject: [Asterisk-Users] Cant set H323 up Hi. I need to set up H323 on an Asterisk box. I've succesfuly compiled the asterisk oh323 (including of course all the dependencies: PWlib and OpenH323), and then compiled asterisk. However, asterisk doesn't report a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP channels, however, the 323 word doesn't appear in the whole output). Is there anything I'm missing? I've read the documentation on the wiki, and none said nothing about editing a config file. I did noticed that they talked about the oh323.conf file, which I dont have. Any help will be great. Thanks in advance, RODOLFO ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What might be blocking RTP
Howard Lowndes wrote: When I make a call from a SIP phone to a speaking extension on *, such as one that speaks digits or similar, when I monitor * in very verbose mode I can see it running through the routine associated with the extension, but I am getting no RTP data stream back to the phone. Does the machine housing * need a sound card? Does it need OSS or ALSA modules installed? What actually generates the RTP data stream? You don't need a soundcard. Is Asterisk behind NAT? If so look at localnet= and externip= in sip.conf and look into portforwarding and rtp.conf. Remember AUDIO on SIP/H323/MGCP/SCCP are sent using the RTP protocol. SIP is just a signaling protocol. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cant set H323 up
Hi , I have this schenario: SIPUAs---SER---asterisk(sip-h323)GNUGK(RadiusBilling)-h323Clients | | GNUGK(ProxyMode)-H323-LD_Provider It works perfect for me (Linux RH9), you just have to be carefull with versions, you need: - PWLIB : pwlib_1.5.2.tar.gz - OpenH323 : openh323_1.12.2.tar.gz - Inaccessnetworks-asterisk-oh323 : asterisk-oh323-0.7.0.tar.gz (see readme file, there is a note about a patch) - Asterisk 1.0+ - GNUGK (from www.gnugk.org) versions 2.2+ or 2.0.9, radius billing for sip-to-h323 calls does not work properly with older versions, I am using CVS Head version... ( cvs -d :pserver:[EMAIL PROTECTED]:/cvsroot/openh323gk checkout openh323gk ) send us more details about your versions. Rafael Risco Millicom Peru SA On Sat, 11 Dec 2004 16:49:12 +, Corvin [EMAIL PROTECTED] wrote: Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa: Hi. I need to set up H323 on an Asterisk box. I've succesfuly compiled the asterisk oh323 (including of course all the dependencies: PWlib and OpenH323), and then compiled asterisk. However, asterisk doesn't report a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP channels, however, the 323 word doesn't appear in the whole output). Is there anything I'm missing? I've read the documentation on the wiki, and none said nothing about editing a config file. I did noticed that they talked about the oh323.conf file, which I dont have. Which OS? :) How do you did it ?? I'm sitting on this problem second day and nothing. :(. Slackware Current and Mandrake 10.1 Offcial. Would you be so nice and explain how you did it? versions etc. :) Kind Regards, Corvin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- rrgv ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 636 Area Code Asterisk Compatible DIDs
Anybody know of good reliable Asterisk compatible DIDs in the 636 (Missouri, USA) area code? Voicepulse doesn't go there, and Broadvoice seems unreliable in my Asterisk installation -- so I'm reluctant to recommend it. Thanks, /edg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Handling raw audio (8000 signed 16bit big-endian)
format_sln.c is what you want. It should compile on 0.9.0 but WHY are you using such an old version? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jim O'Brien Sent: Saturday, December 11, 2004 6:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Handling raw audio (8000 signed 16bit big- endian) Does anyone know if there is a format-raw.c routine available for Asterisk-0.9.0? Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: canterburyfortmyers.org returned mail
Its not a moderator issue, it is a bounce issue, Mailman can be setup to deal with this. However, if this guy bounces messages, just remove him from the list. On Sat, 11 Dec 2004, Leif Madsen wrote: On Sun, 12 Dec 2004 04:36:56 +1300, Matt Riddell [EMAIL PROTECTED] wrote: Wilson Pickett wrote: Mr Risk has been returning messages for quite some time now. Maybe it's been long enough for someone to remove him? Maybe its time for mailing list moderators? (just throwing that out there :)) Leif Madsen. http://www.leifmadsen.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] looking for input on broadband router with QoS andVPN support
I have three remote offices with VPN's into my main office. Two Offices use the IAXy's with ATT 958 Phones (Functional and inexpensive $30, does not use the vpn, but I have a port opened to it). The IAXy's are easy to set up (no NAT to worry about), low bandwidth with the IAX. The third office uses a Sipura SPA-2000 (w/ ATT 958 Phones) which works over the VPN. I don't have Qos, and it seems to work pretty good, except my home office, I have a dsl connection that when I downloading large files, it sometime effects the quality of the call. HTH Randy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Rich Sent: Saturday, December 11, 2004 1:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] looking for input on broadband router with QoS andVPN support Hi, We're installing an * box next week (pbxtra from fonality) and I'm trying to come up with a solution for remote users that want a phone in their home. I need VPN and QoS capability, wireless support would be a nice to have. Ethernet handoff is fine, i don't need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces (some using vpn, many using qos), but not a whole lot of anything on folks that are using voip over qos over vpn So far i've come up with this list: Linksys WRT54G(S) running Sveasoft Linksys WRV54G (stock) Draytek 2900 m0n0wall on a Soekris 4801 I do want a small fanless box b/c this will be the primary access router for the remote user (including when they are just at home surfing). I like the soekris because i can have a 'work' interface and a 'home' interface. The draytek seems to have everything i want out of the box, but there are some concerns over the quality of the product and the support level. The linksys wrt+sveasoft is attractive from a cost perspective, but i'm not sure of the ability for the sveasoft firmware to handle everything i'm after...(and i need to cough up $20 just to ask the question on their board). The wrv54g seems to get a lot of complaints, and it's not clear if there is any traffic shaping support. Am i missing anything? I'm sending this through m0n0wall on a PC now, i've got a wrt54g i'm going to test, and will probably place an order for the Soekris board next week. I can't get any response from the US reseller of Draytek, so i'm not sure i'm willing to put any money on that one yet. Thanks so much for any input!!! Bob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 11/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 11/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-2000 NAT Problems
I had a Grandstream 286 at my home hitting my Asterisk box at the office, all worked well and I received phone calls fine until the device just up and died. I replaced this unit with an SPA-2000 because I have been impressed with the Sipura devices and decided to use them for most of my needs in the future. Problem is that my phone attached to the device rings shortly after power up of the device but seems to lose it's head after a period of time and stops ringing until I power cycle the unit or reboot it. My Asterisk config is the same regarding NAT for this extension and I have the Sipura registering with * so I am at a loss as to why Asterisk loses or stops ringing this device. I have dug around and can't seem to solve this issue so far, any help would be appreciated. -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Soyo G668 IP Phone
Hi All, Asterisk newbie here. Have recently got my system working with Xten softphones and now want to expand to real ones. Was considering Grandstream but am concerned about possible RFI problems with them (after reading these archives for the past few weeks), because I plan to deploy using a wireless network. Came across this Soyo G668 phone today. Has anyone tried it yet with Asterisk? If so does it work? http://phone.soyo.com/doc/G668%20spec.pdf Thanks in advance, chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best option for FSX: IAXy or TDM400P or Voip phone?
Hi, I currently have a * server with a IAXy adapter and a Voip phone. My doubt is: which is the best option? I personally find IAXy to be very effective, except from the fact that they don't support G729. The other option would be to use the TDM400P, which I have heard that it has some problems with echo, is this true? And finally to use a VOIP phone which look good and includes several extra features. Oops, I forgot there's still the gateway option, including ATA186, VoicePlanet, Mediatrix and so on. The problem is that they are expensive compared to prior options, except the VOIP phone. What I really need is a solution that works without the usual * echo problems. The major issue with IAXy is the price at US$99. I can buy for US$ 75 a Grandstream BT102. Can anyone share their experience with the above solutions? Thanks in advance, Humberto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] does aanyone have an example of how to dial outwith a sip phone on a pstn line?
Charles S. Antrim wrote:
I am using a card that has an fxo
and fxs module.
I am no where near
an expert but I have my sip phone working through my pstn line and this is my
config.
/etc/asterisk/sip.conf
[general]port =
5060bindaddr = 192.168.69.1context = sipdisallow = gsmallow =
alawdisallow =
ulawnat=disablesrvlookup=nolocalnet=192.168.69.0/255.255.255.0subscribecontext
= sip
[snom-james]type=friendsecret=passwordhost=dynamiccallerid="James
Bean"
690defaultip=192.168.69.250dtmfmode=rfc2833mailbox=690
[bt-karen]type=friendsecret=passwordhost=dynamiccallerid="Karen
Colomb"
691defaultip=192.168.69.251dtmfmode=infomailbox=691
/etc/asterisk/extensions.conf
[pstn]
exten =
s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a comment in the CLI
for info.exten = s,2,SetMusicOnHold(random)exten =
s,3,Dial(SIP/snom-jamesSIP/bt-karen,45,t) exten =
s,4,Hangup;exten = s,5,VoiceMail(u100) ;Whatever box
you want.
[internal]
exten =
i,1,Playback(invalid)exten = i,2,Hangupexten =
t,1,Hangup
exten =
099,1,Echo ;simple echo test when you dial 099 on your
phone
include =
outgoinginclude = sip
[outgoing]
exten =
_9X.,1,Dial(Zap/g1/${EXTEN:1})exten = _9X.,2,Congestion()exten =
_9X.,3,Hangup
include =
sip
[sip]
exten =
690,1,SetMusicOnHold(random)exten =
690,2,Dial(SIP/snom-james,30,tr)exten = 690,3,voicemail2,u690exten
= 690,102,voicemail2,b690
exten =
691,1,SetMusicOnHold(random)exten =
691,2,Dial(SIP/bt-karen,30,tr)exten = 691,3,voicemail2,u691exten
= 691,102,voicemail,b691
include =
internalinclude = outgoing
[from-sip]
include =
internal
This isn't the best example of how to do it but it
works.
I hope it helps.
James
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Re: [Asterisk-Users] SCRIPT: Fax Remvoal Please Call: 1-800...
--On Thursday, December 09, 2004 19:19 -0700 Joseph [EMAIL PROTECTED] wrote: On Thu, 2004-12-09 at 18:11 -0800, Lee Howard wrote: On 2004.12.09 17:56 Joseph wrote: At time to time I receive some junk faxes from some advertising companies that play smart and don't provide any TSI number so I can not bock them by the number in Hylafax. Do they not provide Caller*ID either? No they don't provide any caller ID, if they did they would be on my junk_fax_list long time ago. I think it is illegal to send faxes with-out any identifier like caller ID. Though I don't know who to complain to about it. It's illegal to send junk faxes though, PERIOD. If you didn't request the fax in the first place they can, and will face steep fines/penalties at the hands of the FCC, if you report them. So report them, include any information you can, and cooperate with the FCC if they want to continue gathering more evidence. Same thing with telemarketers. The FCC is actually pretty good about finding and hurting these sleezebags, contrary to popular government images, they do get stuff done. SPAM is another matter, but junk faxes and telemarketers have well established procedures for being dealt with. Despite calling their Fax Removal Service 1-800-... number several time they refuse to obey my request. Not that I particularly want to advocate litigiousness, but filing a complaint with FCC will get their attention very quickly, believe me. http://www.fcc.gov/cgb/consumerfacts/unwantedfaxes.html Thank you for the link, will save it for future reference and use it for sure. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with TDM400P and cidstart=polarity
I'm testing a TDM400P with FXO module to receive incoming calls from an analogue line and send it to a SIP device. To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity. The problem is that when a call is finished, the TDM400P seems to require about 20 seconds to prepare for the next incoming call. If a new call comes in within 20 seconds after the previous call was hungup, the TDM400P answers with a modem carrier, sounding like you're calling a modem pool..! The caller hangs up and retries the call, and the next time everything is OK. If the second call comes in later than 20 seconds after the previous call was finished, or if I remove the cidstart=polarity (and don't get callerid) everything works fine. I can't see any difference in the Asterisk debug logs worth mentioning... Has anyone experienced anything similar..? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail from mysql / change password
Answering my own question - after a few more hours googling, the way to prevent users changing there voicemail password via the voicemal "advanced options (0) menu", is described at: http://bugs.digium.com/bug_view_page.php?bug_id=0002386 All thats required is to preceed voicemail pins with a "-" character. When they try to change the password, they are just played "No". Now If they want to change there voicemail password I can force them to do it via a web interface, which will ensure the mysql db remains consistent with the Voicemail app, including after a reload! Seems to beincluded as of CVS HEAD 09-06-04 Brad - Original Message - From: Brad Hughes To: [EMAIL PROTECTED] Sent: Sunday, December 12, 2004 3:13 AM Subject: [Asterisk-Users] voicemail from mysql / change password Im having a problem where I've just switched from static configs to "realtime" configs stored in mysql It's all working fine (in terms of it reading the configs and loading them as it should), except my problem is that if a user changes there voicemail password via the "Advanced Options (0)" in the Voicemail menu via there SIP phone, the password doesn't get updated in the mysql database (like it used to in the static voicemail.conf file) - and consequently the next time I reload asterisk, there voicemail password gets reset back to whatever it was/is in the mysql database. Am I overlooking something, or is there an easy solution? If I could just disable the change password option in Voicemail, that'd be enough for me (and force them to change it via a web interface). Is that do-able? Here's the line from my extconfig.conf: voicemail = mysql,asterisk,users And the mysql users table schema: CREATETABLEusers( context char(79)DEFAULT''NOTNULL, mailbox char(79)DEFAULT''NOTNULL, passwordchar(79)DEFAULT''NOTNULL, fullnamechar(79)DEFAULT''NOTNULL, emailchar(79) DEFAULT''NOTNULL, pagerchar(79) DEFAULT''NOTNULL, options char(159)DEFAULT''NOTNULL, stamptimestamp, PRIMARYKEY(context,mailbox) ); Thanks Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't capture -1 return on Dial command
How can I capture a -1 result on a Dial command? Basically, I have the
following setup, and I want to be able to process the audio file after the
outbound call has been done regardless how how it ends.
No matter how the call ends, I can't get macro-record-stop to run.
Any help would be great.
-Eric
from extensions.conf
[macro-dialanalog]
exten = s,1,Macro(record-start)
exten = s,2,Dial(${TRUNK}/${MACRO_EXTEN},70)
exten = s,3,Macro(record-stop)
exten = s,102,Macro(record-stop)
exten = h,1,Macro(record-stop)
[macro-record-start]
exten = s,1,SetVar(CFN=${MONITORPATH}/${TIMESTAMP}--${CALLERIDNUM})
exten = s,2,Monitor(${MONITORFILETYPE},${CFN})
[macro-record-stop]
exten = s,1,System(/usr/bin/soxmix ${CFN}-in.${CFE} ${CFN}-out.WAV
${CFN}.WAV)
exten = s,2,System(rm ${CFN}-in.WAV ${CFN}-out.WAV)
exten = s,3,System(/usr/bin/sox ${CFN}.WAV -t uw - | /usr/bin/lame -r
-m m -s 8 - ${CFN}.mp3)
exten = s,4,System(rm ${CFN}.WAV)
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[Asterisk-Users] Many similar contexts - can I use Macro or some other template concept ?
Hi, I'd like to make small 20 users setup with BTs. I'd like each of them to have its own context (for recording prompts, conference, ...). For them to have same extensions I should put them in separate contexts and let BT call them offhook. But these contexts are pretty similar (for instance dial to conference on 5 goes to different conf. number for each user, ...) How could I describe those contexts with some sort of template (macro probably cannot do that - but as newbie I could be wrong...) ? Are there any other ways of context templates filled with data in dialplan ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PoE VOIP phones in Australia
On Fri, 2004-12-10 at 18:31, James Andrewartha wrote: Hi, Are there any resellers of phones that can take power over ethernet in Australia? All I can find for sale online is the BT-10[12], which is cheap but not featureful enough, and the Snom 190, which is about right, but neither of them support PoE. I'm particularly intereseted in the Snom 220 with the keypad expansion for our receptionist. Although, could you make a PoE split-out cable for the Snom 190? See the polycom IP 300/500/600 phones. There are many resellers of these phones in Australia. Note the 300/500 require an additional cable for PoE. Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Variable-length dialing with a Quicknet Inetnet PhoneJACK card
Hi, all. On a lark, I have gotten Asterisk 1.0.2 (from Debian testing) up and running, but I have found one problem when I try to use it with Free World Dialup: Dialing doesn't work properly. I have everything else working; it receives calls, ringing my phone and everything, but when I try to dial using the configs found on FWD's site for Asterisk at this URL: http://www.fwd.pulver.com/advanced/iax When I use a real telephone on a QuickNet card, it takes anything after the 393, and only one digit of it. I confirmed it by replacing the . with XXX so I can try dialing the three-digit number to test. It works just fine, but then, I'm only limited to dialing three-digit numbers with FWD. What happens I hear ringing after the 393-6 in 393-612 which I would dial to get the time. And the following appears in the messages log file: Dec 11 18:26:18 WARNING[131080]: Call rejected by 65.39.205.121: No such context/extension That's because it's trying to dial 6 and it hadn't even taken the rest of my dialing yet. Near as I can tell, the pattern matching on . only seems to work as advertised when you have a phone which sends a dial packet, rather than taking indidividual DTMFs from a real telephone because you don't have an idea of when the stream of digits will actually end or you can somehow tell Asterisk to look for something, like a hash. Anyone have any advice on this one? I'm at wit's end. --Ian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
