RE: [Asterisk-Users] So what if I can't dial out ... or in ... Asteriskjust blows my mind!
From: Lane
Sent: Wednesday, December 29, 2004 6:00 PM
I subscribed to this list for about two months before I began posting, so
I've
got a buttload of email to sift through ... I'm doing this BEFORE I flood
the
list with my inane questions ...
But here goes:
I read a reply from one guy to another about recording. The message
included
this context from extensions.conf:
[recordings]
exten = 500,1,Festival('Please record your message')
exten = 500,2,Record(mymessage:gsm)
exten = 500,3,Festival('You said')
exten = 500,4,Playback(mymessage)
exten = 500,5,Festival('Press 1 to continue or 2 to change your message')
exten = 500,6,ResponseTimeout(3)
So I figgered out how to make selected conferences automatically record
the
minutes, and I was SO PSYCHED
But then I thought, what if I answer the phone and it's my ex-wife
claiming
that she's gonna sue me for malfeasance because of my new boyfriend, and I
wanna make a recording of that call? How could I discreetly begin a
recording of that call?
Thanks,
lane
P.S. I don't gotta ex-wife, I'm just saying what if?
Check out twisted's patch on the bug tracker
http://bugs.digium.com/bug_view_page.php?bug_id=0002955 which allows you to
press a configurable dtmf key while on a call to begin recording.
If ya wannta see it in CVS test it out and post a note :)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recording/Monitoring a call mid-stream?
Yup...Check out twisted's patch http://bugs.digium.com/bug_view_page.php?bug_id=0002955 . It does almost exactly what you're looking for. Don't forget to reply to the bug if the patch works for you -Brian On Wed, 29 Dec 2004, Paul Rodan wrote: Is there a way to monitor a call mid-stream? I did look on the Wiki and found that AstGUI can do it, but it's a bit of an overkill. What I want is for a customer service rep, sitting in front of a Cisco 7960, to be able to hit a button (either on their phone, or maybe a specific webpage) that will start recording the call from that point on. I'm thinking the services button on the Cisco could be rigged to send the proper command to the manager interface, to start recording the call. But I don't know how to write such a program. I'm hoping something already exists. Anybody? There was something on the bug tacker that did exactly this - it allowed ne key to start/stp ecrding the current call. You should be able to find the patch there fairly easily. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Woes continue
Thnx! -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Final call for departments
Accounts by itself would be useful. -Original Message- From: David Boyd [mailto:[EMAIL PROTECTED] Sent: 30 December 2004 00:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Final call for departments HOw about : development Dave On Wed, 2004-12-29 at 04:51, Alspach Family wrote: I am getting ready to submit a list of department names to be recorded. This is what I have so far: Accounting Accounts payable Accounts receivable Administration Billing Collections Complaint Customer Service Engineering Facilities Help desk Human Resources Information Technology Inside Sales Investor Relations Legal Mail room Marketing Printing Projects Public Relations Purchasing Receiving Sales Sales Floor Shipping Shop Support Systems Technical Support Travel If any one has additional suggestions, please e-mail them to me ([EMAIL PROTECTED] or [EMAIL PROTECTED]). I am fairly sure that none of the above exist (I was only able to search through the WIKI list, so if there are other prompts in the CVS that are not listed there, I do not know about them.) If I have made a dupe, please let me know so that I can remove it. I was fairly certain that 'Operator' was already available but I was unable to find it by its self. Thanks for your help. I plan on sending these off on Friday the 31st so please try to get them to me by then. Thanks; James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Woes continue
On December 29, 2004 22:35 pm, Andrew Kohlsmith wrote: Well it is hard to go back to a specific configuration since I have used the system to test the rpm packages I compile. Yikes. Yep. But there is only one way to know for sure that a new package is working. I have had much success in 2004. So much that I figured all the warnings on the list to not upgrade a working system were for the ultra paranoid. Still think that is mostly true. Nothing like using a production server for testing, eh? I have reverted to a (actually several) pre 1.0 release that worked well, changed the port, moved the PCI slot, changed out the motherboard three times, enabled and disabled onboard devices, tried several kernels, rerun the cabling from the smart jack, checked the powersupply voltages, UPS, power cabling, etc etc etc. Basic troubleshooting? yeah man. That wasn't meant to be flip -- Perhaps I've just been bitten too many times myself by doing the exact same thing you just did -- I back up my config (going as far as to rsync or image the partition if I need) before changing something like that on a production system... especially something as important as our main telephone system. :-) Understood. I want to believe that each day will only bring improvements to the code. Sure, I know that bugs can slip in to the updates but that should be temporary if it happens at all... right? hahahahaha! Perhaps my doctor is right and I am crazy. I dont have a T100P lying around so I cant do much in the way of changing the interface. Yet. Before I commit to changing that I want to rule out any other possibilities... How can one determine without a shadow of a doubt that it is the card or otherwise? I have enabled all the debugging I can find BUT the output is foriegn to me... shrug Yeah -- I don't know -- I am the last to blame hardware (10 years as an embedded electronics designer does that to you) but failing everything else it really does seem that this is the issue, does it not? Yes it does, but I still want to think it is me. After all, I have spent more time compiling RPMS than I have learning the dialplan. With that said my production dialplan is the most basic you can get. I have an alternate, more complicated dialplan that I am developing but I only switch that in for brief periods testing during low traffic hours. My plan is to move the complicated part of the dialplan to a secondary server that will handle the the real work - VoIP calls, AVR, Voicemail, etc. over TDMoE or IAX2... still thinking that one out. Something else I learned the hard way -- have any criticial hardware available onhand, not at a distributor, even if they can ship overnight -- I have a story about a DS3 MUX that had both controllers die and the manufacturer shipped one overnight but UPS lost it... true story. It's expensive to have hardware sitting on the shelf idle but better that than be without phone service or whatever other critical system you've got. :-) yep I need another T400p, a spare engine for my truck and not one, but two legal age females on hand in case my wife gets the flu, or worse. Of course I have a 7206VXR here sitting on a shelf with a lot of pretty cards stuffed in it just waiting for a chance to prove itself... but thats a different story altogether. Is there a way to log all communication on the D Channel? Have I missed some critical debugging reference? I'm going crosseyed looking, tweaking and trying the same things over again. pri debug span 1 will show you all q.931 traffic and intense will show you the q.921 traffic too, but this seems deeper than that -- I am not a telco expert but it certainly seems like something very low level is buggered. I am sorry I can't be more help. Well you tried and I commend you for that. I wouldn't even be asking here if I had even the slightest clue on what to do next. Perhaps Digium will come through with some testing tomorow / err / I mean today. I have to do whatever it takes to regain 99.999% reliability for my dial-up customers. I would like to accomplish this with Asterisk, as a proof of concept if for nothing else, but I will be forced to pull it out of the chain if a resolution can't be found in the next couple days. This has gone on t long and I am looking bad. oops. Gotta get some sleep. Thanks for you comments! Regards, -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Final call for departments
Since Friday is the last day I can accept new requests fro this run, I wanted to post to the list what I have as of about 1:30am Pacific Time 30 Dec. This way people have Thursday to make any additions / suggestions and then Friday, I will send what I have on. The list is getting longer so don't forget to donate what you can, via PayPal, to robf at geekthing dot com Thanks; James PS I attached the Open Office sxc file since the PDF is too big. If you can not read this and want the PDF, contact me off list ([EMAIL PROTECTED]) and I will send it to you. asterisk departments.sxc Description: OpenOffice Calc spreadsheet ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp-0.0.2pre6
Hallo Thomas Niesel On Wed, 29 Dec 2004 22:03:05 +0100 you wrote: Hi Folks, hi Steve I get following error on loading app_rx/txfax.so: ...WARNING[10458]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: symbol errno, version GLIBC_2.0 not defined in file libc.so.6 with link time reference Unable to load app_rxfax.so ...cut Answer myself: use the wiki, open your eyes! Its all in there! -- Tho/\/\as ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hook/Flash, Hold, Call Waiting, Three Way Calling
For threeway calling (analog phone) I just hit the flash button get a dial tone, dial the number and hit the flash key again. It doesn't work for me when I'm using asterisk. No problems without it. So is my hardware broken or my dialplan? When you hit the flash key is anything displayed in the CLI ? A while back, someone posted a list of built-in extension numbers that are built into the zap channel module. The list included: *0 Send hook flash *67 Disable Caller ID *69 Say last caller's Caller ID *70 Disable call waiting *72 Activate call forward immediate *73 Deactivate *78 Enable Do Not Disturb *79 Disable Do Not Disturb *80 Add last caller's caller ID to blacklist *82 Enable Caller ID (only if disabled with *67) I don't use the above, but they certainly appear to be the ones your looking for. Obviously some of the features noted in that list do not exist in asterisk, therefore it would suggest they apply to the pstn/zap interface . That same posting indicated the above extensions could be overrode with other entries in extensions.conf. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] automatic startup
I've been thinking about taking steps to make my * server more reliable. In particular I'd like to have it automatically start * after a power loss. Can anyone here provide some guidance as to how to accomplish this. Keep in mind that I have a TDM400p that needs a couple of modprobe commands before I can start * itself. Did you do a 'make config' in the zaptel source, etc? That should have installed the startup scripts. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Digium TDM04B
I have installed Digium TDM04B with the latest CVS. However I have encountered following problems: 1. When it dials out, many times the digits are not properly recognized by telco as I hear the announcement please check the number and dial again although I see on the screen that the dialed number is correct. 2. When the call is forwarded outside, with something like exten = 22,1,Dial(SIP/22,18,rtT) exten = 22,2,Dial(Zap/g1/7038988235,18) exten = 22,3,Voicemail(22) Most of the time, I get an answer when the call is forwarded on the PSTN line so that Voicemail line never kicks in. Any suggestions will be highly appreciated. Try inserting a 'w' in the Dial command. Something like: exten = 22,2,Dial(Zap/g1/w7038988235,18) Had the same problem with an older central office and the 'w' fixed it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Doubts about the Monitoring command
Hi all, I have some doubts concerning the way asterisk records calls using the Monitor command. I ´ve done some jitter and packet loss tests in a such way that, from asterisk 1, I send a file to asterisk 2 and record this file in asterisk 2 using the Monitor command. To simulate the jitter and packet loss, I use the Cloud software, so with that one, I can control the jitter and packet loss to any value I want, and simulate the network characteristics I could have in a real network. Setting the cloud with 25 % of packet loss only, without jitter, I ´ve got the file recorded, in asterisk 2, with a kind of acceleration, ie, this file plays a bit faster than the original file sent from asterisk 1 to asterisk 2. Hearing the sent file with a handset, without recording, I listen a deteriorated file different from the recorded one. My question is: Is asterisk able to detect the packet loss and modify the file recorded in a such way that compensate this packet loss? How can I get the file recorded as I hear in the handset,ie , with the deteriorated audio? Thanks and best regards __ Do you Yahoo!? Send a seasonal email greeting and help others. Do good. http://celebrity.mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk and Capi
Bruno Hertz is believed to have said: Hi Aldo don't know about Suse, but I have a working setup with asterisk 1-0 stable, chan_capi 0.3.5 and fcpci-suse9.1-3.11-02 on Debian Sarge, though not prepackaged but all self compiled. Looking at your log messages, chan_capi obviously is installed, but the load of app_capiCD.so fails due to an undefined symbol capidebug. [...] So either your modules.conf is messed up, or there's a problem with the chan_capi package itself, which you should then report to Suse. But take a look at your modules.conf. I myself have autoload enabled, and all works automagically. Maybe you have it disabled, and the module load order is affected by this ? Also, you can check if you like wether the symbol is actually defined in chan_capi.so: # nm chan_capi.so | grep capidebug 00010720 B capidebug If you see a 'U' instead of a 'B' there, your chan_capi package is messy. Regards, Bruno. Bruno, thanks for your hints! They were precious as I could easily trace the state of my box. The drivers were OK, but I was loading other ISDN drivers on the one hand and missing the config file for chan_capi. Quite a messy situation. So I looked into the wiki and the example file from the chan_capi driver and came up with a neutral config file. Now my box happily reports to see the Fritz! card, with the two B channels sitting idle (I am away from the office and just did send the updated config files to the unconnected box to see if they were the right approach..). So thanks again for the hints (and have a fine new year) Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. Again, welcome to the Asterisk.org Open Source PBX Project! Meet you on the IRC channel :-) ...and a Happy New Asterisk-year! /oej ** Asterisk version information At this moment we have two current versions of Asterisk, the developer version and the stable version. The stable version is distributed as .tar.gz archives on several servers. The current stable version of Asterisk is 1.0.3. The stable version contains no new functions and only changes when bugs are fixed. The development version is to be used by people that can test new functions and live with bugs and unexpected shortcomings. ** The mailing list is growing Today, we propably have over 10,000 readers on the -users list. This means that everything anyone write to this mailing list, is sent to thousands of mailboxes that are already flowing over with messages. That's why we all need to follow some simple rules on how to use the mailing list and the other tools that are available. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. Do not use this list as a secondary support line if you do not get an answer on the -users list. It is meant for developer discussions, not advanced support. If you need answers, there is a better chance that you will get help on the irc channel. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services (asterisk-biz). You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. If you are unsure which list to use, send only to the -users list. Make sure that you remove unnecessary text when you reply, to make it easy to browse the mailing list quickly. And please do not send HTML mail to a mailing list. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate reports slow things down for everyone, so please spend a few minutes searching first. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base. This is how Asterisk grows, free contributions and consultants that are paid to add functionality on a case by case basis. ** Be a community member - contribute! The Asterisk software growth is very much
Re: [Asterisk-Users] SMS - how to send one
In extensions.conf
[smsdial]
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup
[local]
exten = 07,1,wait(1)
exten = 07,2,Answer
exten = 07,3,GotoIf($[foo${CALLERIDNUM} = foo]?12:4)
exten = 07,4,GotoIf($[${CALLERIDNUM:0:10} = 8005875290]?9:5) //this is
the number sms text messages come from
exten = 07,5,system(play /var/lib/asterisk/sounds/ring3.wav -v3 )
exten = 07,6,Playback(welcome)
exten = 07,7,musiconhold
exten = 07,8,Hangup
exten = 07,9,SMS(${EXTEN:3},a)
exten = 07,10,System(/usr/lib/asterisk/smsin ${EXTEN:3})
exten = 07,11,Hangup
exten = 07,12,system(play /var/lib/asterisk/sounds/uh-uhhh.wav -v1 )
exten = 07,13,Wait(1)
exten = 07,14,Playback(withheld)
exten = 07,15,Hangup
I have a PHP program to send the messages
$timeout = 7500;
$socket = fsockopen(10.0.0.99,5038, $errno, $errstr,
$timeout);
if ($socket)
{
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: manageruser\r\n); // must be defined
in manager.conf
fputs($socket, Secret: mysecretpassword\r\n\r\n);
fputs($socket, Action: Originate\r\n);
fputs($socket, callerid: . $your_text_message .
$your_sending_number\r\n); // your sending number
fputs($socket, exten: . $mobile_number . \r\n);
fputs($socket, Channel: Zap/g1/147017094009\r\n); //this is
the bt message center
fputs($socket, Context: smsdial\r\n);
fputs($socket, Priority: 1\r\n\r\n);
}
incoming messages go into /var/spool/asterisk/sms/sc-me.777
i had to register with BT first by sending a blank message to telephone
number 0
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 19, 2004 7:41 PM
Subject: [Asterisk-Users] SMS - how to send one
I've read quite a bit in the older mailing list posts and the wiki but
I'm missing some simple point.
1) What is required to send an SMS to a mobile outside the office given:
Channel: ZAP/1
send it to $SMS_RECIPIENT (which includes the final extra digit)
via
$SMS_CENTER=the national message center server for sending messages
$MESSAGE= the message text
How is the .call file organized?
2) When an SMS is received from $SMS_CENTER2, how to get the $MESSAGE from
it?
using
exten = s/${SMS_CENTER2},NoOp(${CALLERID})
exten = wait, answer
then?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial with no phone line connected
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in zap
group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial succeeded
even though there is neither line voltage nor dial tone. Can at least the
lack of voltage be detected? It would be good in case one of the phone
wires fell out that it would just move on to the next outgoing line.
Yes, the chip set on the TDM card does provide flags for indicating no
voltage (disconnected), low voltage (something is off hook), and normal
pstn voltage (on-hook).
About three months ago, Mark added code that detected when a pstn line
was unavailable (eg, rj11 disconnected, damaged cable, someone disconnected
the wrong pstn line). The code created a problem for someone (I don't remember
the details), and he changed the code to be a compile-time config option.
I don't have any past references to that other then from memory. Maybe
someone that can read code can find that option for you.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSLink modem freeze
Hi Eric, Thanks every body that answered about this problem. About change de default SIP port (5060), I tried it at first and the UAC could authenticate but when I made a call and another side pick the phone up DSLink 200E freeze again. ie. there wasn't any port 5060 on transactions. I will have this DSL modem on my LAB asap and I will give feedback to the list. Thanks Eric Wieling aka ManxPower escreveu: On Cisco routers you can do something like no nat sip fixup 5060 and that will disable only the special SIP related nat features, but leave in all of the other NAT features. If a vendor does not include a similar ability in their SIP aware router they should be shot. --Eric C F wrote: I have this problem with Best Data DSL Modems, If I disable NAT (on the router, not in SIP) it works fine. You might be able to do the same just disable NAT and it will work, if you disable NAT then you will have to get a different router to be able to share the same IP, and if you use PPPoE you might not be able to do it, in which case you will have to get a different DSL modem. On Wed, 29 Dec 2004 20:00:28 -0600, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Rodrigo P. Telles wrote: Hi Folks, I've been having troubles with a DSL router (DSLink 200E) and SIP phones. When I put any SIP phone (software or hardware) to work behind that DSL router, it completely freeze. I ready tech specs of that DSL router and it says that SIP protocol is supported. ie. I tested two DSLink 200E with the same results. Turn off SIP support and let the generic NAT deal with it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel.it TDKOM Group ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy reliability issues
Paul Fielding wrote: I've just picked up a pair of IAXy devices. They work fine except that they keep going offline. As in, I plug it in, it connects to Asterisk, I can dial and phone and all is dandy. Then, maybe 12h later, maybe 24, maybe 36, maybe 48, I'll either try to phone the device and not get through or I'll pick it up and the dialtone is gone. it's simply lost it's connection to Asterisk. If I unplug and plug back in, it reconnects and all is well. I'm running firmware v. 22. Anyone else experiencing this? Paul Paul, I have 30 of them sitting in a box that I can sell until these problems get resolved! Want mine? Your best bet is to get SER runing with the NAT proxy and use SIP ATA endpoints. Best, Todd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoDSL without using IAD
Hi *, Would anyone know about solutions that let you use a VoDSL connection without using and IAD? VoDSL is starting to come from many vendors now in The Netherlands and it seems silly to have an IAD that turns VoDSL into POTS/ISDN to connect it to a card in the Asterisk box that turns it into e.g. G.711. Afaik VoDSL is ultimately A-law in ATM cells and G.711 A-law in IP packets. Kind regards, Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Doubts about the Monitoring command
On Thu, 30 Dec 2004, Guild Jackson wrote: Hearing the sent file with a handset, without recording, I listen a deteriorated file different from the recorded one. My question is: Is asterisk able to detect the packet loss and modify the file recorded in a such way that compensate this packet loss? How can I get the file recorded as I hear in the handset,ie , with the deteriorated audio? Hi, Asterisk just dumps the arriving audio into the Monitor file as it comes - IE missing packets just disappear. This accounts for the speedp and the different distortion. Steve Kann has been working on a generic jitter-buffer with packet-loss-concealment. Once that's in Asterisk it will facilitate changing this Monitor behaviour so it records the reconstructed stream rather than the raw frames. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Digium TDM04B
1. When it dials out, many times the digits are not properly recognized by telco as I hear the announcement please check the number and dial again although I see on the screen that the dialed number is correct. Had the same problem with an older central office and the 'w' fixed it. I can also confirm that a w fixed our problem as well. We had used two X1000P's in this office without the 'w' but after the upgrade to a 3port FXO TDM card, we had to place a w for one of the lines. The other line worked fine, but for safe measure we left the w in place. - B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy reliability issues
On Thu, 30 Dec 2004, Gary wrote: On Thu, 30 Dec 2004 00:12:51 -0700, Paul Fielding wrote: I've just picked up a pair of IAXy devices. They work fine except that they keep going offline. As in, I plug it in, it connects to Asterisk, I can dial and phone and all is dandy. Then, maybe 12h later, maybe 24, maybe 36, maybe 48, I'll either try to phone the device and not get through or I'll pick it up and the dialtone is gone. it's simply lost it's connection to Asterisk. If I unplug and plug back in, it reconnects and all is well. I'm running firmware v. 22. Anyone else experiencing this? Paul DHCP timeouts ?? Didn't somebody say that the IAXy doesn't renew its DHCP lease (ie its a BOOTP client). In which case, your DHCP server needs to give it an infinite lease. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] verbose setting changed?
Up until last night, I could run: asterisk -vvvr as root to connect to a running * session and have the verbosity set to 3. Last night, however, I updated to CVS-v1-0-12/29/04-16:47:20 and the behavior is different. Now the -v flags don't seem to make a difference, I have to issue: set verbose 3 to change verbosity. Is that a planned change? One nice thing is that I only have to issue that one time on a running session is seems and the verbosity is remembered. However, my nightly asterisk -rx restart gracefully resets the verbosity back to 0. Is there a settings file that I can set verbosity in? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What happened with the 'reinvitation' on SIP?
Of course I try canreinvite=yes - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 6:46 PM Subject: Re: [Asterisk-Users] What happened with the 'reinvitation' on SIP? Did you try canreinvite=yes? -Matthew - Original Message - From: Megan Willigs [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 4:31 PM Subject: [Asterisk-Users] What happened with the 'reinvitation' on SIP? Hi everybody in new versions of Asterisk the RTP on SIP pass only througt the Asterisk, not directly between the endpoints like olders versions. What happened whit this feature? (reinvite) Can you help me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is asterisk that unstable ????
from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Final call for departments
On Wed, 29 Dec 2004 01:51:16 -0800, Alspach Family [EMAIL PROTECTED] wrote: I am getting ready to submit a list of department names to be recorded. This is what I have so far: QA or Quality Assurance. -Chuji ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Diax version 0.9.9f
Hi all, Diax version 0.9.9f is ready to be tested by the interested people. You can download it for the moment from the following location only: http://www.geocities.com/tdanro/diax/diax099f.zip Please do not use older config files with 0.9.9f !!! You have some command line options now for diax.exe: /d- start with debug mode enabled /u - start with ATCOM USB phone support enabled (keyboard, ring. etc.) There is a html example file to show you how to launch a Diax call from a web page using a link like: diax://102/danpbx You must register first the application from the Config Menu. I still have some work to do on the updated help file and web page which will be ready ASAP. If you need help with all the new features in 0.9.9f please send me a mail directly. They are many internal changes from the previous version and I have not enough free time to througly test the app. I need your help for that. What's new in 0.9.9f (they are a lot of other small changes which can be observed playing with the app): - ulaw, Speex, GSM and iLBC codec support with auto-negotiation capabilities - display currently negociated codec (flash if is different that the preferred one) - fully support for the ATCOM AU-100 USB phone (http://www.atcom.com.cn/engweb/bUSBPhone.html) - web browser integration (start app and/or dial using a link like diax://number/alias) - configurable audio latency - user defined rings based on midi files (like for the GSM phones) with ring volume adjustment - include all the iaxclient library updates till today - save forms positions between restarts - home automation support (start applications/scripts, send X10 commands, Infrared to come), based on CallerID - IP address for CallMe function changed to the actual one - thai language support - configurable keyboard support (USB phone keyboard) - midi file as ringin signal (polyphonic) - you can launch DIAX with command line switches (/d for debug mode, /u for USB phone (Atcom) support) - can start application without an audio device installed (for Home Automation purpose), even from a Terminal Server session - clicking on DELETE for more than 2s delete all from the display - the application is minimized if clicking on 'X' in the right up corner - launch DIAX without an audio device - send X10 commands based on CallerID - prevent phonebook entries without any name - better handling LEFT/RIGHT keys from some SonyEricsson T610 Bluetooth phones - better display format for BT phones, based on currently bSelected (in the phone) text size - extended debug info in debug mode solved bugs: - if reducing the number of registration servers, the deleted one goes to red even not defined. - call volume - no counter incrementing - audio configuration with different sound device for playback and ring - Missing MSSTDFMT.DLL in WinXP SP2 and some Win98 systems - no need to close the application in order to save the debug log file Thank you all and a Happy New Year! Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Greg - Cirelle Enterprises wrote: from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? I'm runing Asterisk for a year now as the IPBX of our little consulting firm. It stopped working only 4 times: two of these where power failures and the other two turned out to be Telco company problems (dead line). We have 2 PSTN lines (using Digium X101P cards), 5 intrernal VoIP extentions (Grandstream budgettone - one of which is located on another continent, using a Wifi connection to a near by village that hosts an ADSL router... don't ask) and 2 VoIP termination/origination lines. Of course, your mileage may very, but at least here there is no nightly restart script. Hope that helps you in any way. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with 2 E100P cards behind an Alcatel 440
Hello,
I'm actually trying
to connect an asterisk PBX with 2 E100P card to an alcatel 440, but I'm facing
some problems.
In fact, i had one
E100P connected to the public PSTN and the other one connected to the
Alcatel.
I can receive call
from the PSTN without any problems but I can't place call from my Alcatel to the
PSTN.
Here is my conf
files:
zaptel.conf:
span=1,0,0,ccs,hdb3bchan=1-15,17-31dchan=16loadzone=fr
span=2,1,0,ccs,hdb3bchan=32-46,48-62dchan=47loadzone=fr
zapata.conf:
[channels]
context=pri-publicswitchtype=euroisdnpridialplan=localusecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesgroup=1musiconhold=defaultsignalling
= pri_cpechannel = 1-15channel = 17-31
context=pri-alcatelgroup=2overlapdial=yessignalling =
pri_netchannel = 32-46channel = 48-62
Extensions.conf:
[pri-public]
; Nous sommes a paris
;include =
default
exten =
s,1,DigitTimeout(1)exten =
_X.,1,Dial(ZAP/g2/${EXTEN})
; Si la ligne ADSL est tombe envoi du numero vers la carte quadBRI1exten
=
_0X.,1,Dial(ZAP/g1/${EXTEN})
; Si la ligne ADSL est tombe envoi du numero vers la carte
quadBRI1
exten =
8xxx,1,Dial(ZAP/g2/${EXTEN})exten =
2xxx,1,Dial(ZAP/g2/${EXTEN})
exten =
8249,1,Dial(SIP/[EMAIL PROTECTED])exten
= 0149718249,1,Dial(SIP/[EMAIL PROTECTED])
exten =
149718249,1,Dial(SIP/[EMAIL PROTECTED])
[pri-alcatel]
;include =
default
; Nous sommes a parisexten = s,1,DigitTimeout(1)exten =
_X.,1,Dial(ZAP/g1/${EXTEN})
; Envoi du numero vers la carte quadBRI2
Here is my debug
span error:
ConnecteurAzennUlis*CLI Protocol Discriminator: Q.931
(8) len=48 Call Ref: len= 2 (reference 4346/0x10FA)
(Originator) Message type: SETUP (5) [a1] Sending
Complete (len= 1) [04 03 80 90 a3] Bearer Capability (len= 5) [
Ext: 1 Q.931 Std: 0 Info transfer capability: Speech
(0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35) [9e]
Non-Locking Shift (len=01): Requested codeset 6 [24 01 80]Dec 30
16:31:22 WARNING[229390]: chan_zap.c:6806 zt_pri_error: PRI: !! Unknown IE
1572 (len = 3) [6c 0b 21 81 31 34 39 37 31 38 33 34 32] Calling
Number (len=13) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163)
(1)
Presentation: Presentation permitted, user number passed network screening (1)
'149718342' ] [70 0b 80 30 36 37 32 30 38 33 35 31 36] Called
Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown
Number Plan (0) '0672083516' ] [7d 02 91 81] IE: High-layer
Compatibility (len = 4) [7e 01 04] User-User Information (len=
3) [ 04 ]-- Making new call for cr 4346-- Processing Q.931 Call
Setup-- Processing IE 161 (cs0, Sending Complete)-- Processing IE 4
(cs0, Bearer Capability)-- Processing IE 36 (cs6, Unknown Information
Element)!! Unknown IE 36 (cs6, Unknown Information Element)-- Processing
IE 108 (cs0, Calling Party Number)-- Processing IE 112 (cs0, Called Party
Number)-- Processing IE 125 (cs0, High-layer Compatibility)-- Processing
IE 126 (cs0, User-User) Protocol Discriminator: Q.931 (8)
len=10 Call Ref: len= 2 (reference 37114/0x90FA) (Terminator)
Message type: CALL PROCEEDING (2) [18 03 a9 83 9f] Channel ID
(len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan:
0
ChanSel:
Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 31 ] -- Starting simple switch on
'Zap/62-1' -- Accepting overlap call from '149718342' to
'0672083516' on channel 0/31, span 2 Protocol Discriminator: Q.931
(8) len=9 Call Ref: len= 2 (reference 4346/0x10FA)
(Originator) Message type: RELEASE (77) [08 02 87 d2]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: International network
(7)
Ext: 1 Cause: Unknown (82), class = Invalid message (5) ]-- Processing
IE 8 (cs0, Cause) -- Channel 0/31, span 2 got
hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate
Release Request Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 37114/0x90FA) (Terminator) Message type:
RELEASE COMPLETE (90) [08 02 81 90]Ulis*CLI Cause (len= 4)
[ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate
NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate
Null -- Hungup 'Zap/62-1' ConnecteurAzennUlis*CLI
exit
Can someone help me
please?
Thanks.
GIBERT
FrédéricDirect
: +33 (0) 1 7072 5101Mobile: +33 (0) 6 7208
3516Fax : +33 (0) 1 4692
0569
[EMAIL PROTECTED]http://www.viginetworks.fr
Ste
VIGINETWORKS1, rue Craiova92000 NanterreFrance
logo.gif___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Is asterisk that unstable ????
At 09:19 AM 12/30/04, you wrote: Greg - Cirelle Enterprises wrote: from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? I'm runing Asterisk for a year now as the IPBX of our little consulting firm. It stopped working only 4 times: two of these where power failures and the other two turned out to be Telco company problems (dead line). We have 2 PSTN lines (using Digium X101P cards), 5 intrernal VoIP extentions (Grandstream budgettone - one of which is located on another continent, using a Wifi connection to a near by village that hosts an ADSL router... don't ask) and 2 VoIP termination/origination lines. Of course, your mileage may very, but at least here there is no nightly restart script. Hope that helps you in any way. Gilad Are you running a stable (v 1.0 - 1.0.3) or cvs Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
No.. it's not that unstable. Some people are just paranoid. With my X100p's I do notice that caller id gives me trouble after about a week. Could just be in my head though. On Thu, 30 Dec 2004 08:52:15 -0500, Greg - Cirelle Enterprises [EMAIL PROTECTED] wrote: from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
On Thu, 2004-12-30 at 08:52 -0500, Greg - Cirelle Enterprises wrote: from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? There may be certain days when you check asterisk from the -HEAD branch that might be less stable than other days. The comments above seem to come from a certain type of admin personality. That personality is rampant in MS Windows shops and in some big iron shops. Right now this is the uptime from my main PBX. phone*CLI show uptime System uptime: 21 weeks, 21 hours, 16 minutes, 50 seconds Last reload: 1 week, 1 day, 15 hours, 53 minutes, 40 seconds As of this message, we have run about 7200 calls this month alone, or about 250 calls average per day right now. For November and December minus 2 days, 15300 calls or about 259 calls a day average. I don't have problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
This is the age old difference between Microsoft environments and Unix/Novell environments. I like to joke that Microsoft uptime is measured in hours Unix/Novell is always in years,months, and days. Although, I have to admit that Win 2k (server) and XP have substantially improved uptime and install lifetime. A lot of that can be traced back to leave it the alone if it works! I would be surprised if * actually HAD to be rebooted or restarted on any frequency. I suspect maintenance and changes would force restarts more often then clutter. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Thursday, December 30, 2004 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is asterisk that unstable On Thu, 2004-12-30 at 08:52 -0500, Greg - Cirelle Enterprises wrote: from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? There may be certain days when you check asterisk from the -HEAD branch that might be less stable than other days. The comments above seem to come from a certain type of admin personality. That personality is rampant in MS Windows shops and in some big iron shops. Right now this is the uptime from my main PBX. phone*CLI show uptime System uptime: 21 weeks, 21 hours, 16 minutes, 50 seconds Last reload: 1 week, 1 day, 15 hours, 53 minutes, 40 seconds As of this message, we have run about 7200 calls this month alone, or about 250 calls average per day right now. For November and December minus 2 days, 15300 calls or about 259 calls a day average. I don't have problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Thursday, December 30, 2004 7:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is asterisk that unstable On Thu, 2004-12-30 at 08:52 -0500, Greg - Cirelle Enterprises wrote: from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? There may be certain days when you check asterisk from the -HEAD branch that might be less stable than other days. The comments above seem to come from a certain type of admin personality. That personality is rampant in MS Windows shops and in some big iron shops. Right now this is the uptime from my main PBX. phone*CLI show uptime System uptime: 21 weeks, 21 hours, 16 minutes, 50 seconds Last reload: 1 week, 1 day, 15 hours, 53 minutes, 40 seconds As of this message, we have run about 7200 calls this month alone, or about 250 calls average per day right now. For November and December minus 2 days, 15300 calls or about 259 calls a day average. I don't have problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Any analog FXO or FXS interfaces in that box? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
On Thu, 2004-12-30 at 09:50 -0500, Luke Catranis wrote: I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. Oddly enough, My logs are approaching a year or more back and don't need to be rotated for size yet. I will do it the next time I have to do something with asterisk that time. My debug file is 418megs for over 1 year of logging. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
On Thu, 2004-12-30 at 07:58 -0700, Damon Estep wrote: \ Right now this is the uptime from my main PBX. phone*CLI show uptime System uptime: 21 weeks, 21 hours, 16 minutes, 50 seconds Last reload: 1 week, 1 day, 15 hours, 53 minutes, 40 seconds As of this message, we have run about 7200 calls this month alone, or about 250 calls average per day right now. For November and December minus 2 days, 15300 calls or about 259 calls a day average. I don't have problems. ___ Any analog FXO or FXS interfaces in that box? Of course not. FXO and FXS interfaces are for small deployments. We only have T1 interfaces and IAX2 interfaces. PRI in, a channelized T1 using 16 channels out, and a few calls a day out to our remote system via IAX2. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
I just make it a habit, the only issues I run into are after an IAX2 gridlock and my log files get filled up quickly... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Thursday, December 30, 2004 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Is asterisk that unstable On Thu, 2004-12-30 at 09:50 -0500, Luke Catranis wrote: I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. Oddly enough, My logs are approaching a year or more back and don't need to be rotated for size yet. I will do it the next time I have to do something with asterisk that time. My debug file is 418megs for over 1 year of logging. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callerid
Hi all, I was wondering how the easiest way to restrict the users ability to set caller ID would be ? I have some users that uses IAX to connect with me. multiple numers via iax. on outgoing calls I would like the user to only be able to set his range of numbers on the outgoing calls. Is there an easy way to do this ? /Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Helping communications to Asia area.
ALL, As a community is there anything we can do to help with communications to the Tsunami area ? we all sit on top of a welth of knowledge on communications can we use this to help these area's in any way? IE free sip calls , maybe there are * users in the area that we can send US calls to ? Jason enzo86 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield ___ Any analog FXO or FXS interfaces in that box? Of course not. FXO and FXS interfaces are for small deployments. We only have T1 interfaces and IAX2 interfaces. PRI in, a channelized T1 using 16 channels out, and a few calls a day out to our remote system via IAX2. -- Steven Critchfield [EMAIL PROTECTED] ___ Only for small deployments? How do you interface with your fax machines? analog alarm systems? pc modems? All of my large deployments require one or more of these elements, and one that I am currently working on is a MAX TNT - SER - * implemetation with 12 PRIs over a DS3 to the MAX TNT. Sure is a shame that I have to run 6 analog lines to the building because * can not provide analog TDM interfaces. I realize I could use a channel bank, but keep in mind, we have a DS3 coming in, so a channle bank would require demux of a DS1, and then demux to DS0 on a channel bank, and ebay pricing not withstanding, that costs a boatload of money. Problem with ebay gear is you have to buy two of everything to be safe (not to say we do not do it, TNTs are still cheaper on ebay even if you have to buy 3 to be safe). My point is that your assumption that only linux boxes will run for more than 30 days is opinionated and wrong. Any PC platform is only as stable as the sum of what you run on it, put a single analog interface in a red hat ES on $10,000 worth of hardware and you will have to reboot every 3 days. Run only stable software on a Linux OR Microsoft Server and uptime is not an issue unless you have a need to load every patch that ip put out, in which case both platforms typically require more frequent restarts. A better solution, use a good firewall and load patches less frequently. Your boxes have better uptime because of competent and educated decisions you have made (yes that is a compliment, you appear to be brilliant) when implementing them, like not installing known buggy interfaces. My MS boxes have similar uptime for the same reason. I see the value and need for both platforms on a daily basis. I realize several of my replies to you have been opinionated, but you frequently show your bias as well. In the end I respect your experience with * and have learned a few things from your posts after I filter the opinions out. With all due respect, Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Open ports on router in front of asterisk
- Original Message - From: Helder Rogério [MICROREDE] To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, December 30, 2004 3:13 PM Subject: Open ports on router in front of asterisk Hi, what are the ports that I must have open to Asterisk work correctly ? I have a Draytek 2500 (not V model) on one ip and a 2600V on another ip (both fixed ips). If I call 200 (echo test) I can hear the voice but can hear my own rtp ports from 1 to 2. Thanks in advance Helder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] callerid
Use a separate context for the outbound calls for that customer, check the caller ID in the dialplan before completing an outbound call using a PATTERN MATCH, and IF the pattern does not match the pattern of the customers numbers GOTO a step that sets the caller ID to the customers main phone number, then resume (GOTO) where you left off in the dialplan. Advise your customer that the caller ID they transmit must match known numbers or it will be changed by * before the call is completed. Make sure your terms of service agreement explains this carefully because it is not typical, with a lot of commercial PRIs you can set your caller ID to anything you wish. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, December 30, 2004 8:11 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] callerid Hi all, I was wondering how the easiest way to restrict the users ability to set caller ID would be ? I have some users that uses IAX to connect with me. multiple numers via iax. on outgoing calls I would like the user to only be able to set his range of numbers on the outgoing calls. Is there an easy way to do this ? /Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
[EMAIL PROTECTED] wrote: I like to joke that Microsoft uptime is measured in hours Unix/Novell is always in years,months, and days. It's not just you. A while back Microsoft was running a TV ad where a server was bragging that it was so reliable that it hadn't even seen the sysadmin for DAYS. Can you imagine what would have happened to a Unix company who ran the same ad? Everyone would be laughing their butts off... Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX hardware
Hi, I've been loosing my mind with NAT and read that IAX doesn't have problems about nat. Does anyone knows about hadware (routers and etc) support IAX? Best regards helder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. How do you rotate your logs? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 12/28/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF skipped when calling from ISDN to SIP...
Hello I have done the following test-network: IP-Phone = ASTERISK == ISDN PSTN Phone (SIP) + SER When I'm calling from the PSTN phone to the IP (SIP) phone: I cannot get ANY DTMF from PSTN, they seem destroyed by the codec (small scratches). I listen DTMF from IP-Phone (SIP INBAND!) When I'm calling from SIP phone to PSTN: Same result, no PSTN = IP DTMF ! Any ideas ? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 inbound FXO problem
I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nagios and Asterisk
Does anyone have some decent Nagios scripts out there that do more than monitor the proc itself? Rather than reinvite the wheel, figured I'd ask. I already saw the one on the wiki. Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSLink modem freeze
The device may also be doing RTP fixup, I guess. SIP uses RTP for the audio. Rodrigo P. Telles wrote: Hi Eric, Thanks every body that answered about this problem. About change de default SIP port (5060), I tried it at first and the UAC could authenticate but when I made a call and another side pick the phone up DSLink 200E freeze again. ie. there wasn't any port 5060 on transactions. I will have this DSL modem on my LAB asap and I will give feedback to the list. Thanks Eric Wieling aka ManxPower escreveu: On Cisco routers you can do something like no nat sip fixup 5060 and that will disable only the special SIP related nat features, but leave in all of the other NAT features. If a vendor does not include a similar ability in their SIP aware router they should be shot. --Eric C F wrote: I have this problem with Best Data DSL Modems, If I disable NAT (on the router, not in SIP) it works fine. You might be able to do the same just disable NAT and it will work, if you disable NAT then you will have to get a different router to be able to share the same IP, and if you use PPPoE you might not be able to do it, in which case you will have to get a different DSL modem. On Wed, 29 Dec 2004 20:00:28 -0600, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Rodrigo P. Telles wrote: Hi Folks, I've been having troubles with a DSL router (DSLink 200E) and SIP phones. When I put any SIP phone (software or hardware) to work behind that DSL router, it completely freeze. I ready tech specs of that DSL router and it says that SIP protocol is supported. ie. I tested two DSLink 200E with the same results. Turn off SIP support and let the generic NAT deal with it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Hi Randy, Randy MacKay wrote: I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. How do you rotate your logs? I have made a script to rotate mine, it's a little over complicated, but it works. asterisk is run as user, and logs are kept in /var/log/asterisk old logs are kept in /var/log/asterisk/old_logs crontab for root: # this is to rotate asterisk logs daily at 11:58 pm 58 23 * * * /etc/asterisk_logr.sh | mail - -s [asterisk] daily log rotate root asterisk_logr.sh: #!/bin/sh #Rotates log files for asterisk #variables today=`/bin/date +%m%d%Y` chown=/bin/chown mv=/bin/mv ls='/bin/ls -sh' #tell asterisk to do its thing echo echo --- echo # MESSAGES # echo --- /usr/sbin/asterisk -rx logger rotate echo # sleepy sleepy #sleep 2 #set shit up sourcef1=/var/log/asterisk/queue_log.0 sourcef2=/var/log/asterisk/event_log.0 sourcef3=/var/log/asterisk/asterisk_norm.log.0 sourcef4=/var/log/asterisk/asterisk_debug.log.0 sourcef5=/var/log/asterisk/screenlog.0 destf1=/var/log/asterisk/old_logs/queue_log.$today destf2=/var/log/asterisk/old_logs/event_log.$today destf3=/var/log/asterisk/old_logs/asterisk_norm.log.$today destf4=/var/log/asterisk/old_logs/asterisk_debug.log.$today destf5=/var/log/asterisk/old_logs/screenlog.0.$today #moveem to dest dir echo --- echo # QUEUE LOG # echo --- if [ -f $sourcef1 ]; then $mv $sourcef1 $destf1 echo - rotated $sourcef1 to $destf1 $chown root:wheel $destf1 echo - $destf1 file attributes set echo - file size: `$ls $destf1` echo else echo - no queue log to rotate echo - no queue log to give permissions to echo fi echo --- echo # EVENT LOG # echo --- if [ -f $sourcef2 ]; then $mv $sourcef2 $destf2 echo - rotated $sourcef2 to $destf2 $chown root:wheel $destf2 echo - $destf2 file attributes set echo - file size: `$ls $destf2` echo else echo - no event log to rotate echo - no event log to give permissions to echo fi echo --- echo # NORM LOG # echo --- if [ -f $sourcef3 ]; then $mv $sourcef3 $destf3 echo - rotated $sourcef3 to $destf3 $chown root:wheel $destf3 echo - $destf3 file attributes set echo - file size: `$ls $destf3` echo else echo no normal log to rotate echo no normal log to give permissions to echo fi echo --- echo # DEBUG LOG # echo --- if [ -f $sourcef4 ]; then $mv $sourcef4 $destf4 echo - rotated $sourcef4 to $destf4 $chown root:wheel $destf4 echo - $destf4 file attributes set echo - file size: `$ls $destf4` echo else echo no debug logfile to rotate echo no debug log to give permissions to echo fi echo --- echo # SCREEN LOG # echo --- if [ -f $sourcef5 ]; then $mv $sourcef5 $destf5 echo - rotated $sourcef5 to $destf5 $chown root:wheel $destf5 echo - $destf5 file attributes set echo - file size: `$ls $destf5` echo else echo no screen logfile to rotate echo no screen log to give permissions to echo fi -- Matt Gibson VOIP Administrator NJ Tech Solutions 1.314.480.4550 ex. 6400 1.877.999.4678 ex. 6400 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] callerid
Damon Estep wrote: Use a separate context for the outbound calls for that customer, check the caller ID in the dialplan before completing an outbound call using a PATTERN MATCH, and IF the pattern does not match the pattern of the customers numbers GOTO a step that sets the caller ID to the customers main phone number, then resume (GOTO) where you left off in the dialplan. Advise your customer that the caller ID they transmit must match known numbers or it will be changed by * before the call is completed. Make sure your terms of service agreement explains this carefully because it is not typical, with a lot of commercial PRIs you can set your caller ID to anything you wish. Ok I see. IS there an example to look at somewhere ? /Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
System uptime: 6 weeks, 1 day, 22 hours, 37 minutes, 55 seconds Last reload: 48 seconds Verbosity is atleast 3 System uptime: 7 weeks, 19 hours, 19 minutes, 48 seconds Last reload: 41 seconds Verbosity is atleast 3 System uptime: 7 weeks, 4 days, 9 hours, 25 minutes, 33 seconds Last reload: 36 seconds Verbosity is atleast 3 System uptime: 5 weeks, 5 days, 16 hours, 51 minutes, 43 seconds Last reload: 30 seconds Verbosity is atleast 3 System uptime: 6 weeks, 4 days, 22 hours, 43 minutes, 42 seconds Last reload: 21 seconds Verbosity is atleast 3 System uptime: 7 weeks, 4 days, 9 hours, 23 minutes, 14 seconds Last reload: 21 seconds Verbosity is atleast 3 System uptime: 7 weeks, 4 days, 9 hours, 31 minutes, 6 seconds Last reload: 16 seconds Verbosity is atleast 3 I wouldn't say it's unstable... these boxes all run res_perl and reload 100's of times a day. It all depends on if you know what the hell you're doing. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Randy MacKay Sent: Thursday, December 30, 2004 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Is asterisk that unstable I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. How do you rotate your logs? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 12/28/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF skipped when calling from ISDN to SIP...
Nicolas FOURNIL wrote: Hello I have done the following test-network: IP-Phone = ASTERISK == ISDN PSTN Phone (SIP) + SER When I'm calling from the PSTN phone to the IP (SIP) phone: I cannot get ANY DTMF from PSTN, they seem destroyed by the codec (small scratches). I listen DTMF from IP-Phone (SIP INBAND!) When I'm calling from SIP phone to PSTN: Same result, no PSTN = IP DTMF ! Inband DTMF only works with the ulaw and alaw codecs. This is not an Asterisk issue, it's just the way the other codecs work. You need RFC2833 or INFO DTMF if you want to use the other codecs. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] This item has been released from quarantine.
This file, which was attached to the message titled Asterisk-Users Digest, Vol 5, Issue 407 by [EMAIL PROTECTED] and was quarantined on 12/30/2004 11:01 AM, has been released. NOTE: If AutoProtect is enabled, then this restored attachment will be rescanned during the restore. If the attachment is still infected, the current virus detection policy will apply to this attachment. Message BodySYQb0469d35.txt Description: name ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX hardware
--Original Message Text--- From: Helder Rogério [MICROREDE] Date: Thu, 30 Dec 2004 15:32:59 - Hi, I've been loosing my mind with NAT and read that IAX doesn't have problems about nat. Does anyone knows about hadware (routers and etc) support IAX? Best regards helder That's the thing about IAXrouters don't need to support it...it's design deals with NAT implicitly. You simply port forward 4569 to your * server and you're set. I use four separate ITSPs yet I only have one port forwarded to my * server. Further, when I'm away I simply use Firefly as an IAX soft phone. It connects back in the same manner. IAX is great! Michael -- MichaelGraves[EMAIL PROTECTED] Sr.ProductSpecialistwww.pixelpower.com PixelPowerInc.[EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 inbound FXO problem
Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
Logger rotate from cli -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay Sent: Thursday, December 30, 2004 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Is asterisk that unstable I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. How do you rotate your logs? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 12/28/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX hardware
Hi, I've been loosing my mind with NAT and read that IAX doesn't have problems about nat. Does anyone knows about hadware (routers and etc) support IAX? Best regards helder Well, in fact, IAX doesn't needs an ALG (application level gateway) unlike SIP, IRC or FTP. It uses a one normal socket, so there is no more thinks to worry about, except maybe you will want HW that honors QoS prioritization (or better DiffServ) to make things run smoother. Miguel Ruiz Velasco __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
On Thu, 2004-12-30 at 08:29 -0700, Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield ___ Any analog FXO or FXS interfaces in that box? Of course not. FXO and FXS interfaces are for small deployments. We only have T1 interfaces and IAX2 interfaces. PRI in, a channelized T1 using 16 channels out, and a few calls a day out to our remote system via IAX2. Only for small deployments? How do you interface with your fax machines? analog alarm systems? pc modems? You probably shouldn't run an analog alarm system through a T1 or PRI. Consider them fragile and an alarm system should be on the most robust connection necessary. Fax machines are SO old. In my business, we use a fax machine about 2 times a month. It is connected to our life line analog phone line in our remote office. It is so much easier to send the information via email or a secure pickup on our servers than to fax. Granted we are looking at needing fax service for outbound soon, but that can be done without analog lines. Does any business outside of a ISP still use analog modems? I would think internet connections and good encryption would be the norm for those needs than an analog modem. All of my large deployments require one or more of these elements, and one that I am currently working on is a MAX TNT - SER - * implemetation with 12 PRIs over a DS3 to the MAX TNT. Sure is a shame that I have to run 6 analog lines to the building because * can not provide analog TDM interfaces. I realize I could use a channel bank, but keep in mind, we have a DS3 coming in, so a channle bank would require demux of a DS1, and then demux to DS0 on a channel bank, and ebay pricing not withstanding, that costs a boatload of money. Problem with ebay gear is you have to buy two of everything to be safe (not to say we do not do it, TNTs are still cheaper on ebay even if you have to buy 3 to be safe). Maybe you just need a T100P in your asterisk machine and a channel bank. On an ideal network, you might be able to get faxes working reliably via SIP to an asterisk machine and then out a channel bank. Your talking between $700 and $1000 if you ebay wisely and depending on redundancy of hardware. Granted it takes quite a long time before that price will equal out for the cost of just 6 analog lines. My point is that your assumption that only linux boxes will run for more than 30 days is opinionated and wrong. Any PC platform is only as stable as the sum of what you run on it, I never said anything about 30 days. I said it had to do with admin personalities. While yes it is opinionated, it doesn't reduce the truth that most MS admins as a course of maintenance just reboot machines. I have also seen this same mentality in admins on SAP deployments as well. Your boxes have better uptime because of competent and educated decisions you have made (yes that is a compliment, you appear to be brilliant) when implementing them, like not installing known buggy interfaces. My MS boxes have similar uptime for the same reason. I see the value and need for both platforms on a daily basis. I think you mistook my complaint _this time_ as to a personality trait of many of those who admin the machines as opposed to the OS on the machine. I don't like MS machines and I don't like how unstable they are in my production environment especially when compared to the linux boxes sitting right next to them. I realize several of my replies to you have been opinionated, but you frequently show your bias as well. In the end I respect your experience with * and have learned a few things from your posts after I filter the opinions out. Experience breeds opinions as much as any other influence. From my experience, I can deploy a linux solution with fewer troubles and less pain than a MS solution. So when I approach new problems, I am biased towards linux over anything else. I respect the licenses of FOSS but I am not a ESR or RMS puppet or disciple. With all due respect, I was due respect I must be faltering a bit this should be lightened up a bit more. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
what was wrong with logrotate? On Thu, 2004-12-30 at 10:57 -0500, Matt Gibson wrote: Hi Randy, Randy MacKay wrote: I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. How do you rotate your logs? I have made a script to rotate mine, it's a little over complicated, but it works. asterisk is run as user, and logs are kept in /var/log/asterisk old logs are kept in /var/log/asterisk/old_logs crontab for root: # this is to rotate asterisk logs daily at 11:58 pm 58 23 * * * /etc/asterisk_logr.sh | mail - -s [asterisk] daily log rotate root asterisk_logr.sh: #!/bin/sh #Rotates log files for asterisk #variables today=`/bin/date +%m%d%Y` chown=/bin/chown mv=/bin/mv ls='/bin/ls -sh' #tell asterisk to do its thing echo echo --- echo # MESSAGES # echo --- /usr/sbin/asterisk -rx logger rotate echo # sleepy sleepy #sleep 2 #set shit up sourcef1=/var/log/asterisk/queue_log.0 sourcef2=/var/log/asterisk/event_log.0 sourcef3=/var/log/asterisk/asterisk_norm.log.0 sourcef4=/var/log/asterisk/asterisk_debug.log.0 sourcef5=/var/log/asterisk/screenlog.0 destf1=/var/log/asterisk/old_logs/queue_log.$today destf2=/var/log/asterisk/old_logs/event_log.$today destf3=/var/log/asterisk/old_logs/asterisk_norm.log.$today destf4=/var/log/asterisk/old_logs/asterisk_debug.log.$today destf5=/var/log/asterisk/old_logs/screenlog.0.$today #moveem to dest dir echo --- echo # QUEUE LOG # echo --- if [ -f $sourcef1 ]; then $mv $sourcef1 $destf1 echo - rotated $sourcef1 to $destf1 $chown root:wheel $destf1 echo - $destf1 file attributes set echo - file size: `$ls $destf1` echo else echo - no queue log to rotate echo - no queue log to give permissions to echo fi echo --- echo # EVENT LOG # echo --- if [ -f $sourcef2 ]; then $mv $sourcef2 $destf2 echo - rotated $sourcef2 to $destf2 $chown root:wheel $destf2 echo - $destf2 file attributes set echo - file size: `$ls $destf2` echo else echo - no event log to rotate echo - no event log to give permissions to echo fi echo --- echo # NORM LOG # echo --- if [ -f $sourcef3 ]; then $mv $sourcef3 $destf3 echo - rotated $sourcef3 to $destf3 $chown root:wheel $destf3 echo - $destf3 file attributes set echo - file size: `$ls $destf3` echo else echo no normal log to rotate echo no normal log to give permissions to echo fi echo --- echo # DEBUG LOG # echo --- if [ -f $sourcef4 ]; then $mv $sourcef4 $destf4 echo - rotated $sourcef4 to $destf4 $chown root:wheel $destf4 echo - $destf4 file attributes set echo - file size: `$ls $destf4` echo else echo no debug logfile to rotate echo no debug log to give permissions to echo fi echo --- echo # SCREEN LOG # echo --- if [ -f $sourcef5 ]; then $mv $sourcef5 $destf5 echo - rotated $sourcef5 to $destf5 $chown root:wheel $destf5 echo - $destf5 file attributes set echo - file size: `$ls $destf5` echo else echo no screen logfile to rotate echo no screen log to give permissions to echo fi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RINGBACK then HANGUP
Here's some advice to myself. Why don't I check out the documentation before
I post. I think I'll bear that in mind in the future. Thanks me.
http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message
- Original Message -
From: Gary Ruddock (Swift Drinks) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 2:47 PM
Subject: [Asterisk-Users] RINGBACK then HANGUP
I am using the manager API to sucessfully ORIGINATE a call. I am using
PHP. I connect to asterisk and then connect an internal SIP phone to an
external phone.
?php
$timeout = 7500;
$login_extension = SIP/6001; // agent extension
$call_telephone = 9707; // customer's telephone
$socket = fsockopen(10.0.0.3,5038, $errno, $errstr,
$timeout);
if ($socket)
{
$call_person = Exten: . $call_telephone . \r\n;
$call_extension = Channel: . $login_extension . \r\n;
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: user\r\n);
fputs($socket, Secret: password\r\n\r\n);
fputs($socket, Action: Originate\r\n);
fputs($socket, $call_extension);
fputs($socket, Context: local\r\n);
fputs($socket, $call_person);
fputs($socket, Priority: 1\r\n);
fputs($socket, Callerid: \r\n\r\n);
}
?
The above code is used when I need to ringback a customer to tell them
their driver is outside. Problem is the call centre agent initiates the
call. The agent's SIP phone (SIP/6001) rings then he answers the call from
asterisk, asterisk then dials the customer.
I want asterisk to dial the customer, ring twice and then hangup. This
will save the agent's time and reduce our call costs. I don't want the
agent to be involved. I have tried messing around with the code above but
no result.
So my problem in summary is: I would like dial an external line, let it
ring twice and then hangup all via PHP.
Thanks for your help.
Gary Ruddock
swiftdrinks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
At 11:00 AM 12/30/04, you wrote: I wouldn't say it's unstable... these boxes all run res_perl and reload 100's of times a day. It all depends on if you know what the hell you're doing. bkw why are they reloading 100's of times a day?? greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: OT: [Asterisk-Users] Is asterisk that unstable ????
Steven Critchfield wrote: Does any business outside of a ISP still use analog modems? I would think internet connections and good encryption would be the norm for those needs than an analog modem. Funny story, and not really related, but I was talking to a guy who works upstairs from our office at a tech support place, that handles a lot of stuff for the local banks. Apparently one bank does all their nightly 'updates' to the 'central server' through 28.8 modems connecting via Telex (remember that program?) and pushing updates that way. It's early, I'm bored, and you asked! :) Matt -- Matt Gibson VOIP Administrator NJ Tech Solutions 1.314.480.4550 ex. 6400 1.877.999.4678 ex. 6400 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR IAX calls snafu ?
Hello, anytime I make an IAX2 call to another peer, I am collecting CDR records which are divided into small files, one for each accountholder customer that makes the calls. I have records of this nature: ""123456","1234567890","IAX2/[EMAIL PROTECTED]/5","2004-12-30 22:17:07","2004-12-30 22:17:07","2004-12-30 22:17:51",44,"ANSWERED" however, the ANSWERED status doesn't change on a per call basis, except when we dial ZAP channels from that 123456 extension, and also the call shows as being 44 seconds or N seconds, but the funny thing is that THE CALL WAS ACTUALLY NEVER ANSWERED BY THE REMOTE PARTY. Where can I debug the IAX2 billing / answer supervision / CDR functions ? Asterisk CVS-HEAD-12/02/04-17:57:31 built by [EMAIL PROTECTED] on a i686 running Linux -samudra No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.298 / Virus Database: 265.6.6 - Release Date: 12/28/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Issue with Mediatrix 1124
I have about 40 of these in production with Asterisk, send me an email off list with your sip.conf file and you extensions.conf file and I will help:) [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Malhotra Sent: Wednesday, December 29, 2004 5:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Issue with Mediatrix 1124 Hello I setup Mediatrix 1124, I am able to make incoming call but unable to make outgoing calls. When ever I tried it just gave me a beep sound. I appreciate any help on this. Thanks Deepak Malhotra This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardphones Console o Secretarial One
Hello! Am Mittwoch, 29. Dezember 2004 23:46 schrieb Alvaro Parres: I want to know if there is any console o secretarial hardphone that works with asterisks. I mean a phone in witch i can see the state of the extensions, the phone lineas, etc. Can hold o transfer easly a call, etc. It is not exactly a hardware operator phone, but we are using the Asterisk Flash Operator Panel successfully. http://www.asternic.org/ There is at least one phone with key expansion modules: http://www.snom.com/snom220_en.php I'd always prefere the software solution, because it is much more flexible. Andi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 inbound FXO problem
Kristian Kielhofner wrote: Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is sip show peers: www*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1004/10041.0.24.223 D 255.255.255.255 5060 Unmonitored 1003/10031.0.24.223 D 255.255.255.255 5060 Unmonitored 1002/10021.0.24.222 D 255.255.255.255 5061 Unmonitored 1001/10011.0.24.222 D 255.255.255.255 5060 Unmonitored 1000/1000(Unspecified)D 255.255.255.255 0 Unmonitored 5 sip peers loaded [4 online , 1 offline] Which seems to say the Sipura is registered... Here is sip.conf: [EMAIL PROTECTED] asterisk]# cat sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls [1000] type=friend username=1000 fromuser=1000 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1001] type=friend username=1001 fromuser=1001 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1002] type=friend username=1002 fromuser=1002 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1003] type=friend username=1003 secret=1003 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1003 nat=no disallow=all allow=ulaw [1004] type=friend username=1004 secret=1004 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1004 nat=no disallow=all allow=ulaw [EMAIL PROTECTED] asterisk]# Not sure what I'm doing wrong but any suggestions would be welcomed. And BTW - Happy Hollidays! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Justin Carlson wrote: what was wrong with logrotate? nothing, i just like doing things my own way :) this makes use of the asterisk rotate feature, and my own daily log rotating. meh. to each their own :) matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail and Zapatel
My PSTN line doesn't allways hang up properly after it goes to voicemail.
The problem occurs when a caller hangs up during the initial greeting.
Even though the hangup accured, voicemail continues to record, usually a
fast busy and/or a teleco generated please hangup now message. After the
voicemail.conf 'maxmessage=180' expires the line simply stays offhook.
The hardware is a X100P card and this is my extensions.conf for incoming
PSTN calls:
; Parameters for calls from PSTN
PSTN_RNG_EXTEN=SIP/251SIP/261SIP/211
PSTN_RNG_TIME=20 ; 20 seconds are about 6 rings
[inbound-pstn-local]
;
; Our local telephone line. We do not have caller ID so we set it to
; a sensible value.
;
exten = s,1,SetCallerID(Outside Caller 7377296)
exten = s,2,Dial(${PSTN_RNG_EXTEN},${PSTN_RNG_TIME},tr)
exten = s,3,Voicemail(u01)
exten = s,4,Hangup
exten = s,103,Voicemail(b01)
exten = s,104,Hangup
Adi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 and DTMF
For efficiency reliability, when SIP transmits DTMF as non-audio data, it uses RFC2833 or INFO. My question is - (not knowing much about IAX2) - when IAX2 transmits DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it using some other IAX2-specific mechanism with its own name? Thank you, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapatel ringing multiple SIP devices
My incoming PSTN line is configured to ring multiple extensions and eventually fall trough to voicemail if the call goes unanswered. If a SIP phone gets picked up just before voicemail should kick in, the call quite often goes to the phone but voicemail happens as well, the greeting plays and the who conversation up to maxmessage=180 gets recorded. Any idea how to fix that? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callerid
On Thu, 30 Dec 2004 [EMAIL PROTECTED] wrote: I was wondering how the easiest way to restrict the users ability to set caller ID would be ? I have some users that uses IAX to connect with me. multiple numers via iax. on outgoing calls I would like the user to only be able to set his range of numbers on the outgoing calls. Is there an easy way to do this ? Either use different contexts with the tests in the dialplan like another poster suggested or do a database lookup and check if the number is valid. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Matt Gibson wrote: nothing, i just like doing things my own way :) this makes use of the asterisk rotate feature, and my own daily log rotating. meh. to each their own :) matt Know you can make your own wheel before you drive someone else's car. This sums up the way I live - kind of goes along with your statement. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
Steven Critchfield wrote: On Thu, 2004-12-30 at 08:29 -0700, Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield ___ Any analog FXO or FXS interfaces in that box? Of course not. FXO and FXS interfaces are for small deployments. We only have T1 interfaces and IAX2 interfaces. PRI in, a channelized T1 using 16 channels out, and a few calls a day out to our remote system via IAX2. Only for small deployments? How do you interface with your fax machines? analog alarm systems? pc modems? I think most alarm companies continuously monitor the impedience of the line to detect tampering. This is the type of thing you'd want to install and forget. And Steve, why are you flaming Fedora Core users? When I jumped from Windows to Linux in 1965, RedHat 4.? was about the only thing available. At that time there was _zero_ Linux representation in the computer stores. If it weren't for Linus and RedHat, I'd be a VB programmer right now. There is a certain amount of loyalty, you know... -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp-0.0.2pre6
On Thu, Dec 30, 2004 at 01:38:43PM +1100, Adam Goryachev wrote: On Thu, 2004-12-30 at 01:48, Steve Underwood wrote: Hi Adam, You must be using a prehistoric GCC. Before 3.0, GCC didn't understand this C99 construct. Hmmm, well I have: gcc version 2.96 2731 (Red Hat Linux 7.3 2.96-110) You have gcc 3.0.4 or something for RH73, but it is undermaintained, IIRC and buggier than the 2.96 one. Yeah, I know, I'm using redhat, and it is really old. I'm waiting for a 'good' time to replace it with debian spandsp builds fine on Sarge. Anybody needs debs? -- Tzafrir Cohen [EMAIL PROTECTED] http://www.technion.ac.il/~tzafrir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 inbound FXO problem
On Thu, 30 Dec 2004 09:04:32 -0800, Steven P. Donegan wrote: Kristian Kielhofner wrote: Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is sip show peers: www*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1004/10041.0.24.223 D 255.255.255.255 5060 Unmonitored 1003/10031.0.24.223 D 255.255.255.255 5060 Unmonitored 1002/10021.0.24.222 D 255.255.255.255 5061 Unmonitored 1001/10011.0.24.222 D 255.255.255.255 5060 Unmonitored 1000/1000(Unspecified)D 255.255.255.255 0 Unmonitored 5 sip peers loaded [4 online , 1 offline] Which seems to say the Sipura is registered... Here is sip.conf: [EMAIL PROTECTED] asterisk]# cat sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls [1000] type=friend username=1000 fromuser=1000 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1001] type=friend username=1001 fromuser=1001 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1002] type=friend username=1002 fromuser=1002 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1003] type=friend username=1003 secret=1003 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1003 nat=no disallow=all allow=ulaw [1004] type=friend username=1004 secret=1004 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1004 nat=no disallow=all allow=ulaw [EMAIL PROTECTED] asterisk]# Not sure what I'm doing wrong but any suggestions would be welcomed. And BTW - Happy Hollidays! When I used the SPA-3000 I had to setup a special context in extensions.conf and then use a hotline dialplan setup in the SPA. This caused all calls incomming on the POTS line to immediately be forwarded to the Asterisk context. I essentially bypassed the SPA diaplan logic. You can find out more about this at www.voxilla.com which hosts a forum for SPA users. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and Zapatel
Is your X100P set for loop start or Kewl Start? I am betting loop start,
try changing to ks instead.
Lyle
- Original Message -
From: Adi Linden [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, December 30, 2004 11:08 AM
Subject: [Asterisk-Users] Voicemail and Zapatel
My PSTN line doesn't allways hang up properly after it goes to voicemail.
The problem occurs when a caller hangs up during the initial greeting.
Even though the hangup accured, voicemail continues to record, usually a
fast busy and/or a teleco generated please hangup now message. After the
voicemail.conf 'maxmessage=180' expires the line simply stays offhook.
The hardware is a X100P card and this is my extensions.conf for incoming
PSTN calls:
; Parameters for calls from PSTN
PSTN_RNG_EXTEN=SIP/251SIP/261SIP/211
PSTN_RNG_TIME=20 ; 20 seconds are about 6 rings
[inbound-pstn-local]
;
; Our local telephone line. We do not have caller ID so we set it to
; a sensible value.
;
exten = s,1,SetCallerID(Outside Caller 7377296)
exten = s,2,Dial(${PSTN_RNG_EXTEN},${PSTN_RNG_TIME},tr)
exten = s,3,Voicemail(u01)
exten = s,4,Hangup
exten = s,103,Voicemail(b01)
exten = s,104,Hangup
Adi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is asterisk that unstable ????
Hi Matt, Thanks for the information. I didn't mean for you to get beat up on this;-) I'm still learning linux, so your information is very helpful and I'm now going to try and figure it out. It will be a good challenge. I have been able to locate very little information about logs, so your reply and the others were very informative. Again Thanks, Randy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Gibson Sent: Thursday, December 30, 2004 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is asterisk that unstable Justin Carlson wrote: what was wrong with logrotate? nothing, i just like doing things my own way :) this makes use of the asterisk rotate feature, and my own daily log rotating. meh. to each their own :) matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 12/28/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 12/28/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 inbound FXO problem
Steven P. Donegan wrote: Kristian Kielhofner wrote: Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is sip show peers: www*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1004/10041.0.24.223 D 255.255.255.255 5060 Unmonitored 1003/10031.0.24.223 D 255.255.255.255 5060 Unmonitored 1002/10021.0.24.222 D 255.255.255.255 5061 Unmonitored 1001/10011.0.24.222 D 255.255.255.255 5060 Unmonitored 1000/1000(Unspecified)D 255.255.255.255 0 Unmonitored 5 sip peers loaded [4 online , 1 offline] Which seems to say the Sipura is registered... ..snip.. Steven, You need to create another friend for the Sipura FXO. You then need to configure PSTN Line to register as that user. You need to make sure that context= for your new friend allows the Sipura to forward those PSTN calls to where they need to go. Think of it like this - on a Sipura 2000, you have lines 1 + 2. On a Sipura 3000 your have lines 1 + 2 - it just so happens that they call line 2 PSTN Line. It still needs valid login information to get to *. Example (1003 is the Sipura 3000 Line 1 user): [1003] type=friend username=1003 secret=1003 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1003 nat=no disallow=all allow=ulaw [1003-in] --- can be anything, so long as you know what it is type=friend username=1003-in secret=1003-in canreinvite=no host=dynamic context=friends set to whatever you need it to be dtmfmode=rfc2833 nat=no disallow=all allow=ulaw Then, configure PSTN Line on the 3000 to register with your * machine as 1003-in. Hopefully this helps. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 inbound FXO problem
Michael Graves wrote: On Thu, 30 Dec 2004 09:04:32 -0800, Steven P. Donegan wrote: Kristian Kielhofner wrote: Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is sip show peers: www*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1004/10041.0.24.223 D 255.255.255.255 5060 Unmonitored 1003/10031.0.24.223 D 255.255.255.255 5060 Unmonitored 1002/10021.0.24.222 D 255.255.255.255 5061 Unmonitored 1001/10011.0.24.222 D 255.255.255.255 5060 Unmonitored 1000/1000(Unspecified)D 255.255.255.255 0 Unmonitored 5 sip peers loaded [4 online , 1 offline] Which seems to say the Sipura is registered... Here is sip.conf: [EMAIL PROTECTED] asterisk]# cat sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls [1000] type=friend username=1000 fromuser=1000 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1001] type=friend username=1001 fromuser=1001 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1002] type=friend username=1002 fromuser=1002 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 [EMAIL PROTECTED] disallow=all allow=ulaw [1003] type=friend username=1003 secret=1003 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1003 nat=no disallow=all allow=ulaw [1004] type=friend username=1004 secret=1004 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1004 nat=no disallow=all allow=ulaw [EMAIL PROTECTED] asterisk]# Not sure what I'm doing wrong but any suggestions would be welcomed. And BTW - Happy Hollidays! When I used the SPA-3000 I had to setup a special context in extensions.conf and then use a hotline dialplan setup in the SPA. This caused all calls incomming on the POTS line to immediately be forwarded to the Asterisk context. I essentially bypassed the SPA diaplan logic. You can find out more about this at www.voxilla.com which hosts a forum for SPA users. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry for all the included text - but it is relevant. The problem is not the Sipura-Asterisk connection - that is definitely happening - the problem is that Asterisk seems to want to authenticate the call in some way. And I have no clue at present as to how to make Asterisk happy with the inbound call. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 inbound FXO problem
Kristian Kielhofner wrote: Steven P. Donegan wrote: Kristian Kielhofner wrote: Steven P. Donegan wrote: I have a Sipura 3000, apparently configured correctly, when incoming calls arrive on the telco port they arrive properly on the Asterisk system - however they don't get routed properly. The Asterisk message: Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER sip:[EMAIL PROTECTED];tag=7f8072c0c46250f7o1 X's are there to not advertise my phone number :-) Any idea as to why any kind of authenticate would be done or would fail would be appreciated. Steven, It really seems like you need to setup an entry in sip.conf that PSTN Line on the sipura can register with. Do you have an entry in sip.conf for it? How is PSTN Line programmed? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Here is sip show peers: www*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 1004/10041.0.24.223 D 255.255.255.255 5060 Unmonitored 1003/10031.0.24.223 D 255.255.255.255 5060 Unmonitored 1002/10021.0.24.222 D 255.255.255.255 5061 Unmonitored 1001/10011.0.24.222 D 255.255.255.255 5060 Unmonitored 1000/1000(Unspecified)D 255.255.255.255 0Unmonitored 5 sip peers loaded [4 online , 1 offline] Which seems to say the Sipura is registered... ...snip.. Steven, You need to create another friend for the Sipura FXO. You then need to configure PSTN Line to register as that user. You need to make sure that context= for your new friend allows the Sipura to forward those PSTN calls to where they need to go. Think of it like this - on a Sipura 2000, you have lines 1 + 2. On a Sipura 3000 your have lines 1 + 2 - it just so happens that they call line 2 PSTN Line. It still needs valid login information to get to *. Example (1003 is the Sipura 3000 Line 1 user): [1003] type=friend username=1003 secret=1003 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1003 nat=no disallow=all allow=ulaw [1003-in] --- can be anything, so long as you know what it is type=friend username=1003-in secret=1003-in canreinvite=no host=dynamic context=friends set to whatever you need it to be dtmfmode=rfc2833 nat=no disallow=all allow=ulaw Then, configure PSTN Line on the 3000 to register with your * machine as 1003-in. Hopefully this helps. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The Sipura has registration entries in sip.conf for both lines - and from my earlier post appears to register just fine. I'm still clueless on the failure originally reported. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and Zapatel
On Thu, 30 Dec 2004, Lyle Giese wrote: Is your X100P set for loop start or Kewl Start? I am betting loop start, try changing to ks instead. Lyle This is what I have in /etc/asterisk/zapata.conf so it should be Kewl Start. [channels] ; X100P signalling=fxs_ks echocancel=yes ; You can set this to 32, 64, or 128, ; tweak to your needs. echocancelwhenbridged=yes echotraining=400; Asterisk trains to the beginning of the call, ; number is in milliseconds usecallerid=no ; This cause Asterisk to answer the call immediately ;callerid=asreceived context=inbound-pstn-local group=1 channel = 1 Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy reliability issues
Hmmm I could certainly see that being the issue. If it is the issue, though, then I think it's something that needs to be addressed. In my opinion, Digium needs to address it, as well as the whole provisioning via cli thing. I know Asterisk itself is a CLI oriented piece of software, but the more one can do do decrease configuration timing and issues the better off one is. I think it would be a benefit to allow the IAXy to be programmed via web interface. For that matter, from what I can tell via my own experimentation, it appears that you cannot use DNS to define the asterisk server to it. This is bad, since it means that if the IP of the asterisk server changes, you need to directly reprovision *all* of your IAXy devices For a new product, it has potential, hopefully these things will be addressed regards Paul - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 30, 2004 6:14 AM Subject: Re: [Asterisk-Users] IAXy reliability issues On Thu, 30 Dec 2004, Gary wrote: On Thu, 30 Dec 2004 00:12:51 -0700, Paul Fielding wrote: I've just picked up a pair of IAXy devices. They work fine except that they keep going offline. As in, I plug it in, it connects to Asterisk, I can dial and phone and all is dandy. Then, maybe 12h later, maybe 24, maybe 36, maybe 48, I'll either try to phone the device and not get through or I'll pick it up and the dialtone is gone. it's simply lost it's connection to Asterisk. If I unplug and plug back in, it reconnects and all is well. I'm running firmware v. 22. Anyone else experiencing this? Paul DHCP timeouts ?? Didn't somebody say that the IAXy doesn't renew its DHCP lease (ie its a BOOTP client). In which case, your DHCP server needs to give it an infinite lease. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More * weirdness
Well I am about to reserve a small padded room so I can bounce off the
walls without inflicting tooo much damage... Nothing is making sense at
this point. I tried several releases last night before settling on the
latest CVS (seemed to work the best). Asterisk was running GREAT for the
first few hours. Now since around 10AM EST SIP can't register and incoming
calls are rejected with all circuits are busy.
version:Asterisk CVS-HEAD-12/29/04-23:50:16
uptime: 12 hours, 26 minutes, 32 seconds
Console shows this when an incoming call is placed:
===
Don't know what to do if second ROSE component is of type 0x6
Dec 30 12:24:48 WARNING[2715]: chan_zap.c:7667 pri_dchannel: Ring requested on
channel 0/8 already in use on span 1. Hanging up owner.
-- B-channel 0/8 restarted on span 1
Don't know what to do if second ROSE component is of type 0x6
Dec 30 12:24:48 WARNING[2715]: chan_zap.c:7667 pri_dchannel: Ring requested on
channel 0/9 already in use on span 1. Hanging up owner.
-- B-channel 0/9 restarted on span 1
===
Existing calls have not been dropped so far!!! There are a couple channels
stuck like the ALSA call I tried to see if I could ring my extension.
===
phone*CLI show channels
Channel (ContextExtensionPri ) State Appl. Data
Zap/8-1 (lec-pri5737 5 ) Up Dial
SIP/2000SIP/2001SIP/2002SIP/2003SIP/2005SIP/2006|20|trhH
ALSA/default (local 2005 1 ) Ringing Dial
SIP/2005|20|t
Zap/9-1 (sales s2 ) Up Dial
SIP/2000SIP/2001SIP/2003SIP/2005|20|t
Zap/29-1 (outbound-max s1 ) Up Bridged Call Zap/5-1
Zap/5-1 (lec-pri8014 1 ) Up Dial
Zap/g2/8014
Zap/33-1 (outbound-max s1 ) Up Bridged Call Zap/10-1
Zap/10-1 (lec-pri8014 1 ) Up Dial
Zap/g2/8014
Zap/26-1 (outbound-max s1 ) Up Bridged Call Zap/2-1
Zap/2-1 (lec-pri8014 1 ) Up Dial
Zap/g2/8014
Zap/25-1 (outbound-max s1 ) Up Bridged Call Zap/1-1
Zap/1-1 (lec-pri8014 1 ) Up Dial
Zap/g2/8014
Zap/31-1 (outbound-max s1 ) Up Bridged Call Zap/7-1
Zap/7-1 (lec-pri8014 1 ) Up Dial
Zap/g2/8014
Zap/30-1 (outbound-max s1 ) Up Bridged Call Zap/6-1
Zap/6-1 (lec-pri8014 1 ) Up Dial
Zap/g2/8014
Zap/28-1 (outbound-max s1 ) Up Bridged Call Zap/4-1
Zap/4-1 (lec-pri8014 1 ) Up Dial
Zap/g2/8014
Zap/27-1 (outbound-max s1 ) Up Bridged Call Zap/3-1
Zap/3-1 (lec-pri8014 1 ) Up Dial
Zap/g2/8014
19 active channel(s)
==
Asterisk running as user asterisk. File permissions are according to the
non-root wiki. UDP packets are queuing up... I/O is borked somewhere:
=== strace ===
0.078863 --- SIGURG (Urgent I/O condition) @ 0 (0) ---
0.52 write(1, Urgent handler\n, 15) = 15
0.000566 rt_sigaction(SIGURG, {0x80a5d60, [URG], SA_RESTORER|SA_RESTART,
0x34aa58}, {0x80a5d60, [URG], SA_RESTORER|SA_RESTART, 0x34aa58}, 8) = 0
0.000181 sigreturn() = ? (mask now [])
0.000143 read(0, 0xbff63c2b, 1)= ? ERESTARTSYS (To be restarted)
311.584380 --- SIGURG (Urgent I/O condition) @ 0 (0) ---
0.56 write(1, Urgent handler\n, 15) = 15
0.000634 rt_sigaction(SIGURG, {0x80a5d60, [URG], SA_RESTORER|SA_RESTART,
0x34aa58}, {0x80a5d60, [URG], SA_RESTORER|SA_RESTART, 0x34aa58}, 8) = 0
0.000172 sigreturn() = ? (mask now [])
0.000142 read(0, 0xbff63c2b, 1)= ? ERESTARTSYS (To be restarted)
21.605115 --- SIGURG (Urgent I/O condition) @ 0 (0) ---
0.70 write(1, Urgent handler\n, 15) = 15
0.000146 rt_sigaction(SIGURG, {0x80a5d60, [URG], SA_RESTORER|SA_RESTART,
0x34aa58}, {0x80a5d60, [URG], SA_RESTORER|SA_RESTART, 0x34aa58}, 8) = 0
0.000222 sigreturn() = ? (mask now [])
0.000575 read(0, 0xbff63c2b, 1)= ? ERESTARTSYS (To be restarted)
0.066369 --- SIGURG (Urgent I/O condition) @ 0 (0) ---
0.49 write(1, Urgent handler\n, 15) = 15
0.000134 rt_sigaction(SIGURG, {0x80a5d60, [URG], SA_RESTORER|SA_RESTART,
0x34aa58}, {0x80a5d60, [URG], SA_RESTORER|SA_RESTART, 0x34aa58}, 8) = 0
0.000609 sigreturn() = ? (mask now [])
Re: [Asterisk-Users] Sipura 3000 inbound FXO problem
Steven P. Donegan wrote: The Sipura has registration entries in sip.conf for both lines - and from my earlier post appears to register just fine. I'm still clueless on the failure originally reported. Steven, So, of the 1001, 1002, 1003, etc. one of those in the PSTN line? Confusing at best. Anyways, what context are all of these? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Helping communications to Asia area.
I think this is a great idea...I have up to 5000 minutes I could donate, but unfortunetly my SIP service only allows calls to/from US and Canada. Gabe - Original Message - From: Jason p [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, December 30, 2004 7:18 AM Subject: [Asterisk-Users] Helping communications to Asia area. ALL, As a community is there anything we can do to help with communications to the Tsunami area ? we all sit on top of a welth of knowledge on communications can we use this to help these area's in any way? IE free sip calls , maybe there are * users in the area that we can send US calls to ? Jason enzo86 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems starting *
Hello, Hope this isn't TOO much of a newb question... I just created a new WBEL server with a fresh install of asterisk. When I try to load asterisk, it dies with some cryptic error messages. I've googled for them, but haven't found anything helpful. If anyone can point me in the right direction, I'd much appreciate it... Below is the output when I try to load asterisk. Thanks. --sk. = [EMAIL PROTECTED] root]# asterisk -cAsterisk CVS-v1-0-12/22/04-17:53:38, Copyright (C) 1999-2004 Digium.Written by Mark Spencer [EMAIL PROTECTED]=[ Booting.Junk at the beginning 49443303Warning, flexibel rate not heavily tested![EMAIL PROTECTED] root]# Ouch ... error while writing audio data: : Broken pipe [EMAIL PROTECTED] root]# ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp-0.0.2pre6
Hi, Tzafrir Cohen schrieb: spandsp builds fine on Sarge. Anybody needs debs? It does? I ITPed it a while ago, but placed it somewhat lower on my list when I saw it needed libtiff internals. I have debs for sarge that depend on libtiff3g, however I could not get it to work reliably with the more current libtiff4. I will try again tonight. Simon signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk dialing a Zap channel FXS instead of bridging to PSTN FXO
Hi All,
Channels 25-28 on a customers PBX are regular Zaptel FXO cards that
are hooked into 4 incomming phone lines. They are all in a group to do
automatic rollover for outgoing calls (if channel 25 is being used,
dial on channel 26, etc.).
Sometimes when a user is dialing a number, instead of bridging to one
of the FXO cards it goes and rings to Zap/1-1.
This doesnt occur all the time but some of the time, when it does
occur, I restart asterisk and it goes away for some time. I have also
tried changing the group number to something else, this doesnt seem to
help either.
I have a wait (w) before the numbers because the phone line doesnt
pick up right away and its to prevent asterisk from dialing before
there is a dial tone.
FYI, I have a rhino channel bank on the system going to a digium T100P
card, this is why my 4 FXO ports are so high.
Below I have snippets from my extensions.conf dial plan for the
outgoing context and my zapata.conf along with the error.
Error:
-- Executing Dial(Zap/6-1, Zap/g3/ww5632111) in new stack
-- Called g3/ww5632111
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
Extensions.conf context for outgoing calls
exten = _1NXXNXX,1,Dial(Zap/g3/ww${EXTEN})
exten = _NXXNXX,1,Dial(Zap/g3/w1${EXTEN})
exten = _NXX,1,Dial(Zap/g3/ww${EXTEN})
Zapata.conf snippet for the group
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=10.0
txgain=-4.5
group=3
channel = 25
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 26
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 27
context=from-pstn
signalling=fxs_ks
callerid=asrecieved
;echocancel=yes
;echocancelwhenbridged=yes
;echotraining=400
rxgain=12.0
txgain=-4.5
group=3
channel = 28
Thanks,
James
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
On Thu, 2004-12-30 at 10:22 -0700, Michael Welter wrote: Steven Critchfield wrote: On Thu, 2004-12-30 at 08:29 -0700, Damon Estep wrote: Only for small deployments? How do you interface with your fax machines? analog alarm systems? pc modems? I think most alarm companies continuously monitor the impedience of the line to detect tampering. This is the type of thing you'd want to install and forget. And Steve, why are you flaming Fedora Core users? When I jumped from Windows to Linux in 1965, RedHat 4.? was about the only thing available. 1965??? Neither windows nor linux existed in '65. '95 would be more plausible, but RH in '95 was pre 3.0.3 according to this historical version release dates page. http://www.owlriver.com/redhat_versions.html If you have been a RH user that long, you SHOULD know how abysmal the security and stability track record of a *.0 release of RH has been. Way too often it was rushed out the door for whatever reason. Upgrades from one release to another where painful or problematic. http://www.robotwisdom.com/linux/timeline.html 1993: 02Aug: SLS linux 1993: Aug: Debian linux 1994: 29Jan: Debian version 0.91 1994: 05Feb: Slackware 1.1.2 1994: RH 1.0 1994: 30Mar: MCC Interim 1.0+ 1994: Apr: SeSE Linux 1994: Oct: Xdenu Linux Of course in that time frame I was running NetBSD since linux caused me trouble with the cdrom drive I had at the time. At that time there was _zero_ Linux representation in the computer stores. If it weren't for Linus and RedHat, I'd be a VB programmer right now. There is a certain amount of loyalty, you know... I was burned without ever using redhat myself. Loyalty is a bandaid that hurts worse the longer you use it to cover trouble. All that and I'll tell you I have been burned with debian too, but less severe and only when I was asking for it by running testing or unstable code. I was at least the one who could choose my risk level. Of course, then I have disdain for a lot of the RH users and even more for a good portion of Fedora Core users who seem to be wanting ES but are too cheap to pay for it. Either way, there is usually a lot of either newbish or blinded by a contract users on RH and FC. Both are blinded to other options and tend to not want to think much about options. That specific behavior is one that I despise in people even outside of the computer realm. I don't understand why someone wouldn't want to know a fair amount about what they are doing. Of course I am the one who will fret and fuss over the tires I put on my car for nearly a month before I feel comfortable with actually buying the tires. All the times I actually spent that time mulling the options, turned out to be a good decision. This last time I put what quickly seemed to be good tires on my car, and I am quickly having to get reused to driving my car and limiting my driving style to not over drive the tires. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fw: [Asterisk-Users] Cisco 7690 Voicemail Problem
Anyone? :-) - Original Message - From: Matt Klein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 7:37 PM Subject: Re: [Asterisk-Users] Cisco 7690 Voicemail Problem a faint scratching sound of your voice coming out of the speaker? or loud and clear? I would say a medium crackly version..Actually its the voice from the vmail system ( ' The person at extension blah blah blah') So not too loud but not really clear either Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 and DTMF
Brent Goran wrote: For efficiency reliability, when SIP transmits DTMF as non-audio data, it uses RFC2833 or INFO. My question is - (not knowing much about IAX2) - when IAX2 transmits DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it using some other IAX2-specific mechanism with its own name? I believe it uses it's own method. IAX and IAX2 do not support inband dtmf in anyway. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy reliability issues
Paul Fielding wrote: Hmmm I could certainly see that being the issue. If it is the issue, though, then I think it's something that needs to be addressed. In my opinion, Digium needs to address it, as well as the whole provisioning via cli thing. I know Asterisk itself is a CLI oriented piece of software, but the more one can do do decrease configuration timing and issues the better off one is. I think it would be a benefit to allow the IAXy to be programmed via web interface. For that matter, from what I can tell via my own experimentation, it appears that you cannot use DNS to define the asterisk server to it. This is bad, since it means that if the IP of the asterisk server changes, you need to directly reprovision *all* of your IAXy devices For a new product, it has potential, hopefully these things will be addressed The IAXy does not have the CPU, RAM, or Flash to be able to add any significant features. I think it has 4k or RAM and 4k of Flash. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Helping communications to Asia area. ( I WILL!!!)
I can also Donate minutes ,, please contact if in that area are Asterisk users with Satellite, to interconnect. also if someone needs help I am available as far I can. regards Humberto On Thu, 30 Dec 2004 10:56:50 -0800, Gabriel Afana [EMAIL PROTECTED] wrote: I think this is a great idea...I have up to 5000 minutes I could donate, but unfortunetly my SIP service only allows calls to/from US and Canada. Gabe - Original Message - From: Jason p [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, December 30, 2004 7:18 AM Subject: [Asterisk-Users] Helping communications to Asia area. ALL, As a community is there anything we can do to help with communications to the Tsunami area ? we all sit on top of a welth of knowledge on communications can we use this to help these area's in any way? IE free sip calls , maybe there are * users in the area that we can send US calls to ? Jason enzo86 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] Cisco 7690 Voicemail Problem
On 30/12/2004 19:01, Paul A Brown wrote: Anyone? :-) If you turn down the volume on the phone slightly (Just one or two units) it goes away. I assume the output volume is overloading the phone and the DSP isn't clever enough to clip it. A longer term solution would be to boost the gain of whatever input you're using so that people don't have their phones turned up so loud. - Original Message - From: Matt Klein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 29, 2004 7:37 PM Subject: Re: [Asterisk-Users] Cisco 7690 Voicemail Problem a faint scratching sound of your voice coming out of the speaker? or loud and clear? I would say a medium crackly version..Actually its the voice from the vmail system ( ' The person at extension blah blah blah') So not too loud but not really clear either ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 and DTMF
On Thu, 30 Dec 2004, Brent Goran wrote: My question is - (not knowing much about IAX2) - when IAX2 transmits DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it using some other IAX2-specific mechanism with its own name? Yep - IAX's protocol is quite different from SIP/RTP. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXy reliability issues
[EMAIL PROTECTED] wrote: Paul Fielding wrote: Hmmm I could certainly see that being the issue. If it is the issue, though, then I think it's something that needs to be addressed. In my opinion, Digium needs to address it, as well as the whole provisioning via cli thing. I know Asterisk itself is a CLI oriented piece of software, but the more one can do do decrease configuration timing and issues the better off one is. I think it would be a benefit to allow the IAXy to be programmed via web interface. For that matter, from what I can tell via my own experimentation, it appears that you cannot use DNS to define the asterisk server to it. This is bad, since it means that if the IP of the asterisk server changes, you need to directly reprovision *all* of your IAXy devices For a new product, it has potential, hopefully these things will be addressed The IAXy does not have the CPU, RAM, or Flash to be able to add any significant features. I think it has 4k or RAM and 4k of Flash. Well, that certainly limits it's useful future. A neat toy, with limited market potential. I'd certainly like to hear about it's successor, then, because any kind of IAX-based ATA is something that would seem to have a future with Asterisk. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycomm IP500 dropping incoming calls
/snip Hello everyone. I can place outgoing calls no problem with my IP500 (using teliax as our provider). Thing is, when a call comes in, 90% of the time when I pick up the handset it drops the call immediately. I turned on SIP debug, and have listed my extension config from sip.conf. Any help is greatly appreciated sooo close TIA! -Ron /snip Figured out it was a NAT issue. We were using 1-1 NAT behind a Sonicwall. Changed it to simple port forwarding and all is fine. -Ron___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent login state saving?
Has there been any consideration of having asterisk save to a file the state of which agents are logged in such that after a restart (or crash) all agents don't have to manually re-login (after eventually realizing they're no longer logged in and not receiving calls :) ? -- Jon Lewis | I route Senior Network Engineer | therefore you are Atlantic Net| _ http://www.lewis.org/~jlewis/pgp for PGP public key_ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More * weirdness
Andrew McRory wrote: Well I am about to reserve a small padded room so I can bounce off the walls without inflicting tooo much damage... Nothing is making sense at this point. I tried several releases last night before settling on the latest CVS (seemed to work the best). Asterisk was running GREAT for the first few hours. Now since around 10AM EST SIP can't register and incoming calls are rejected with all circuits are busy. There is some heavy-duty stuff going on now in CVS-HEAD, and at least some of last night's builds are broken wrt SIP. I had to downgrade, for the first time in recent memory. I knew I was taking a risk, though; Mark had just added the skeletal code to do native encryption in IAX to the HEAD code. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
